From agranig at sipwise.com  Fri Mar  1 03:07:43 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Fri, 01 Mar 2013 09:07:43 +0100
Subject: [Spce-user] inbound rewriting rule
In-Reply-To: <512FD66A.6080501@anutel.ca>
References: <060FD34D-F154-43C1-A202-8A98286074BB@optionsdsl.ca>
	<512E75CB.7060309@anutel.ca>
	<CEEE718799582B47A55D7897DCB264FC280134@bvusaex.bestvoipusa.local>
	<512FD66A.6080501@anutel.ca>
Message-ID: <513061CF.80902@sipwise.com>

Hi,

On 02/28/2013 11:12 PM, Blagoja wrote:
> This is the line where i can see the call entering the server.
> New request - M=ACK R=sip:127.0.0.1:5080
> F=sip:025101165 at X.X.X.X:5060;user=phone;cpc=ordinary
> T=sip:5101199 at X.X.X.X:5060;user=phone IP=X.X.X.X:5060 (127.0.0.1:5060)
> ID=418ebe9cl43yuf9oa at 172.16.17.10
>   My rulle is : ^([1-9][0-9]+)$ and i believe is muching the number 5101199
> I tried instead of the above rulle with the number but no luck.
> Where and how the server is geting incoming did i cant findout

The ACK is the peer acknowledging the 200-ok of the CE, so this doesn't 
provide any suitable information. You need to look for INVITE messages.

Andreas



From jbonilla at sipwise.com  Fri Mar  1 04:30:21 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Fri, 1 Mar 2013 10:30:21 +0100
Subject: [Spce-user] inbound PSTN peer routing to local sip uri
In-Reply-To: <013501ce15b3$90327c00$b0977400$@murray@skyracktelecom.com>
References: <013501ce15b3$90327c00$b0977400$@murray@skyracktelecom.com>
Message-ID: <20130301103021.4a56afaf@quenya>

El Thu, 28 Feb 2013 13:00:19 -0000
"John Murray" <john.murray at skyracktelecom.com> escribi?:

> Hello,
> 
>  
> 
> I am trying to get calls coming from a PSTN peer (it's a PBX) to route to a
> local user with the sip uri.
> 
> The local user is a001002 at skyrack.net which is in the To: field but SPCE
> doesn't find it as a local user.
> 
>  
> 
> The RURI contains the user part but the domain part is the ip address of our
> server. Can I route this?
>
> The incoming INVITE
> 
> 
> Feb 28 11:55:43 skt-salem02 /usr/sbin/kamailio[2499]: INFO: <script>: New
> request - M=INVITE R=sip:a001002 at 100.100.100.182:5060
> 
> Feb 28 11:55:43 skt-salem02 /usr/sbin/kamailio[2499]: INFO: <script>: Callee
> is not local - R=sip:a001002 at 100.100.100.182:5060
> ID=BW115543276280213-1824721248 at 192.168.30.10
> 
>  

That should work. I usually route based on the username but as subscriber.
There shouldn't be any differences if the call comes from a peer.

You sure that you have an 'a001002' subscriber?

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130301/22e8ff5e/attachment.asc>

From jbonilla at sipwise.com  Fri Mar  1 04:38:28 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Fri, 1 Mar 2013 10:38:28 +0100
Subject: [Spce-user] SOAP array problem
In-Reply-To: <5119FF27.8020107@hfcnetwork.hu>
References: <5119FF27.8020107@hfcnetwork.hu>
Message-ID: <20130301103828.3b3b31ce@quenya>

El Tue, 12 Feb 2013 09:36:55 +0100
HFC-Network Kft - Szab? J?nos <szabojanos at hfcnetwork.hu> escribi?:

> Hi all!
> 
> I want to create a call forward set with a hunting group via SOAP using 
> php with nuSOAP, but I always get this answer:
> "*[faultcode] => soap:Client [faultstring] => Application failed during 
> request deserialization: Unspecified namespace for type 
> 'unnamed_struct_use_soapval[1]' *"
> 
> Am I have to create a special declaration to use this function?
> 
> 

hi

I see that no one answered your question. Did you manage to solve the issue?

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130301/3734837a/attachment.asc>

From john.murray at skyracktelecom.com  Fri Mar  1 04:52:42 2013
From: john.murray at skyracktelecom.com (John Murray)
Date: Fri, 1 Mar 2013 09:52:42 -0000
Subject: [Spce-user] inbound PSTN peer routing to local sip uri
In-Reply-To: <20130301103021.4a56afaf@quenya>
References: <013501ce15b3$90327c00$b0977400$@murray@skyracktelecom.com>
	<20130301103021.4a56afaf@quenya>
Message-ID: <51307a6a.8665b40a.4f88.ffffa26e@mx.google.com>

Jon hi,

Yes it looks like the subscriber is there.

>From admin interface:

SIP URI: a001002 at skyrack.net

And from mysql:
mysql> use kamailio;
Database changed
mysql> select * from subscriber where username='a001002';
+----+----------+-------------+----------+---------------+----------------------------------+----------------------------------+------+--------------------------------------+----------+---------------------+
| id | username | domain      | password | email_address | ha1                              | ha1b                             | rpid | uuid                                 | timezone | datetime_created    |
+----+----------+-------------+----------+---------------+----------------------------------+----------------------------------+------+--------------------------------------+----------+---------------------+
| 21 | a001002  | skyrack.net | xxxxxxxx |               | 90370930c7c47557f23cb76c4bce0f80 | e64a7fc0f08bc93569e556bc18df0b38 | NULL | 3978f230-c438-4e81-b6a3-7ddb90324777 |          | 2013-02-14 11:52:42 |
+----+----------+-------------+----------+---------------+----------------------------------+----------------------------------+------+--------------------------------------+----------+---------------------+
1 row in set (0.00 sec)

Thanks

John
-----Original Message-----
From: spce-user-bounces at lists.sipwise.com [mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of Jon Bonilla (Manwe)
Sent: 01 March 2013 09:30
To: spce-user at lists.sipwise.com
Subject: Re: [Spce-user] inbound PSTN peer routing to local sip uri

El Thu, 28 Feb 2013 13:00:19 -0000
"John Murray" <john.murray at skyracktelecom.com> escribi?:

> Hello,
> 
>  
> 
> I am trying to get calls coming from a PSTN peer (it's a PBX) to route 
> to a local user with the sip uri.
> 
> The local user is a001002 at skyrack.net which is in the To: field but 
> SPCE doesn't find it as a local user.
> 
>  
> 
> The RURI contains the user part but the domain part is the ip address 
> of our server. Can I route this?
>
> The incoming INVITE
> 
> 
> Feb 28 11:55:43 skt-salem02 /usr/sbin/kamailio[2499]: INFO: <script>: 
> New request - M=INVITE R=sip:a001002 at 100.100.100.182:5060
> 
> Feb 28 11:55:43 skt-salem02 /usr/sbin/kamailio[2499]: INFO: <script>: 
> Callee is not local - R=sip:a001002 at 100.100.100.182:5060
> ID=BW115543276280213-1824721248 at 192.168.30.10
> 
>  

That should work. I usually route based on the username but as subscriber.
There shouldn't be any differences if the call comes from a peer.

You sure that you have an 'a001002' subscriber?





From agranig at sipwise.com  Fri Mar  1 06:11:14 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Fri, 01 Mar 2013 12:11:14 +0100
Subject: [Spce-user] inbound PSTN peer routing to local sip uri
In-Reply-To: <51307a6a.8665b40a.4f88.ffffa26e@mx.google.com>
References: <013501ce15b3$90327c00$b0977400$@murray@skyracktelecom.com>
	<20130301103021.4a56afaf@quenya>
	<51307a6a.8665b40a.4f88.ffffa26e@mx.google.com>
Message-ID: <51308CD2.7040802@sipwise.com>

Hi,
On 03/01/2013 10:52 AM, John Murray wrote:
> SIP URI: a001002 at skyrack.net

Well, the problem here is that the call comes in as 
sip:a001002 at 100.100.100.182, but the subscriber is sip:a001002 at skyrack.net.

There is currently no way to also change the domain part via rewrite 
rules, so you'd have to do that hardcoded in the kamailio-lb config, 
like this:

# after the whole loose-route block:
if($rd == "100.100.100.182") {
   $rd = "skyrack.net";
}

You can use it more elegantly of course by using the template values of 
network.yml, but that's basically the idea for now.

Andreas



From szabojanos at hfcnetwork.hu  Fri Mar  1 06:12:53 2013
From: szabojanos at hfcnetwork.hu (=?ISO-8859-1?Q?HFC-Network_Kft_-_Szab=F3_J=E1nos?=)
Date: Fri, 01 Mar 2013 12:12:53 +0100
Subject: [Spce-user] SOAP array problem
In-Reply-To: <20130301103828.3b3b31ce@quenya>
References: <5119FF27.8020107@hfcnetwork.hu> <20130301103828.3b3b31ce@quenya>
Message-ID: <51308D35.4070804@hfcnetwork.hu>

Hi,

No, it's still under investigation.

?dv?zlettel,

Szab? J?nos
HFC-Network Kft

E-mail:  szabojanos at hfcnetwork.hu
Telefon: +36-20/344-4449

HFC Logo

2013.03.01. 10:38 keltez?ssel, Jon Bonilla (Manwe) ?rta:
> El Tue, 12 Feb 2013 09:36:55 +0100
> HFC-Network Kft - Szab? J?nos <szabojanos at hfcnetwork.hu> escribi?:
>
>> Hi all!
>>
>> I want to create a call forward set with a hunting group via SOAP using
>> php with nuSOAP, but I always get this answer:
>> "*[faultcode] => soap:Client [faultstring] => Application failed during
>> request deserialization: Unspecified namespace for type
>> 'unnamed_struct_use_soapval[1]' *"
>>
>> Am I have to create a special declaration to use this function?
>>
>>
> hi
>
> I see that no one answered your question. Did you manage to solve the issue?
>
>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130301/a1290b94/attachment.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: hfc_alairas.jpg
Type: image/jpeg
Size: 13079 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130301/a1290b94/attachment.jpg>

From agranig at sipwise.com  Fri Mar  1 06:21:14 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Fri, 01 Mar 2013 12:21:14 +0100
Subject: [Spce-user] inbound PSTN peer routing to local sip uri
In-Reply-To: <51308CD2.7040802@sipwise.com>
References: <013501ce15b3$90327c00$b0977400$@murray@skyracktelecom.com>
	<20130301103021.4a56afaf@quenya>
	<51307a6a.8665b40a.4f88.ffffa26e@mx.google.com>
	<51308CD2.7040802@sipwise.com>
Message-ID: <51308F2A.2080204@sipwise.com>

Another way would be to disable multi-domain handling at all by setting 
kamailio.proxy.ignore_auth_realm to "yes" in config.yml.

That way, the domain part will be ignored completely for authentication 
and also for looking up the callee.

Not sure if you want that though, it's actually a work-around for 
scenarios where lots of subscribers with arbitrary domains are deployed 
in your environment, and you still want to handle all of them on the 
platform without provisioning them within these arbitrary domains.

Andreas

On 03/01/2013 12:11 PM, Andreas Granig wrote:
> Hi,
> On 03/01/2013 10:52 AM, John Murray wrote:
>> SIP URI: a001002 at skyrack.net
>
> Well, the problem here is that the call comes in as
> sip:a001002 at 100.100.100.182, but the subscriber is sip:a001002 at skyrack.net.
>
> There is currently no way to also change the domain part via rewrite
> rules, so you'd have to do that hardcoded in the kamailio-lb config,
> like this:
>
> # after the whole loose-route block:
> if($rd == "100.100.100.182") {
>    $rd = "skyrack.net";
> }
>
> You can use it more elegantly of course by using the template values of
> network.yml, but that's basically the idea for now.
>
> Andreas
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user



From blagojaanusev at anutel.ca  Fri Mar  1 07:39:27 2013
From: blagojaanusev at anutel.ca (Blagoja)
Date: Fri, 01 Mar 2013 07:39:27 -0500
Subject: [Spce-user]   inbound rewriting rule
In-Reply-To: <CEEE718799582B47A55D7897DCB264FC280153@bvusaex.bestvoipusa.local>
References: <512FD9E2.3060003@anutel.ca>
	<CEEE718799582B47A55D7897DCB264FC28014A@bvusaex.bestvoipusa.local>
	<7B51A6CD-1098-44DE-B67B-35688C683785@anutel.ca>
	<CEEE718799582B47A55D7897DCB264FC280153@bvusaex.bestvoipusa.local>
Message-ID: <5130A17F.9060502@anutel.ca>

On 2/28/2013 6:11 PM, Kevin Masse wrote:
> That should still be ok.  Sipwise will see that the alias_number or
> E.164 number assigned to the Subscriber will match and stay local and
> not forward off to an Asterisk box.
> Kevin
>
>
> -----Original Message-----
> From: Blagoja [mailto:blagojaanusev at anutel.ca]
> Sent: Thursday, February 28, 2013 6:08 PM
> To: Kevin Masse
> Cc: <spce-user at lists.sipwise.com>
> Subject: [Spce-user] inbound rewriting rule
>
> Thanks for reply
> There is no asterisk involved i'm trying to route the call to local user
> and the number in my case is 7 digits eg. 510 1199
>
>
> Sent from my iPhone
>
> On 2013-02-28, at 5:35 PM, "Kevin Masse" <kmasse at questblue.com> wrote:
>
>> Good evening:
>>
>> Try to think of it this way:
>>
>> 15551231234 ->Your PSTN Carrier--> inbound call SIPWISE Outbound
> Rewrite
>> Rules for Callee ^(1|)([1-9][0-9][0-9]+)$               \2
>> -->  SIP Subscriber Asterisk Box where the call will go:
>> This will take a call inbound for your carrier like this 15551231234
> and
>> send it to your asterisk box (Subscriber) as 5551231234  The pattern
>> that your Asterisk Box is looking for is 5551231234 then send the call
>> to a destination.  Notice it dropped the 1 for the destination.
>>
>>
>> Let me know if this helps any.
>>
>> Kevin
>>
>>
>> -----Original Message-----
>> From: spce-user-bounces at lists.sipwise.com
>> [mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of Blagoja
>> Sent: Thursday, February 28, 2013 5:28 PM
>> To: spce-user at lists.sipwise.com
>> Subject: [Spce-user] inbound rewriting rule
>>
>> Hello,
>> Thanks for your help I start to understand but i'm stuck again
>>
>> My peer authenticate to remote server so when i call the number call
> is
>> entering my server and disconect. I did rewrite rulle but in
>> kamailio.proxy.log file saying No matching rewrite rules for 'ngcp-lb'
>> found
>>
>> This is the line where i can see the call entering the server.
>> New request - M=ACK R=sip:127.0.0.1:5080
>> F=sip:025101165 at X.X.X.X:5060;user=phone;cpc=ordinary
>> T=sip:5101199 at X.X.X.X:5060;user=phone IP=X.X.X.X:5060 (127.0.0.1:5060)
>> ID=418ebe9cl43yuf9oa at 172.16.17.10
>>   My rulle is : ^([1-9][0-9]+)$ and i believe is muching the number
>> 5101199 I tried instead of the above rulle with the number but no
> luck.
>> Where and how the server is geting incoming did i cant findout
>>
>> Regards.
>>
>>
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/listinfo/spce-user
>>
>
         I think that SCPE is not reading dialed incomung  number 
properly. The number from my provider is 5101199 and the rulle I set for 
that is ^([1-9][0-9]+)$
I had this problem in asterisk when provider is not sending did number 
in sip header . I understand that asterisk is diferent from spce but if 
I can tell somehow to spce how to get  the did like i did in asterisk. 
This is the link for asterisk to help understend what is my problem.
http://www.freepbx.org/support/documentation/howtos/how-to-get-the-did-of-a-sip-trunk-when-the-provider-doesnt-send-it-and-

Blagoja



From blagojaanusev at anutel.ca  Fri Mar  1 07:57:32 2013
From: blagojaanusev at anutel.ca (Blagoja)
Date: Fri, 01 Mar 2013 07:57:32 -0500
Subject: [Spce-user]  inbound rewriting rule
In-Reply-To: <513061CF.80902@sipwise.com>
References: <513061CF.80902@sipwise.com>
Message-ID: <5130A5BC.3010208@anutel.ca>

How do I do that. How do I look in to INVITE messages



From gary at netdiverse.com  Fri Mar  1 11:34:31 2013
From: gary at netdiverse.com (Gary Nieboer)
Date: Fri, 1 Mar 2013 09:34:31 -0700
Subject: [Spce-user] RTCP, Media Monitoring, Media Stats, MOS, etc..
Message-ID: <CAMXUi2TBTRXLeYm=x=rUOSdBVzvS2dbWDyymsiLrCWi6RbLcAA@mail.gmail.com>

This is a request for an update on media monitoring / performance stats,
RTCP, etc...

Although it would be great to have a reporting engine to capture
information on each call and provide summary stats and details per call, it
will still be good if SPCE stores jitter packet loss, etc.. for each leg of
a call that can be found in a log somewhere. If there is a way to see real
time stats on multiple legs like asterisk >sip show channelstats, that
would be great too.

What is available with SPCE?

Thanks,

Gary

>I saw a message #3 on May 31, 2012,  Vol 18   Issue 44
>
>Message: 3
>Hi Matthew,
>
>On 05/31/2012 12:33 PM, Matthew Ogden wrote:
>> There were som replies to my stream monitor, to use pcap etc, which is
>> helpful, but this is a specific case of being able to view or report on
>> jitter/packet loss per client, or per peer etc.
>
>Not yet. I had an interesting discussion with other kamailio developers
>last week where we decided to redesign the rtpproxy control protocol,
>which would give us the flexibility to pass back from mediaproxy-ng to
>the SIP proxy some RTP statistics when ending a media session, which can
>then for example be added as a SIP header or be processed in any other way.
>
>It won't be ready for prime time before late summer, I'd say. Sorry.
>
>Andreas
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130301/7c38f811/attachment.html>

From blagojaanusev at anutel.ca  Fri Mar  1 16:07:07 2013
From: blagojaanusev at anutel.ca (Blagoja)
Date: Fri, 01 Mar 2013 16:07:07 -0500
Subject: [Spce-user] Inbound DID in SIP header
In-Reply-To: <CEEE718799582B47A55D7897DCB264FC280134@bvusaex.bestvoipusa.local>
References: <060FD34D-F154-43C1-A202-8A98286074BB@optionsdsl.ca>
	<512E75CB.7060309@anutel.ca>
	<CEEE718799582B47A55D7897DCB264FC280134@bvusaex.bestvoipusa.local>
Message-ID: <5131187B.9040208@anutel.ca>



I think that SCPE is not reading dialed incoming  number
properly. The number from my provider is 5101199 and the rulle I set for
  ^([1-9][0-9]+)$
But as I can see in kamailio-proxy.log spce is not reding DID number so there is nothing to rewrite or route.
This is the part of the log where spce is treating that call and in BOLD where is not finding any number to rewrite

Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: New request - M=INVITE R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' F=sip:025101165 at 217.16.67.5:5060;user=phone;cpc=ordinary T=sip:5101199 at 217.16.67.5:5060;user=phone IP=217.16.67.5:5060 (127.0.0.1:5060) ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Call from PSTN - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Load peer preferences for peer host '1' - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Load dialplan IDs for peer host '1' - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Fetched dialplan IDs caller_in='1', callee_in='2' - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Dialog set mark total - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Dialog set mark peer to '1' - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: IP authorization for caller not provisioned, allow call - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: User-Provided CLI '025101165' taken from PAI - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Network-Provided CLI '025101165' taken from PAI - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Applying caller-in peer rewrite rules on user-provided CLI using dpid '1' - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Rewriting user-provided CLI '025101165' to '5101165' - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Applying caller-in peer rewrite rules on network-provided CLI using dpid '1' - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Rewriting network-provided CLI '025101165' to '5101165' - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Setting '5101165 at 172.16.17.10' as initiating user-provided CLI - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Setting '5101165 at 172.16.17.10' as initiating network-provided CLI - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
_*Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Searching for callee - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10*__*
*__*Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Applying callee-in peer rewrite rules using dpid '2' - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10*__*
*__*Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: No matching rewrite rules for 'ngcp-lb' found - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10*__*
*__*Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Clear callee preferences - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10*__*
*__*Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Callee is not local - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10*__*
*_Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Check Administrative NCOS level - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Check User NCOS level - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Checking URI 'ngcp-lb at 24.150.194.162' and CLI 'ngcp-lb' against block out lists - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Admin caller block mode is 'open' - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Caller block mode is 'open' - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Call to unknown user - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Accounting failed request for uuid '0' - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Setting acc source-leg for uuid '0': '0|025101165|217.16.67.5|5101165|||<null>|||0|call|<null>|1362171203.778528' - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Setting acc destination-leg for uuid '0': '0|||<null>|ngcp-lb|0|ngcp-lb|24.150.194.162|ngcp-lb|24.150.194.162' - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11991]: INFO: <script>: New request - M=ACK R=sip:127.0.0.1:5080 F=sip:025101165 at 217.16.67.5:5060;user=phone;cpc=ordinary T=sip:5101199 at 217.16.67.5:5060;user=phone IP=217.16.67.5:5060 (127.0.0.1:5060) ID=41a5d5f337v18e8j5 at 172.16.17.10

Can anybody helpme how to pach this

  I had this problem in asterisk when provider is not sending did number
in sip header . I understand that asterisk is diferent from spce but if
I can tell somehow to spce how to get  the DID number when provider is not sending in header.
This is the link for asterisk to help understend what is my problem.
http://www.freepbx.org/support/documentation/howtos/how-to-get-the-did-of-a-sip-trunk-when-the-provider-doesnt-send-it-and-


Blagoja

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130301/defe63c4/attachment.html>

From jchism2 at gmail.com  Fri Mar  1 16:37:41 2013
From: jchism2 at gmail.com (Jeremie Chism)
Date: Fri, 1 Mar 2013 15:37:41 -0600
Subject: [Spce-user] SMS
Message-ID: <EBB6CE73-9366-45FD-8E9E-8585458FA2B9@gmail.com>

Does sipwise support SMS and if so is there something that needs to be changed in the config to make it work. 

Sent from my iPhone


From jbonilla at sipwise.com  Fri Mar  1 16:46:15 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Fri, 1 Mar 2013 22:46:15 +0100
Subject: [Spce-user] SMS
In-Reply-To: <EBB6CE73-9366-45FD-8E9E-8585458FA2B9@gmail.com>
References: <EBB6CE73-9366-45FD-8E9E-8585458FA2B9@gmail.com>
Message-ID: <20130301224615.1255f743@quenya>

El Fri, 1 Mar 2013 15:37:41 -0600
Jeremie Chism <jchism2 at gmail.com> escribi?:

> Does sipwise support SMS and if so is there something that needs to be
> changed in the config to make it work. 


Could you tell us what exactly you mean by supporting sms? I guess the answer
is no, but sending an http request on events to a sms gateway shouldn't be too
difficult.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130301/584c21d1/attachment.asc>

From jchism2 at gmail.com  Fri Mar  1 18:11:40 2013
From: jchism2 at gmail.com (Jeremie Chism)
Date: Fri, 1 Mar 2013 17:11:40 -0600
Subject: [Spce-user] SMS
In-Reply-To: <20130301224615.1255f743@quenya>
References: <EBB6CE73-9366-45FD-8E9E-8585458FA2B9@gmail.com>
	<20130301224615.1255f743@quenya>
Message-ID: <CAG-pSShJ3mf8LdKKfwYjTj3=NyZ0gm_NkOBBBvmSin08MEYFrQ@mail.gmail.com>

Yes it is sent by sip message.

On Friday, March 1, 2013, Jon Bonilla wrote:

> El Fri, 1 Mar 2013 15:37:41 -0600
> Jeremie Chism <jchism2 at gmail.com <javascript:;>> escribi?:
>
> > Does sipwise support SMS and if so is there something that needs to be
> > changed in the config to make it work.
>
>
> Could you tell us what exactly you mean by supporting sms? I guess the
> answer
> is no, but sending an http request on events to a sms gateway shouldn't be
> too
> difficult.
>
>

-- 
Jeremie Chism
Triton Communications
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130301/a3639634/attachment.html>

From skchopperguy at gmail.com  Fri Mar  1 20:41:22 2013
From: skchopperguy at gmail.com (Skyler)
Date: Fri, 01 Mar 2013 17:41:22 -0800
Subject: [Spce-user] SMS
In-Reply-To: <CAG-pSShJ3mf8LdKKfwYjTj3=NyZ0gm_NkOBBBvmSin08MEYFrQ@mail.gmail.com>
References: <EBB6CE73-9366-45FD-8E9E-8585458FA2B9@gmail.com>
	<20130301224615.1255f743@quenya>
	<CAG-pSShJ3mf8LdKKfwYjTj3=NyZ0gm_NkOBBBvmSin08MEYFrQ@mail.gmail.com>
Message-ID: <513158C2.1000900@gmail.com>

In theory, if your SMSC can translate MESSAGE it should not be too hard 
to treat your SMSC in the same way peers are handled. ie: lookup local, 
if not found then go to smsc gateway. Mind you I'm sure it would be more 
involved than that.

what are you using as an smsc? I'd be into tinkering for fun to see if a 
stable solution could come of it.

Skyler

On 3/1/2013 3:11 PM, Jeremie Chism wrote:
> Yes it is sent by sip message.
>
> On Friday, March 1, 2013, Jon Bonilla wrote:
>
>     El Fri, 1 Mar 2013 15:37:41 -0600
>     Jeremie Chism <jchism2 at gmail.com <javascript:;>> escribi?:
>
>      > Does sipwise support SMS and if so is there something that needs
>     to be
>      > changed in the config to make it work.
>
>
>     Could you tell us what exactly you mean by supporting sms? I guess
>     the answer
>     is no, but sending an http request on events to a sms gateway
>     shouldn't be too
>     difficult.
>
>
>
> --
> Jeremie Chism
> Triton Communications
>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
>



From jchism2 at gmail.com  Fri Mar  1 20:50:57 2013
From: jchism2 at gmail.com (Jeremie Chism)
Date: Fri, 1 Mar 2013 19:50:57 -0600
Subject: [Spce-user] SMS
In-Reply-To: <513158C2.1000900@gmail.com>
References: <EBB6CE73-9366-45FD-8E9E-8585458FA2B9@gmail.com>
	<20130301224615.1255f743@quenya>
	<CAG-pSShJ3mf8LdKKfwYjTj3=NyZ0gm_NkOBBBvmSin08MEYFrQ@mail.gmail.com>
	<513158C2.1000900@gmail.com>
Message-ID: <9EF84F26-E19D-42E7-B77A-0FE3EB44BB21@gmail.com>

Here is a link to the wiki our carrier has

http://wiki.voipinnovations.com/SMS.ashx

Sent from my iPhone

On Mar 1, 2013, at 7:41 PM, Skyler <skchopperguy at gmail.com> wrote:

> In theory, if your SMSC can translate MESSAGE it should not be too hard to treat your SMSC in the same way peers are handled. ie: lookup local, if not found then go to smsc gateway. Mind you I'm sure it would be more involved than that.
> 
> what are you using as an smsc? I'd be into tinkering for fun to see if a stable solution could come of it.
> 
> Skyler
> 
> On 3/1/2013 3:11 PM, Jeremie Chism wrote:
>> Yes it is sent by sip message.
>> 
>> On Friday, March 1, 2013, Jon Bonilla wrote:
>> 
>>    El Fri, 1 Mar 2013 15:37:41 -0600
>>    Jeremie Chism <jchism2 at gmail.com <javascript:;>> escribi?:
>> 
>>     > Does sipwise support SMS and if so is there something that needs
>>    to be
>>     > changed in the config to make it work.
>> 
>> 
>>    Could you tell us what exactly you mean by supporting sms? I guess
>>    the answer
>>    is no, but sending an http request on events to a sms gateway
>>    shouldn't be too
>>    difficult.
>> 
>> 
>> 
>> --
>> Jeremie Chism
>> Triton Communications
>> 
>> 
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/listinfo/spce-user
> 
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130301/08aee657/attachment.html>

From skchopperguy at gmail.com  Fri Mar  1 22:05:18 2013
From: skchopperguy at gmail.com (Skyler)
Date: Fri, 01 Mar 2013 19:05:18 -0800
Subject: [Spce-user] SMS
In-Reply-To: <9EF84F26-E19D-42E7-B77A-0FE3EB44BB21@gmail.com>
References: <EBB6CE73-9366-45FD-8E9E-8585458FA2B9@gmail.com>
	<20130301224615.1255f743@quenya>
	<CAG-pSShJ3mf8LdKKfwYjTj3=NyZ0gm_NkOBBBvmSin08MEYFrQ@mail.gmail.com>
	<513158C2.1000900@gmail.com>
	<9EF84F26-E19D-42E7-B77A-0FE3EB44BB21@gmail.com>
Message-ID: <51316C6E.2060201@gmail.com>

That's very similar to what I was about to start work on. IM between 
multiple SPCE boxes. What the link describes is basically an IM being 
passed along to another system if not found locally. Can definitely be 
done but will take more thought into the handling and testing though.

Skyler

On 3/1/2013 5:50 PM, Jeremie Chism wrote:
> Here is a link to the wiki our carrier has
>
> http://wiki.voipinnovations.com/SMS.ashx
>
> Sent from my iPhone
>
> On Mar 1, 2013, at 7:41 PM, Skyler <skchopperguy at gmail.com
> <mailto:skchopperguy at gmail.com>> wrote:
>
>> In theory, if your SMSC can translate MESSAGE it should not be too
>> hard to treat your SMSC in the same way peers are handled. ie: lookup
>> local, if not found then go to smsc gateway. Mind you I'm sure it
>> would be more involved than that.
>>
>> what are you using as an smsc? I'd be into tinkering for fun to see if
>> a stable solution could come of it.
>>
>> Skyler
>>
>> On 3/1/2013 3:11 PM, Jeremie Chism wrote:
>>> Yes it is sent by sip message.
>>>
>>> On Friday, March 1, 2013, Jon Bonilla wrote:
>>>
>>>    El Fri, 1 Mar 2013 15:37:41 -0600
>>>    Jeremie Chism <jchism2 at gmail.com <mailto:jchism2 at gmail.com>
>>> <javascript:;>> escribi?:
>>>
>>>     > Does sipwise support SMS and if so is there something that needs
>>>    to be
>>>     > changed in the config to make it work.
>>>
>>>
>>>    Could you tell us what exactly you mean by supporting sms? I guess
>>>    the answer
>>>    is no, but sending an http request on events to a sms gateway
>>>    shouldn't be too
>>>    difficult.
>>>
>>>
>>>
>>> --
>>> Jeremie Chism
>>> Triton Communications
>>>
>>>
>>> _______________________________________________
>>> Spce-user mailing list
>>> Spce-user at lists.sipwise.com <mailto:Spce-user at lists.sipwise.com>
>>> http://lists.sipwise.com/listinfo/spce-user
>>>
>>
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com <mailto:Spce-user at lists.sipwise.com>
>> http://lists.sipwise.com/listinfo/spce-user



From blagojaanusev at anutel.ca  Sun Mar  3 18:06:39 2013
From: blagojaanusev at anutel.ca (Blagoja)
Date: Sun, 03 Mar 2013 18:06:39 -0500
Subject: [Spce-user] non existing right pvar
In-Reply-To: <CEEE718799582B47A55D7897DCB264FC280153@bvusaex.bestvoipusa.local>
References: <512FD9E2.3060003@anutel.ca>
	<CEEE718799582B47A55D7897DCB264FC28014A@bvusaex.bestvoipusa.local>
	<7B51A6CD-1098-44DE-B67B-35688C683785@anutel.ca>
	<CEEE718799582B47A55D7897DCB264FC280153@bvusaex.bestvoipusa.local>
Message-ID: <5133D77F.6090402@anutel.ca>

I'm geting this error on incoming call in proxy.cfg



Mar  3 17:59:52 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[2254]: ERROR: 
<core> [lvalue.c:347]: non existing right pvar
Mar  3 17:59:52 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[2254]: ERROR: 
<core> [lvalue.c:407]: assignment failed at pos: (1920,9-1920,28)
Mar  3 17:59:52 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[2254]: ERROR: 
avpops [avpops_impl.c:334]: incomplet uri <sip:x.x.x.x:5060>





From axessofficetheo at gmail.com  Mon Mar  4 01:59:03 2013
From: axessofficetheo at gmail.com (Theo)
Date: Mon, 4 Mar 2013 08:59:03 +0200
Subject: [Spce-user] Upper Registration to Asterisk - transfers not
	working
In-Reply-To: <CA+9+E=XRB-hPxK1ZTBU5ebhnaq-y0wS0=B=8ZaeVCheBC6+tWg@mail.gmail.com>
References: <CA+9+E=X_qiRReX=c_XubTD89HEV2HApxFw1yRDgDa2nW0o_SZg@mail.gmail.com>
	<20130215213405.58b3a93a@quenya>
	<CA+9+E=Wnubt_Wy+xHjk0C6kw0XQ8tc2-agbFvkevcTpYJEqg4Q@mail.gmail.com>
	<720785848-1361030221-cardhu_decombobulator_blackberry.rim.net-863913820-@b16.c11.bise7.blackberry>
	<CA+9+E=Xh_1QHALsUzOhnUOK7q4uD8ZyoLwHoEs2R-zHh=7Ox5g@mail.gmail.com>
	<CA+9+E=U+N0GJY7QOmvTYgW5fVsetEZ_e2rYTOtp_EWeH5U7HTg@mail.gmail.com>
	<CA+9+E=U215fg8=y_ff46U2pyUxB4imXi1-i1Sx3bMbtECZ+28A@mail.gmail.com>
	<51224FFD.4020408@sipwise.com> <51225420.5000607@sipwise.com>
	<CA+9+E=U-t1paf0KW+H-vHkOwGFBkbn1=8NFd5YUPdEwm_xyZsg@mail.gmail.com>
	<5122598B.1060703@sipwise.com>
	<CA+9+E=XeYZRHuBQkhsHBTMCarq2bOdX9c+93suqyCCb-Gj+3Ag@mail.gmail.com>
	<20130226152313.2d977e1d@quenya>
	<CA+9+E=WQUrn1HbeekF2fchdZ+7kX54Snphe0QY6W_x4oXN6yyA@mail.gmail.com>
	<CA+9+E=XRB-hPxK1ZTBU5ebhnaq-y0wS0=B=8ZaeVCheBC6+tWg@mail.gmail.com>
Message-ID: <CA+9+E=WU=Sa9NTZ=HJrzLXZ+VnTON_--+9fBaHtivs+Lfh7Vwg@mail.gmail.com>

Hi

Not sure if I am breaking protocol by bumping - if I did - apologies. I am
almost ready to put this in as our SBC upper registration solution (more to
follow after that I hope :-) - but need to resolve the attended transfer
problem. Any ideas on this? Sip Header filtering is off altogether (does
that have any caveats?). Transfer working fine now for all phones tested,
but attended transfer not.

Cheers

On Tue, Feb 26, 2013 at 4:43 PM, Theo <axessofficetheo at gmail.com> wrote:

> ah,
>
> It did not fix the issue of attended transfer, but it DID fix the other
> phone (fanvil) that didn't work on any type of transfer. So, yes, something
> to do with the headers.
>
> Now onto the attended transfer :-)
>
>
> On Tue, Feb 26, 2013 at 4:41 PM, Theo <axessofficetheo at gmail.com> wrote:
>
>> Hi Jon
>>
>> Assuming that commenting out like this:
>>
>> #header_filter=whitelist
>>
>> Is all i have to do to disable it, it doesn't work. Am I correct in
>> assuming that will disable header filtering?
>>
>>
>>
>> On Tue, Feb 26, 2013 at 4:23 PM, Jon Bonilla <jbonilla at sipwise.com>wrote:
>>
>>> El Tue, 26 Feb 2013 15:27:26 +0200
>>> Theo <axessofficetheo at gmail.com> escribi?:
>>>
>>> > Hi
>>> >
>>> > Real life took over for a little bit, back at this now. It's not
>>> turning
>>> > out to be as easy as I had hoped :-(
>>> >
>>> > I have 2 issues currently with this.
>>> >
>>> > 1. SNOM 300 - blind transfer works fine but attended transfer doesn't.
>>> It
>>> > does send the refer-to and using ngrep I don't see a difference
>>> between the
>>> > two, but then again I am usually wrong. The attended transfer just
>>> seems to
>>> > put me on hold and not get out of that.
>>> >
>>> > 2. I have another phone for testing - a Fanvil. Transferring connected
>>> > directly to asterisk works fine, but through SCPE it doesn't - error
>>> > message "refer-to missing"
>>> >
>>> > How do I troubleshoot this further?
>>> >
>>> >
>>>
>>> Try to check if the headers of are the same on both legs. Maybe sems is
>>> eating
>>> headers. Another option would be to disable header filtering
>>> in /etc/sems/etc/ngcp.sbcprofile.conf ans retry.
>>>
>>>
>>> _______________________________________________
>>> Spce-user mailing list
>>> Spce-user at lists.sipwise.com
>>> http://lists.sipwise.com/listinfo/spce-user
>>>
>>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130304/ffd08cdf/attachment.html>

From joan at megatro.com  Mon Mar  4 04:24:29 2013
From: joan at megatro.com (Joan Jou)
Date: Mon, 04 Mar 2013 10:24:29 +0100
Subject: [Spce-user] Statistics
Message-ID: <5134684D.203@megatro.com>

Hi,
On a new intallation.
Everythink looks running but Information/statistics  stay "loading...".

Thank you,


-- 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130304/58f04b58/attachment.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signatura1.gif
Type: image/gif
Size: 13564 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130304/58f04b58/attachment.gif>

From chessmancaryl at gmail.com  Mon Mar  4 06:48:51 2013
From: chessmancaryl at gmail.com (chess man)
Date: Mon, 4 Mar 2013 12:48:51 +0100
Subject: [Spce-user] class diagramm
Message-ID: <CAB9Pg_qH=nYDqmXF6UYdFcvW8Y8h-zfDsNZDdvA3Fp_MhavELQ@mail.gmail.com>

Good morning

Could you please provide me with the class UML diagramm of the sip wise
solution
and the schema of the database management.

Best Regards.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130304/d4915ae9/attachment.html>

From kmasse at questblue.com  Mon Mar  4 07:15:34 2013
From: kmasse at questblue.com (Kevin Masse)
Date: Mon, 4 Mar 2013 07:15:34 -0500
Subject: [Spce-user] Statistics
References: <5134684D.203@megatro.com>
Message-ID: <CEEE718799582B47A55D7897DCB264FC280176@bvusaex.bestvoipusa.local>

Good morning Joan, if you are using Internet Explorer try using Chrome
or Firefox.

 

I have noticed that IE does have this issue with the stats page.

 

Kevin

 

 

 

From: spce-user-bounces at lists.sipwise.com
[mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of Joan Jou
Sent: Monday, March 04, 2013 4:24 AM
To: spce-user at lists.sipwise.com
Subject: [Spce-user] Statistics

 

Hi,
On a new intallation.
Everythink looks running but Information/statistics  stay  "loading...".

Thank you,



-- 
 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130304/9e1dfeb5/attachment.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.gif
Type: image/gif
Size: 13564 bytes
Desc: image001.gif
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130304/9e1dfeb5/attachment.gif>

From kmasse at questblue.com  Mon Mar  4 07:41:22 2013
From: kmasse at questblue.com (Kevin Masse)
Date: Mon, 4 Mar 2013 07:41:22 -0500
Subject: [Spce-user] Statistics
References: <5134684D.203@megatro.com>
	<CEEE718799582B47A55D7897DCB264FC280176@bvusaex.bestvoipusa.local>
	<51349462.1080701@megatro.com>
Message-ID: <CEEE718799582B47A55D7897DCB264FC280177@bvusaex.bestvoipusa.local>

Ok, I will step away from this one.  I do not have additional
suggestions.

 

 

Kevin

 

 

 

From: Joan Jou [mailto:joan at megatro.com] 
Sent: Monday, March 04, 2013 7:33 AM
To: Kevin Masse
Subject: Re: [Spce-user] Statistics

 

Good morning,
I try it whith  Explorer, Chrome and Firefox.

Joan,

Al 04/03/2013 13:15, En/na Kevin Masse ha escrit:

	Good morning Joan, if you are using Internet Explorer try using
Chrome or Firefox.

	 

	I have noticed that IE does have this issue with the stats page.

	 

	Kevin

	 

	 

	 

	From: spce-user-bounces at lists.sipwise.com [
mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of Joan Jou
	Sent: Monday, March 04, 2013 4:24 AM
	To: spce-user at lists.sipwise.com
	Subject: [Spce-user] Statistics

	 

	Hi,
	On a new intallation.
	Everythink looks running but Information/statistics  stay
"loading...".
	
	Thank you,
	
	
	

	-- 
	 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130304/b6070cea/attachment.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.gif
Type: image/gif
Size: 13564 bytes
Desc: image001.gif
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130304/b6070cea/attachment.gif>

From agranig at sipwise.com  Mon Mar  4 07:55:01 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Mon, 04 Mar 2013 13:55:01 +0100
Subject: [Spce-user] class diagramm
In-Reply-To: <CAB9Pg_qH=nYDqmXF6UYdFcvW8Y8h-zfDsNZDdvA3Fp_MhavELQ@mail.gmail.com>
References: <CAB9Pg_qH=nYDqmXF6UYdFcvW8Y8h-zfDsNZDdvA3Fp_MhavELQ@mail.gmail.com>
Message-ID: <513499A5.5000208@sipwise.com>

Hi,

On 03/04/2013 12:48 PM, chess man wrote:
> Could you please provide me with the class UML diagramm of the sip wise
> solution

There is http://sipwise.com/doc/2.7/spce/ar01s02.html, but there is no 
such thing as a class UML diagram, as the solution is not a single 
application, but consists of quite a lot of different sub-systems.

> and the schema of the database management.

You can create one using http://schemaspy.sourceforge.net/, which 
produces quite usable db schema descriptions.

Andreas



From agranig at sipwise.com  Mon Mar  4 07:56:47 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Mon, 04 Mar 2013 13:56:47 +0100
Subject: [Spce-user] Statistics
In-Reply-To: <CEEE718799582B47A55D7897DCB264FC280177@bvusaex.bestvoipusa.local>
References: <5134684D.203@megatro.com>
	<CEEE718799582B47A55D7897DCB264FC280176@bvusaex.bestvoipusa.local>
	<51349462.1080701@megatro.com>
	<CEEE718799582B47A55D7897DCB264FC280177@bvusaex.bestvoipusa.local>
Message-ID: <51349A0F.7050403@sipwise.com>

Hi,

Check /var/log/ngcp/oss.log to see if there are any errors. Maybe it 
doesn't find the rrd files in /var/lib/collectd/rrd, maybe because 
collectd is not running?

Andreas

On 03/04/2013 01:41 PM, Kevin Masse wrote:
> Ok, I will step away from this one.  I do not have additional suggestions.
>
> Kevin
>
> *From:*Joan Jou [mailto:joan at megatro.com]
> *Sent:* Monday, March 04, 2013 7:33 AM
> *To:* Kevin Masse
> *Subject:* Re: [Spce-user] Statistics
>
> Good morning,
> I try it whith  Explorer, Chrome and Firefox.
>
> Joan,
>
> Al 04/03/2013 13:15, En/na Kevin Masse ha escrit:
>
>     Good morning Joan, if you are using Internet Explorer try using
>     Chrome or Firefox.
>
>     I have noticed that IE does have this issue with the stats page.
>
>     Kevin
>
>     *From:*spce-user-bounces at lists.sipwise.com
>     <mailto:spce-user-bounces at lists.sipwise.com>
>     [mailto:spce-user-bounces at lists.sipwise.com] *On Behalf Of *Joan Jou
>     *Sent:* Monday, March 04, 2013 4:24 AM
>     *To:* spce-user at lists.sipwise.com <mailto:spce-user at lists.sipwise.com>
>     *Subject:* [Spce-user] Statistics
>
>     Hi,
>     On a new intallation.
>     Everythink looks running but Information/statistics  stay  "loading...".
>
>     Thank you,
>
>
>     --
>
>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
>



From jbonilla at sipwise.com  Mon Mar  4 14:55:27 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Mon, 4 Mar 2013 20:55:27 +0100
Subject: [Spce-user] non existing right pvar
In-Reply-To: <5133D77F.6090402@anutel.ca>
References: <512FD9E2.3060003@anutel.ca>
	<CEEE718799582B47A55D7897DCB264FC28014A@bvusaex.bestvoipusa.local>
	<7B51A6CD-1098-44DE-B67B-35688C683785@anutel.ca>
	<CEEE718799582B47A55D7897DCB264FC280153@bvusaex.bestvoipusa.local>
	<5133D77F.6090402@anutel.ca>
Message-ID: <20130304205527.5c8a411f@multivac>

El Sun, 03 Mar 2013 18:06:39 -0500
Blagoja <blagojaanusev at anutel.ca> escribi?:

> En-Respuesta-A:
> <CEEE718799582B47A55D7897DCB264FC280153 at bvusaex.bestvoipusa.local

> Asunto:
> [Spce-user] non existing right pvar



Please don't answer a mail to create a new thread as you break threading in
mail clients. use the "new mail" button.

Thanks


Jon




From csaroli at isysp.com  Mon Mar  4 16:25:12 2013
From: csaroli at isysp.com (Christopher Saroli)
Date: Mon, 4 Mar 2013 21:25:12 +0000
Subject: [Spce-user]  Billing Software
Message-ID: <EE7149FDD60B3045A4377F05AAFFF3F118333057@ISS-Server.ISS.local>

Anyone know of a decent 3rd party billing system that works well with SPCE.  Preferably open-source.

Any suggestions would be appreciated, thinking of writing my own if I cannot find anything.

Thanks

---------------------------------
Christopher Saroli
csaroli at isysp.com


From gavin.sweet at skyracktelecom.com  Mon Mar  4 16:35:27 2013
From: gavin.sweet at skyracktelecom.com (Gavin Sweet)
Date: Mon, 4 Mar 2013 21:35:27 -0000
Subject: [Spce-user] Billing Software
In-Reply-To: <EE7149FDD60B3045A4377F05AAFFF3F118333057@ISS-Server.ISS.local>
References: <EE7149FDD60B3045A4377F05AAFFF3F118333057@ISS-Server.ISS.local>
Message-ID: <015901ce1920$3072f090$9158d1b0$@sweet@skyracktelecom.com>

Hi Christopher - Try jbilling. I'm partisan as we happen to act as
professional services partners for jbilling, but we do also run our own
hosted voice operations from a combination of open source software that
includes both sipwise and jbilling. It is actually designed to deliver
telecoms billing with mediation and rating.

Cheers
Gavin



> -----Original Message-----
> From: spce-user-bounces at lists.sipwise.com [mailto:spce-user-
> bounces at lists.sipwise.com] On Behalf Of Christopher Saroli
> Sent: 04 March 2013 21:25
> To: spce-user at lists.sipwise.com
> Subject: [Spce-user] Billing Software
> 
> Anyone know of a decent 3rd party billing system that works well with
> SPCE.  Preferably open-source.
> 
> Any suggestions would be appreciated, thinking of writing my own if I
> cannot find anything.
> 
> Thanks
> 
> ---------------------------------
> Christopher Saroli
> csaroli at isysp.com
> 
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
> -----
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2013.0.2899 / Virus Database: 2641/6145 - Release Date:
> 03/03/13




From orenyny at gmail.com  Mon Mar  4 19:43:02 2013
From: orenyny at gmail.com (Oren Yehezkely)
Date: Mon, 4 Mar 2013 19:43:02 -0500
Subject: [Spce-user] Billing Software
In-Reply-To: <513513b2.41ae0e0a.5d09.0c12SMTPIN_ADDED_BROKEN@mx.google.com>
References: <EE7149FDD60B3045A4377F05AAFFF3F118333057@ISS-Server.ISS.local>
	<513513b2.41ae0e0a.5d09.0c12SMTPIN_ADDED_BROKEN@mx.google.com>
Message-ID: <CAL+WrhJpsp0pvUAYL-rgxt09zQ-SALqN9joEbY4DnQ18bObWkw@mail.gmail.com>

Christopher,

If you are writing your own, I may be interested to participate.

The market for open source does not look very good.
jBilling seems to be open, only by name. I think they found it to be a
good marketing tool.
You can download the code but what to do after that?

Here are a few projects I have found in my searches . I will be glad
to learn about others you have found.

1. http://www.agileco.com/ - Agilebill - This project seems to be
dead. There is a maintainer, but the founder seems to have abandoned
the project with no one else in his place.
2. www.opencdrrate.org - OpenCDR - They seem to have made some
significant progress in the last year.
It looks like they are active in adding more features and fixing bugs.
Even though small they look vibrant.
3. http://freeside.biz/freeside/ - Freeside - this is an active
company as far as I can tell. They also have some kind of CRM. Take a
look at the demo they have online.
They integrate with multiple platforms.

Regards,

Oren


On Mon, Mar 4, 2013 at 4:35 PM, Gavin Sweet
<gavin.sweet at skyracktelecom.com> wrote:
> Hi Christopher - Try jbilling. I'm partisan as we happen to act as
> professional services partners for jbilling, but we do also run our own
> hosted voice operations from a combination of open source software that
> includes both sipwise and jbilling. It is actually designed to deliver
> telecoms billing with mediation and rating.
>
> Cheers
> Gavin
>
>
>
>> -----Original Message-----
>> From: spce-user-bounces at lists.sipwise.com [mailto:spce-user-
>> bounces at lists.sipwise.com] On Behalf Of Christopher Saroli
>> Sent: 04 March 2013 21:25
>> To: spce-user at lists.sipwise.com
>> Subject: [Spce-user] Billing Software
>>
>> Anyone know of a decent 3rd party billing system that works well with
>> SPCE.  Preferably open-source.
>>
>> Any suggestions would be appreciated, thinking of writing my own if I
>> cannot find anything.
>>
>> Thanks
>>
>> ---------------------------------
>> Christopher Saroli
>> csaroli at isysp.com
>>
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/listinfo/spce-user
>> -----
>> No virus found in this message.
>> Checked by AVG - www.avg.com
>> Version: 2013.0.2899 / Virus Database: 2641/6145 - Release Date:
>> 03/03/13
>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user



From skchopperguy at gmail.com  Mon Mar  4 20:50:11 2013
From: skchopperguy at gmail.com (Skyler)
Date: Mon, 04 Mar 2013 17:50:11 -0800
Subject: [Spce-user] Billing Software
In-Reply-To: <CAL+WrhJpsp0pvUAYL-rgxt09zQ-SALqN9joEbY4DnQ18bObWkw@mail.gmail.com>
References: <EE7149FDD60B3045A4377F05AAFFF3F118333057@ISS-Server.ISS.local>
	<513513b2.41ae0e0a.5d09.0c12SMTPIN_ADDED_BROKEN@mx.google.com>
	<CAL+WrhJpsp0pvUAYL-rgxt09zQ-SALqN9joEbY4DnQ18bObWkw@mail.gmail.com>
Message-ID: <51354F53.9080707@gmail.com>

Blesta looks pretty good. Has an open back-end to create modules. ie: 
SOAP. They have a v3.0 coming out, not sure when though. I'd be into 
collaboration.

Skyler

On 3/4/2013 4:43 PM, Oren Yehezkely wrote:
> Christopher,
>
> If you are writing your own, I may be interested to participate.
>
> The market for open source does not look very good.
> jBilling seems to be open, only by name. I think they found it to be a
> good marketing tool.
> You can download the code but what to do after that?
>
> Here are a few projects I have found in my searches . I will be glad
> to learn about others you have found.
>
> 1. http://www.agileco.com/ - Agilebill - This project seems to be
> dead. There is a maintainer, but the founder seems to have abandoned
> the project with no one else in his place.
> 2. www.opencdrrate.org - OpenCDR - They seem to have made some
> significant progress in the last year.
> It looks like they are active in adding more features and fixing bugs.
> Even though small they look vibrant.
> 3. http://freeside.biz/freeside/ - Freeside - this is an active
> company as far as I can tell. They also have some kind of CRM. Take a
> look at the demo they have online.
> They integrate with multiple platforms.
>
> Regards,
>
> Oren
>
>
> On Mon, Mar 4, 2013 at 4:35 PM, Gavin Sweet
> <gavin.sweet at skyracktelecom.com> wrote:
>> Hi Christopher - Try jbilling. I'm partisan as we happen to act as
>> professional services partners for jbilling, but we do also run our own
>> hosted voice operations from a combination of open source software that
>> includes both sipwise and jbilling. It is actually designed to deliver
>> telecoms billing with mediation and rating.
>>
>> Cheers
>> Gavin
>>
>>
>>
>>> -----Original Message-----
>>> From: spce-user-bounces at lists.sipwise.com [mailto:spce-user-
>>> bounces at lists.sipwise.com] On Behalf Of Christopher Saroli
>>> Sent: 04 March 2013 21:25
>>> To: spce-user at lists.sipwise.com
>>> Subject: [Spce-user] Billing Software
>>>
>>> Anyone know of a decent 3rd party billing system that works well with
>>> SPCE.  Preferably open-source.
>>>
>>> Any suggestions would be appreciated, thinking of writing my own if I
>>> cannot find anything.
>>>
>>> Thanks
>>>
>>> ---------------------------------
>>> Christopher Saroli
>>> csaroli at isysp.com
>>>
>>> _______________________________________________
>>> Spce-user mailing list
>>> Spce-user at lists.sipwise.com
>>> http://lists.sipwise.com/listinfo/spce-user
>>> -----
>>> No virus found in this message.
>>> Checked by AVG - www.avg.com
>>> Version: 2013.0.2899 / Virus Database: 2641/6145 - Release Date:
>>> 03/03/13
>>
>>
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/listinfo/spce-user
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
>



From orenyny at gmail.com  Mon Mar  4 21:36:39 2013
From: orenyny at gmail.com (Oren Yehezkely)
Date: Mon, 4 Mar 2013 21:36:39 -0500
Subject: [Spce-user] Billing Software
In-Reply-To: <51354F53.9080707@gmail.com>
References: <EE7149FDD60B3045A4377F05AAFFF3F118333057@ISS-Server.ISS.local>
	<513513b2.41ae0e0a.5d09.0c12SMTPIN_ADDED_BROKEN@mx.google.com>
	<CAL+WrhJpsp0pvUAYL-rgxt09zQ-SALqN9joEbY4DnQ18bObWkw@mail.gmail.com>
	<51354F53.9080707@gmail.com>
Message-ID: <CAL+WrhL3_g3D4CNZ5DMix4XLYqTfOhbO6g5DEN0EiQ1nO622wA@mail.gmail.com>

Yes, it does look good.
But I think version 3.0 is a long time in the making.

Not sure if it was designed for VoIP.

Oren


On Mon, Mar 4, 2013 at 8:50 PM, Skyler <skchopperguy at gmail.com> wrote:
> Blesta looks pretty good. Has an open back-end to create modules. ie: SOAP.
> They have a v3.0 coming out, not sure when though. I'd be into
> collaboration.
>
> Skyler
>
>
> On 3/4/2013 4:43 PM, Oren Yehezkely wrote:
>>
>> Christopher,
>>
>> If you are writing your own, I may be interested to participate.
>>
>> The market for open source does not look very good.
>> jBilling seems to be open, only by name. I think they found it to be a
>> good marketing tool.
>> You can download the code but what to do after that?
>>
>> Here are a few projects I have found in my searches . I will be glad
>> to learn about others you have found.
>>
>> 1. http://www.agileco.com/ - Agilebill - This project seems to be
>> dead. There is a maintainer, but the founder seems to have abandoned
>> the project with no one else in his place.
>> 2. www.opencdrrate.org - OpenCDR - They seem to have made some
>> significant progress in the last year.
>> It looks like they are active in adding more features and fixing bugs.
>> Even though small they look vibrant.
>> 3. http://freeside.biz/freeside/ - Freeside - this is an active
>> company as far as I can tell. They also have some kind of CRM. Take a
>> look at the demo they have online.
>> They integrate with multiple platforms.
>>
>> Regards,
>>
>> Oren
>>
>>
>> On Mon, Mar 4, 2013 at 4:35 PM, Gavin Sweet
>> <gavin.sweet at skyracktelecom.com> wrote:
>>>
>>> Hi Christopher - Try jbilling. I'm partisan as we happen to act as
>>> professional services partners for jbilling, but we do also run our own
>>> hosted voice operations from a combination of open source software that
>>> includes both sipwise and jbilling. It is actually designed to deliver
>>> telecoms billing with mediation and rating.
>>>
>>> Cheers
>>> Gavin
>>>
>>>
>>>
>>>> -----Original Message-----
>>>> From: spce-user-bounces at lists.sipwise.com [mailto:spce-user-
>>>> bounces at lists.sipwise.com] On Behalf Of Christopher Saroli
>>>> Sent: 04 March 2013 21:25
>>>> To: spce-user at lists.sipwise.com
>>>> Subject: [Spce-user] Billing Software
>>>>
>>>> Anyone know of a decent 3rd party billing system that works well with
>>>> SPCE.  Preferably open-source.
>>>>
>>>> Any suggestions would be appreciated, thinking of writing my own if I
>>>> cannot find anything.
>>>>
>>>> Thanks
>>>>
>>>> ---------------------------------
>>>> Christopher Saroli
>>>> csaroli at isysp.com
>>>>
>>>> _______________________________________________
>>>> Spce-user mailing list
>>>> Spce-user at lists.sipwise.com
>>>> http://lists.sipwise.com/listinfo/spce-user
>>>> -----
>>>> No virus found in this message.
>>>> Checked by AVG - www.avg.com
>>>> Version: 2013.0.2899 / Virus Database: 2641/6145 - Release Date:
>>>> 03/03/13
>>>
>>>
>>>
>>> _______________________________________________
>>> Spce-user mailing list
>>> Spce-user at lists.sipwise.com
>>> http://lists.sipwise.com/listinfo/spce-user
>>
>>
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/listinfo/spce-user
>>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user



From skchopperguy at gmail.com  Mon Mar  4 22:23:09 2013
From: skchopperguy at gmail.com (Skyler)
Date: Mon, 04 Mar 2013 19:23:09 -0800
Subject: [Spce-user] Billing Software
In-Reply-To: <CAL+WrhL3_g3D4CNZ5DMix4XLYqTfOhbO6g5DEN0EiQ1nO622wA@mail.gmail.com>
References: <EE7149FDD60B3045A4377F05AAFFF3F118333057@ISS-Server.ISS.local>
	<513513b2.41ae0e0a.5d09.0c12SMTPIN_ADDED_BROKEN@mx.google.com>
	<CAL+WrhJpsp0pvUAYL-rgxt09zQ-SALqN9joEbY4DnQ18bObWkw@mail.gmail.com>
	<51354F53.9080707@gmail.com>
	<CAL+WrhL3_g3D4CNZ5DMix4XLYqTfOhbO6g5DEN0EiQ1nO622wA@mail.gmail.com>
Message-ID: <5135651D.5060501@gmail.com>

I think it was more designed for hosting. The trial looks like it can 
make service packages and those packages tie-into modules. I looked at 
the modules in there and should be straight forward to create an SPCE 
module. Looks like there is cron also which could be an idea for 
querying SPCE by external_id and building invoice(s). Maybe.

Skyler

On 3/4/2013 6:36 PM, Oren Yehezkely wrote:
> Yes, it does look good.
> But I think version 3.0 is a long time in the making.
>
> Not sure if it was designed for VoIP.
>
> Oren
>
>
> On Mon, Mar 4, 2013 at 8:50 PM, Skyler <skchopperguy at gmail.com> wrote:
>> Blesta looks pretty good. Has an open back-end to create modules. ie: SOAP.
>> They have a v3.0 coming out, not sure when though. I'd be into
>> collaboration.
>>
>> Skyler
>>
>>
>> On 3/4/2013 4:43 PM, Oren Yehezkely wrote:
>>>
>>> Christopher,
>>>
>>> If you are writing your own, I may be interested to participate.
>>>
>>> The market for open source does not look very good.
>>> jBilling seems to be open, only by name. I think they found it to be a
>>> good marketing tool.
>>> You can download the code but what to do after that?
>>>
>>> Here are a few projects I have found in my searches . I will be glad
>>> to learn about others you have found.
>>>
>>> 1. http://www.agileco.com/ - Agilebill - This project seems to be
>>> dead. There is a maintainer, but the founder seems to have abandoned
>>> the project with no one else in his place.
>>> 2. www.opencdrrate.org - OpenCDR - They seem to have made some
>>> significant progress in the last year.
>>> It looks like they are active in adding more features and fixing bugs.
>>> Even though small they look vibrant.
>>> 3. http://freeside.biz/freeside/ - Freeside - this is an active
>>> company as far as I can tell. They also have some kind of CRM. Take a
>>> look at the demo they have online.
>>> They integrate with multiple platforms.
>>>
>>> Regards,
>>>
>>> Oren
>>>
>>>
>>> On Mon, Mar 4, 2013 at 4:35 PM, Gavin Sweet
>>> <gavin.sweet at skyracktelecom.com> wrote:
>>>>
>>>> Hi Christopher - Try jbilling. I'm partisan as we happen to act as
>>>> professional services partners for jbilling, but we do also run our own
>>>> hosted voice operations from a combination of open source software that
>>>> includes both sipwise and jbilling. It is actually designed to deliver
>>>> telecoms billing with mediation and rating.
>>>>
>>>> Cheers
>>>> Gavin
>>>>
>>>>
>>>>
>>>>> -----Original Message-----
>>>>> From: spce-user-bounces at lists.sipwise.com [mailto:spce-user-
>>>>> bounces at lists.sipwise.com] On Behalf Of Christopher Saroli
>>>>> Sent: 04 March 2013 21:25
>>>>> To: spce-user at lists.sipwise.com
>>>>> Subject: [Spce-user] Billing Software
>>>>>
>>>>> Anyone know of a decent 3rd party billing system that works well with
>>>>> SPCE.  Preferably open-source.
>>>>>
>>>>> Any suggestions would be appreciated, thinking of writing my own if I
>>>>> cannot find anything.
>>>>>
>>>>> Thanks
>>>>>
>>>>> ---------------------------------
>>>>> Christopher Saroli
>>>>> csaroli at isysp.com
>>>>>
>>>>> _______________________________________________
>>>>> Spce-user mailing list
>>>>> Spce-user at lists.sipwise.com
>>>>> http://lists.sipwise.com/listinfo/spce-user
>>>>> -----
>>>>> No virus found in this message.
>>>>> Checked by AVG - www.avg.com
>>>>> Version: 2013.0.2899 / Virus Database: 2641/6145 - Release Date:
>>>>> 03/03/13
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Spce-user mailing list
>>>> Spce-user at lists.sipwise.com
>>>> http://lists.sipwise.com/listinfo/spce-user
>>>
>>>
>>> _______________________________________________
>>> Spce-user mailing list
>>> Spce-user at lists.sipwise.com
>>> http://lists.sipwise.com/listinfo/spce-user
>>>
>>
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/listinfo/spce-user



From jbonilla at sipwise.com  Tue Mar  5 01:59:53 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Tue, 5 Mar 2013 07:59:53 +0100
Subject: [Spce-user] Upper Registration to Asterisk - transfers not
 working
In-Reply-To: <CA+9+E=WU=Sa9NTZ=HJrzLXZ+VnTON_--+9fBaHtivs+Lfh7Vwg@mail.gmail.com>
References: <CA+9+E=X_qiRReX=c_XubTD89HEV2HApxFw1yRDgDa2nW0o_SZg@mail.gmail.com>
	<20130215213405.58b3a93a@quenya>
	<CA+9+E=Wnubt_Wy+xHjk0C6kw0XQ8tc2-agbFvkevcTpYJEqg4Q@mail.gmail.com>
	<720785848-1361030221-cardhu_decombobulator_blackberry.rim.net-863913820-@b16.c11.bise7.blackberry>
	<CA+9+E=Xh_1QHALsUzOhnUOK7q4uD8ZyoLwHoEs2R-zHh=7Ox5g@mail.gmail.com>
	<CA+9+E=U+N0GJY7QOmvTYgW5fVsetEZ_e2rYTOtp_EWeH5U7HTg@mail.gmail.com>
	<CA+9+E=U215fg8=y_ff46U2pyUxB4imXi1-i1Sx3bMbtECZ+28A@mail.gmail.com>
	<51224FFD.4020408@sipwise.com> <51225420.5000607@sipwise.com>
	<CA+9+E=U-t1paf0KW+H-vHkOwGFBkbn1=8NFd5YUPdEwm_xyZsg@mail.gmail.com>
	<5122598B.1060703@sipwise.com>
	<CA+9+E=XeYZRHuBQkhsHBTMCarq2bOdX9c+93suqyCCb-Gj+3Ag@mail.gmail.com>
	<20130226152313.2d977e1d@quenya>
	<CA+9+E=WQUrn1HbeekF2fchdZ+7kX54Snphe0QY6W_x4oXN6yyA@mail.gmail.com>
	<CA+9+E=XRB-hPxK1ZTBU5ebhnaq-y0wS0=B=8ZaeVCheBC6+tWg@mail.gmail.com>
	<CA+9+E=WU=Sa9NTZ=HJrzLXZ+VnTON_--+9fBaHtivs+Lfh7Vwg@mail.gmail.com>
Message-ID: <20130305075953.50b44f18@multivac>

El Mon, 4 Mar 2013 08:59:03 +0200
Theo <axessofficetheo at gmail.com> escribi?:

> Hi
> 
> Not sure if I am breaking protocol by bumping - if I did - apologies. I am
> almost ready to put this in as our SBC upper registration solution (more to
> follow after that I hope :-) - but need to resolve the attended transfer
> problem. Any ideas on this? Sip Header filtering is off altogether (does
> that have any caveats?). Transfer working fine now for all phones tested,
> but attended transfer not.
>

ngrep-sip?




From joan at megatro.com  Tue Mar  5 07:51:11 2013
From: joan at megatro.com (Joan Jou)
Date: Tue, 05 Mar 2013 13:51:11 +0100
Subject: [Spce-user] Statistics
In-Reply-To: <51349A0F.7050403@sipwise.com>
References: <5134684D.203@megatro.com>
	<CEEE718799582B47A55D7897DCB264FC280176@bvusaex.bestvoipusa.local>
	<51349462.1080701@megatro.com>
	<CEEE718799582B47A55D7897DCB264FC280177@bvusaex.bestvoipusa.local>
	<51349A0F.7050403@sipwise.com>
Message-ID: <5135EA3F.6040602@megatro.com>

Hi,
There are not files in rrd directori, only localhost and sip directories.

This is log file content:
ar  4 14:36:40 sip oss: 46.253.41.146 - - [04/Mar/2013:14:36:40 +0000] 
"GET / HTTP/1.1" 200 2803 "https://185.14.187.173:1443/dashboard/system" 
"Mozilla/5.0 (Windows NT 6.1; WOW64; rv:18.0) Gecko/20100101 
Firefox/18.0 FirePHP/0.7.1"
Mar  4 14:36:55 sip apache2: [info] [client 46.253.41.146] (70007)The 
timeout specified has expired: SSL input filter read failed.
Mar  4 14:36:55 sip apache2: [info] [client 46.253.41.146] Connection 
closed to child 7 with standard shutdown (server myserver:443)
Mar  4 14:37:03 sip apache2: [info] [client 46.253.41.146] Connection to 
child 5 established (server myserver:443)
Mar  4 14:37:03 sip apache2: [info] Seeding PRNG with 656 bytes of entropy
Mar  4 14:37:03 sip apache2: [info] Initial (No.1) HTTPS request 
received for child 5 (server myserver:443)
Mar  4 14:37:04 sip oss: 46.253.41.146 - - [04/Mar/2013:14:37:03 +0000] 
"GET /dashboard HTTP/1.1" 302 963 "https://185.14.187.173:1443/" 
"Mozilla/5.0 (Windows NT 6.1; WOW64; rv:18.0) Gecko/20100101 
Firefox/18.0 FirePHP/0.7.1"
Mar  4 14:37:04 sip apache2: [info] Subsequent (No.2) HTTPS request 
received for child 5 (server myserver:443)
Mar  4 14:37:04 sip oss: 46.253.41.146 - - [04/Mar/2013:14:37:04 +0000] 
"GET /dashboard/system HTTP/1.1" 200 3634 "https://185.14.187.173:1443/" 
"Mozilla/5.0 (Windows NT 6.1; WOW64; rv:18.0) Gecko/20100101 
Firefox/18.0 FirePHP/0.7.1"
Mar  4 14:37:19 sip apache2: [info] [client 46.253.41.146] (70007)The 
timeout specified has expired: SSL input filter read failed.
Mar  4 14:37:19 sip apache2: [info] [client 46.253.41.146] Connection 
closed to child 5 with standard shutdown (server myserver:443)

Joan,

Al 04/03/2013 13:56, En/na Andreas Granig ha escrit:
> Hi,
>
> Check /var/log/ngcp/oss.log to see if there are any errors. Maybe it 
> doesn't find the rrd files in /var/lib/collectd/rrd, maybe because 
> collectd is not running?
>
> Andreas
>
> On 03/04/2013 01:41 PM, Kevin Masse wrote:
>> Ok, I will step away from this one.  I do not have additional 
>> suggestions.
>>
>> Kevin
>>
>> *From:*Joan Jou [mailto:joan at megatro.com]
>> *Sent:* Monday, March 04, 2013 7:33 AM
>> *To:* Kevin Masse
>> *Subject:* Re: [Spce-user] Statistics
>>
>> Good morning,
>> I try it whith  Explorer, Chrome and Firefox.
>>
>> Joan,
>>
>> Al 04/03/2013 13:15, En/na Kevin Masse ha escrit:
>>
>>     Good morning Joan, if you are using Internet Explorer try using
>>     Chrome or Firefox.
>>
>>     I have noticed that IE does have this issue with the stats page.
>>
>>     Kevin
>>
>>     *From:*spce-user-bounces at lists.sipwise.com
>>     <mailto:spce-user-bounces at lists.sipwise.com>
>>     [mailto:spce-user-bounces at lists.sipwise.com] *On Behalf Of *Joan Jou
>>     *Sent:* Monday, March 04, 2013 4:24 AM
>>     *To:* spce-user at lists.sipwise.com 
>> <mailto:spce-user at lists.sipwise.com>
>>     *Subject:* [Spce-user] Statistics
>>
>>     Hi,
>>     On a new intallation.
>>     Everythink looks running but Information/statistics  stay 
>> "loading...".
>>
>>     Thank you,
>>
>>
>>     --
>>
>>
>>
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/listinfo/spce-user
>>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130305/f63dc772/attachment.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signatura1.gif
Type: image/gif
Size: 13564 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130305/f63dc772/attachment.gif>

From matthew at tenacit.net  Tue Mar  5 08:34:18 2013
From: matthew at tenacit.net (Matthew Ogden)
Date: Tue, 5 Mar 2013 15:34:18 +0200
Subject: [Spce-user] rewrite rules
Message-ID: <8e982f2ac89360982fd9e03982d5c8d2@mail.gmail.com>

Hi



Its easy enough with a rewrite rule to replace something, but does it
support ?adding? one number to another



For example a replace for 555-555-55XX can make 3201 become 555-555-5501.
But what if you wanted 3201 to become 555-555-5506 and 3202 to become
555-555-5507?



Regards





*Matthew Ogden*

Management

TenacIT







*Strategic IT Consulting *?* Advanced Networking *? *Virtualisation* **

*Custom Development *?* **Hosting *?* **Syspro Support  *? *MS Licensing* **

National Tel: 041 10 10 100 | Cell: 084 205 4445 | Email:
matthew at tenacit.net

CT Tel: 021 201 0333 | Skype Name: matthew.ogden | Web:
http://www.tenacit.net
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130305/b171367d/attachment.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.jpg
Type: image/jpeg
Size: 6077 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130305/b171367d/attachment.jpg>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image002.jpg
Type: image/jpeg
Size: 1240 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130305/b171367d/attachment-0001.jpg>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image003.png
Type: image/png
Size: 4650 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130305/b171367d/attachment.png>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image004.jpg
Type: image/jpeg
Size: 1232 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130305/b171367d/attachment-0002.jpg>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image005.jpg
Type: image/jpeg
Size: 2254 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130305/b171367d/attachment-0003.jpg>

From sn at fest.dk  Tue Mar  5 08:47:19 2013
From: sn at fest.dk (Sylvester Nielsen)
Date: Tue, 5 Mar 2013 13:47:19 +0000
Subject: [Spce-user] rewrite rules
In-Reply-To: <8e982f2ac89360982fd9e03982d5c8d2@mail.gmail.com>
References: <8e982f2ac89360982fd9e03982d5c8d2@mail.gmail.com>
Message-ID: <E50575DD48D9B743925AC3BCE6310A4B2433E1AE@DB3PRD0710MB355.eurprd07.prod.outlook.com>

As far as I know you can't add or subtract in the replacement pattern or do any math for that matter.
However, you can create some clever match/replace to achieve a rewrite as you describe.
Personally I don't see the use of it though.

Med venlig hilsen

   Sylvester Nielsen

---------------------------------------------------------------

Fest Data A/S
Generatorvej 8D, 1.th
2730 Herlev

e-mail : sn at fest.dk<mailto:sn at fest.dk>

Tlf:     +45 4582 7389
Fax:   +45 4582 7387
Dir:    +45 6996 2011

http://www.festdata.dk

[Beskrivelse: Description: Description: Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse: cid:image003.jpg at 01CBCE06.BAABD4A0]




From: spce-user-bounces at lists.sipwise.com [mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of Matthew Ogden
Sent: 5. marts 2013 14:34
To: spce-user at lists.sipwise.com
Subject: [Spce-user] rewrite rules

Hi

Its easy enough with a rewrite rule to replace something, but does it support "adding" one number to another

For example a replace for 555-555-55XX can make 3201 become 555-555-5501. But what if you wanted 3201 to become 555-555-5506 and 3202 to become 555-555-5507?

Regards

[cid:image007.jpg at 01CE19B0.75ABB990]

Matthew Ogden
Management
TenacIT



Strategic IT Consulting * Advanced Networking * Virtualisation
Custom Development * Hosting * Syspro Support  * MS Licensing
National Tel: 041 10 10 100 | Cell: 084 205 4445 | Email: matthew at tenacit.net<mailto:matthew at tenacit.net>
CT Tel: 021 201 0333 | Skype Name: matthew.ogden | Web: http://www.tenacit.net<http://www.tenacit.net/>

[cid:image008.jpg at 01CE19B0.75ABB990] [cid:image009.png at 01CE19B0.75ABB990]  [cid:image010.jpg at 01CE19B0.75ABB990]  [cid:image011.jpg at 01CE19B0.75ABB990]


________________________________
Denne mail er blevet scannet af fest.dk ApS<http://www.fest.dk/> og indeholder ikke virus!
________________________________
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130305/c1578e8e/attachment.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image006.jpg
Type: image/jpeg
Size: 11834 bytes
Desc: image006.jpg
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130305/c1578e8e/attachment.jpg>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image007.jpg
Type: image/jpeg
Size: 6077 bytes
Desc: image007.jpg
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130305/c1578e8e/attachment-0001.jpg>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image008.jpg
Type: image/jpeg
Size: 1240 bytes
Desc: image008.jpg
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130305/c1578e8e/attachment-0002.jpg>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image009.png
Type: image/png
Size: 4650 bytes
Desc: image009.png
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130305/c1578e8e/attachment.png>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image010.jpg
Type: image/jpeg
Size: 1232 bytes
Desc: image010.jpg
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130305/c1578e8e/attachment-0003.jpg>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image011.jpg
Type: image/jpeg
Size: 2254 bytes
Desc: image011.jpg
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130305/c1578e8e/attachment-0004.jpg>

From dgrotti at sipwise.com  Tue Mar  5 10:14:17 2013
From: dgrotti at sipwise.com (Daniel Grotti)
Date: Tue, 05 Mar 2013 16:14:17 +0100
Subject: [Spce-user] rewrite rules
In-Reply-To: <8e982f2ac89360982fd9e03982d5c8d2@mail.gmail.com>
References: <8e982f2ac89360982fd9e03982d5c8d2@mail.gmail.com>
Message-ID: <51360BC9.9000401@sipwise.com>

Hi Matthew,

rewrite rules are based on Regular expression so you can do everything
you want.
Of course you can replace, remove or adding.

Daniel
 


On 03/05/2013 02:34 PM, Matthew Ogden wrote:
>
> Hi
>
>  
>
> Its easy enough with a rewrite rule to replace something, but does it
> support "adding" one number to another
>
>  
>
> For example a replace for 555-555-55XX can make 3201 become
> 555-555-5501. But what if you wanted 3201 to become 555-555-5506 and
> 3202 to become 555-555-5507?
>
>  
>
> Regards
>
>  
>
>  
>
> *Matthew Ogden*
>
> Management
>
> TenacIT
>
>  
>
>  
>
>  
>
> *Strategic IT Consulting *.*Advanced Networking *. *Virtualisation***
>
> *Custom Development *.***Hosting *.***Syspro Support  *. *MS Licensing***
>
> National Tel: 041 10 10 100 | Cell: 084 205 4445 | Email:
> matthew at tenacit.net <mailto:matthew at tenacit.net> 
>
> CT Tel: 021 201 0333 | Skype Name: matthew.ogden | Web:
> http://www.tenacit.net <http://www.tenacit.net/>
>
>  
>
>    
>
>  
>
>  
>
>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130305/5b1fe192/attachment.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: image/jpeg
Size: 6077 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130305/5b1fe192/attachment.jpe>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: image/jpeg
Size: 1240 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130305/5b1fe192/attachment-0001.jpe>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: image/png
Size: 4650 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130305/5b1fe192/attachment.png>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: image/jpeg
Size: 1232 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130305/5b1fe192/attachment-0002.jpe>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: image/jpeg
Size: 2254 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130305/5b1fe192/attachment-0003.jpe>

From axessofficetheo at gmail.com  Wed Mar  6 02:41:52 2013
From: axessofficetheo at gmail.com (Theo)
Date: Wed, 6 Mar 2013 09:41:52 +0200
Subject: [Spce-user] Upper Registration to Asterisk - transfers not
	working
In-Reply-To: <20130305075953.50b44f18@multivac>
References: <CA+9+E=X_qiRReX=c_XubTD89HEV2HApxFw1yRDgDa2nW0o_SZg@mail.gmail.com>
	<20130215213405.58b3a93a@quenya>
	<CA+9+E=Wnubt_Wy+xHjk0C6kw0XQ8tc2-agbFvkevcTpYJEqg4Q@mail.gmail.com>
	<720785848-1361030221-cardhu_decombobulator_blackberry.rim.net-863913820-@b16.c11.bise7.blackberry>
	<CA+9+E=Xh_1QHALsUzOhnUOK7q4uD8ZyoLwHoEs2R-zHh=7Ox5g@mail.gmail.com>
	<CA+9+E=U+N0GJY7QOmvTYgW5fVsetEZ_e2rYTOtp_EWeH5U7HTg@mail.gmail.com>
	<CA+9+E=U215fg8=y_ff46U2pyUxB4imXi1-i1Sx3bMbtECZ+28A@mail.gmail.com>
	<51224FFD.4020408@sipwise.com> <51225420.5000607@sipwise.com>
	<CA+9+E=U-t1paf0KW+H-vHkOwGFBkbn1=8NFd5YUPdEwm_xyZsg@mail.gmail.com>
	<5122598B.1060703@sipwise.com>
	<CA+9+E=XeYZRHuBQkhsHBTMCarq2bOdX9c+93suqyCCb-Gj+3Ag@mail.gmail.com>
	<20130226152313.2d977e1d@quenya>
	<CA+9+E=WQUrn1HbeekF2fchdZ+7kX54Snphe0QY6W_x4oXN6yyA@mail.gmail.com>
	<CA+9+E=XRB-hPxK1ZTBU5ebhnaq-y0wS0=B=8ZaeVCheBC6+tWg@mail.gmail.com>
	<CA+9+E=WU=Sa9NTZ=HJrzLXZ+VnTON_--+9fBaHtivs+Lfh7Vwg@mail.gmail.com>
	<20130305075953.50b44f18@multivac>
Message-ID: <CA+9+E=UKknSzZGHFp5qi2cLx=NdaZk879jQPQi_BvEY7ce-jAg@mail.gmail.com>

Hi

ngrep-sip gives nothing - Here's the process on a SNOM phone.

Call comes in. You put it on hold. You dial another extension and announce
the call. All that shows up with ngrep-sip. Once the call is announced you
press the transfer button. Nothing is recorded for that with ngrep sip
whatsoever, which is obviously why it doesn't work :-(.

Any ideas on that one?

On Tue, Mar 5, 2013 at 8:59 AM, Jon Bonilla <jbonilla at sipwise.com> wrote:

> El Mon, 4 Mar 2013 08:59:03 +0200
> Theo <axessofficetheo at gmail.com> escribi?:
>
> > Hi
> >
> > Not sure if I am breaking protocol by bumping - if I did - apologies. I
> am
> > almost ready to put this in as our SBC upper registration solution (more
> to
> > follow after that I hope :-) - but need to resolve the attended transfer
> > problem. Any ideas on this? Sip Header filtering is off altogether (does
> > that have any caveats?). Transfer working fine now for all phones tested,
> > but attended transfer not.
> >
>
> ngrep-sip?
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130306/e30f9de2/attachment.html>

From axessofficetheo at gmail.com  Wed Mar  6 04:41:29 2013
From: axessofficetheo at gmail.com (Theo)
Date: Wed, 6 Mar 2013 11:41:29 +0200
Subject: [Spce-user] Upper Registration to Asterisk - transfers not
	working
In-Reply-To: <CA+9+E=UKknSzZGHFp5qi2cLx=NdaZk879jQPQi_BvEY7ce-jAg@mail.gmail.com>
References: <CA+9+E=X_qiRReX=c_XubTD89HEV2HApxFw1yRDgDa2nW0o_SZg@mail.gmail.com>
	<20130215213405.58b3a93a@quenya>
	<CA+9+E=Wnubt_Wy+xHjk0C6kw0XQ8tc2-agbFvkevcTpYJEqg4Q@mail.gmail.com>
	<720785848-1361030221-cardhu_decombobulator_blackberry.rim.net-863913820-@b16.c11.bise7.blackberry>
	<CA+9+E=Xh_1QHALsUzOhnUOK7q4uD8ZyoLwHoEs2R-zHh=7Ox5g@mail.gmail.com>
	<CA+9+E=U+N0GJY7QOmvTYgW5fVsetEZ_e2rYTOtp_EWeH5U7HTg@mail.gmail.com>
	<CA+9+E=U215fg8=y_ff46U2pyUxB4imXi1-i1Sx3bMbtECZ+28A@mail.gmail.com>
	<51224FFD.4020408@sipwise.com> <51225420.5000607@sipwise.com>
	<CA+9+E=U-t1paf0KW+H-vHkOwGFBkbn1=8NFd5YUPdEwm_xyZsg@mail.gmail.com>
	<5122598B.1060703@sipwise.com>
	<CA+9+E=XeYZRHuBQkhsHBTMCarq2bOdX9c+93suqyCCb-Gj+3Ag@mail.gmail.com>
	<20130226152313.2d977e1d@quenya>
	<CA+9+E=WQUrn1HbeekF2fchdZ+7kX54Snphe0QY6W_x4oXN6yyA@mail.gmail.com>
	<CA+9+E=XRB-hPxK1ZTBU5ebhnaq-y0wS0=B=8ZaeVCheBC6+tWg@mail.gmail.com>
	<CA+9+E=WU=Sa9NTZ=HJrzLXZ+VnTON_--+9fBaHtivs+Lfh7Vwg@mail.gmail.com>
	<20130305075953.50b44f18@multivac>
	<CA+9+E=UKknSzZGHFp5qi2cLx=NdaZk879jQPQi_BvEY7ce-jAg@mail.gmail.com>
Message-ID: <CA+9+E=UfxgtRPVjC_8BhcEm0kh2HSKnErhro7WrFJ-eLu8SzHA@mail.gmail.com>

A little more now - I upgraded the firmware of a SNOM phone. Following the
same process, upon pressing the transfer button I now get this:

U 2013/03/06 11:36:22.236542 41.177.66.116:22106 -> 196.41.123.113:5060
SIP/2.0 200 OK'
Via: SIP/2.0/UDP
196.41.123.113;branch=z9hG4bKaa2.bc57c1db3d0853f982c750b38e5926b2.0'
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK~IoIoamR;rport=5080'
Record-Route:
<sip:196.41.123.113;lr;r2=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
Record-Route:
<sip:127.0.0.1;r2=on;lr=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
Contact: <sip:user400 at 192.168.10.88:5060
;rinstance=7e064df24c43ccf0;transport=UDP>'
To: <sip:user400 at domain.domain.com>;tag=d37f6717'
From: <sip:2721403 at domain.domain.com
>;tag=304E3AA0-51370E09000C6324-2EE19700'
Call-ID: 51370e076928-0af4wphlmirc_b2b-1'
CSeq: 11 INVITE'
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
SUBSCRIBE'
Content-Type: application/sdp'
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri'
User-Agent: Zoiper rev.11619'
Allow-Events: presence, kpml'
Content-Length: 185'
'
v=0'
o=Z 0 3 IN IP4 192.168.10.88'
s=Z'
c=IN IP4 192.168.10.88'
t=0 0'
m=audio 8000 RTP/AVP 8 101'
a=rtpmap:8 PCMA/8000'
a=rtpmap:101 telephone-event/8000'
a=fmtp:101 0-15'
a=inactive'

#
U 2013/03/06 11:36:22.236896 127.0.0.1:5060 -> 127.0.0.1:5080
SIP/2.0 200 OK'
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK~IoIoamR;rport=5080'
Record-Route:
<sip:196.41.123.113;lr;r2=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
Record-Route:
<sip:127.0.0.1;r2=on;lr=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
Contact: <sip:user400 at 192.168.10.88:5060
;alias=41.177.66.116~22106~1;rinstance=7e064df24c43ccf0;transport=UDP>'
To: <sip:user400 at domain.domain.com>;tag=d37f6717'
From: <sip:2721403 at domain.domain.com
>;tag=304E3AA0-51370E09000C6324-2EE19700'
Call-ID: 51370e076928-0af4wphlmirc_b2b-1'
CSeq: 11 INVITE'
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
SUBSCRIBE'
Content-Type: application/sdp'
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri'
User-Agent: Zoiper rev.11619'
Allow-Events: presence, kpml'
Content-Length: 233'
P-NGCP-Src-Ip: 41.177.66.116'
P-NGCP-Src-Port: 22106'
P-NGCP-Src-Proto: udp'
P-NGCP-Src-Af: 4'
'
v=0'
o=Z 0 3 IN IP4 192.168.10.88'
s=Z'
c=IN IP4 41.177.66.116'
t=0 0'
m=audio 8000 RTP/AVP 8 101'
a=rtpmap:8 PCMA/8000'
a=rtpmap:101 telephone-event/8000'
a=fmtp:101 0-15'
a=inactive'
a=direction:active'
a=oldmediaip:192.168.10.88'

#
U 2013/03/06 11:36:22.236968 127.0.0.1:5080 -> 127.0.0.1:5060
ACK sip:user400 at 192.168.10.88:5060;alias=41.177.66.116~22106~1;rinstance=7e064df24c43ccf0;transport=UDP
SIP/2.0'
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bKTbHrca0H;rport'
From: <sip:2721403 at domain.domain.com
>;tag=304E3AA0-51370E09000C6324-2EE19700'
To: <sip:user400 at domain.domain.com>;tag=d37f6717'
CSeq: 11 ACK'
Call-ID: 51370e076928-0af4wphlmirc_b2b-1'
Contact: <sip:127.0.0.1:5080>'
Route:
<sip:127.0.0.1;r2=on;lr=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>,
<sip:196.41.123.113;lr;r2=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
Max-Forwards: 68'
Content-Length: 0'
'

##
U 2013/03/06 11:36:22.237048 196.41.123.113:5060 -> 41.177.66.116:22106
ACK sip:user400 at 192.168.10.88:5060;rinstance=7e064df24c43ccf0;transport=UDP
SIP/2.0'
Record-Route:
<sip:196.41.123.113;r2=on;lr=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
Record-Route:
<sip:127.0.0.1;r2=on;lr=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
Via: SIP/2.0/UDP
196.41.123.113;branch=z9hG4bKaa2.04ec964b43561d47cbdd4d22812c93a8.0'
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bKTbHrca0H;rport=5080'
From: <sip:2721403 at domain.domain.com
>;tag=304E3AA0-51370E09000C6324-2EE19700'
To: <sip:user400 at domain.domain.com>;tag=d37f6717'
CSeq: 11 ACK'
Call-ID: 51370e076928-0af4wphlmirc_b2b-1'
Max-Forwards: 67'
Content-Length: 0'
Contact: <sip:ngcp-lb at 196.41.123.113:5060;ngcpct='sip:127.0.0.1:5080'>'


On Wed, Mar 6, 2013 at 9:41 AM, Theo <axessofficetheo at gmail.com> wrote:

> Hi
>
> ngrep-sip gives nothing - Here's the process on a SNOM phone.
>
> Call comes in. You put it on hold. You dial another extension and announce
> the call. All that shows up with ngrep-sip. Once the call is announced you
> press the transfer button. Nothing is recorded for that with ngrep sip
> whatsoever, which is obviously why it doesn't work :-(.
>
> Any ideas on that one?
>
>
> On Tue, Mar 5, 2013 at 8:59 AM, Jon Bonilla <jbonilla at sipwise.com> wrote:
>
>> El Mon, 4 Mar 2013 08:59:03 +0200
>> Theo <axessofficetheo at gmail.com> escribi?:
>>
>> > Hi
>> >
>> > Not sure if I am breaking protocol by bumping - if I did - apologies. I
>> am
>> > almost ready to put this in as our SBC upper registration solution
>> (more to
>> > follow after that I hope :-) - but need to resolve the attended transfer
>> > problem. Any ideas on this? Sip Header filtering is off altogether (does
>> > that have any caveats?). Transfer working fine now for all phones
>> tested,
>> > but attended transfer not.
>> >
>>
>> ngrep-sip?
>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130306/b2ede2ad/attachment.html>

From axessofficetheo at gmail.com  Wed Mar  6 05:45:35 2013
From: axessofficetheo at gmail.com (Theo)
Date: Wed, 6 Mar 2013 12:45:35 +0200
Subject: [Spce-user] Upper Registration to Asterisk - transfers not
	working
In-Reply-To: <CA+9+E=UfxgtRPVjC_8BhcEm0kh2HSKnErhro7WrFJ-eLu8SzHA@mail.gmail.com>
References: <CA+9+E=X_qiRReX=c_XubTD89HEV2HApxFw1yRDgDa2nW0o_SZg@mail.gmail.com>
	<20130215213405.58b3a93a@quenya>
	<CA+9+E=Wnubt_Wy+xHjk0C6kw0XQ8tc2-agbFvkevcTpYJEqg4Q@mail.gmail.com>
	<720785848-1361030221-cardhu_decombobulator_blackberry.rim.net-863913820-@b16.c11.bise7.blackberry>
	<CA+9+E=Xh_1QHALsUzOhnUOK7q4uD8ZyoLwHoEs2R-zHh=7Ox5g@mail.gmail.com>
	<CA+9+E=U+N0GJY7QOmvTYgW5fVsetEZ_e2rYTOtp_EWeH5U7HTg@mail.gmail.com>
	<CA+9+E=U215fg8=y_ff46U2pyUxB4imXi1-i1Sx3bMbtECZ+28A@mail.gmail.com>
	<51224FFD.4020408@sipwise.com> <51225420.5000607@sipwise.com>
	<CA+9+E=U-t1paf0KW+H-vHkOwGFBkbn1=8NFd5YUPdEwm_xyZsg@mail.gmail.com>
	<5122598B.1060703@sipwise.com>
	<CA+9+E=XeYZRHuBQkhsHBTMCarq2bOdX9c+93suqyCCb-Gj+3Ag@mail.gmail.com>
	<20130226152313.2d977e1d@quenya>
	<CA+9+E=WQUrn1HbeekF2fchdZ+7kX54Snphe0QY6W_x4oXN6yyA@mail.gmail.com>
	<CA+9+E=XRB-hPxK1ZTBU5ebhnaq-y0wS0=B=8ZaeVCheBC6+tWg@mail.gmail.com>
	<CA+9+E=WU=Sa9NTZ=HJrzLXZ+VnTON_--+9fBaHtivs+Lfh7Vwg@mail.gmail.com>
	<20130305075953.50b44f18@multivac>
	<CA+9+E=UKknSzZGHFp5qi2cLx=NdaZk879jQPQi_BvEY7ce-jAg@mail.gmail.com>
	<CA+9+E=UfxgtRPVjC_8BhcEm0kh2HSKnErhro7WrFJ-eLu8SzHA@mail.gmail.com>
Message-ID: <CA+9+E=X3bfBdtmTSy=TycC=veshUU8Wgb8H6HCyEzjvW1RpSJg@mail.gmail.com>

And the SIP trace from the SNOM logs. I don't see any refer-to being sent
from the phone....


Sent to udp:196.41.123.113:5060 at 6/3/2013 12:18:02:571 (1617 bytes):

INVITE sip:ngcp-lb at 196.41.123.113:5060;ngcpct='sip:127.0.0.1:5080' SIP/2.0
Via: SIP/2.0/UDP 192.168.10.62:2050;branch=z9hG4bK-ttlvkk77nm7u;rport
Route: <sip:196.41.123.113;r2=on;lr=on;ftag=v0z1vszklt;nat=yes;ngcplb=yes>
Route: <sip:127.0.0.1;r2=on;lr=on;ftag=v0z1vszklt;nat=yes;ngcplb=yes>
Route: <sip:127.0.0.1:5062
;lr=on;ftag=v0z1vszklt;did=da1.e171;mpd=ii;rtpprx=yes;vsf=V3V3ZwIILyJUOU1ORmtBYW50Z2JXdXdnRUx4YVQ5TU5mS2E->
From: "AA Office" <sip:user403 at domain.domain.com>;tag=v0z1vszklt
To: <sip:400 at domain.domain.com
;user=phone>;tag=6E486C78-513717CB00030190-2E40F700
Call-ID: 513717c8ba36-755kmq45j2i5
CSeq: 3 INVITE
Max-Forwards: 70
Contact: <sip:user403 at 192.168.10.62:2050;line=pxk7nlcg>;reg-id=1
X-Serialnumber: 0004132FAC49
P-Key-Flags: keys="3"
User-Agent: snom300/8.7.3.19
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 489

v=0
o=root 1854415615 1854415616 IN IP4 192.168.10.62
s=call
c=IN IP4 192.168.10.62
t=0 0
m=audio 64268 RTP/AVP 9 0 8 3 99 108 18 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:fhqyKsE9KbQmap0NxeGKIFRZ3Yj2so3mw/32wC5U
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:108 AAL2-G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendonly
Received from udp:196.41.123.113:5060 at 6/3/2013 12:18:02:719 (386 bytes):

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.62:2050
;received=41.177.66.139;branch=z9hG4bK-ttlvkk77nm7u;rport=14297
From: "AA Office" <sip:user403 at domain.domain.com>;tag=v0z1vszklt
To: <sip:400 at domain.domain.com
;user=phone>;tag=6E486C78-513717CB00030190-2E40F700
Call-ID: 513717c8ba36-755kmq45j2i5
CSeq: 3 INVITE
Server: Sipwise NGCP Proxy 2.X
Content-Length: 0

Received from udp:196.41.123.113:5060 at 6/3/2013 12:18:03:231 (1253 bytes):

SIP/2.0 200 OK
Record-Route: <sip:127.0.0.1:5062;lr=on;ftag=v0z1vszklt;rtpprx=yes;mpd=ii>
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=v0z1vszklt;nat=yes;ngcplb=yes>
Record-Route:
<sip:196.41.123.113;r2=on;lr=on;ftag=v0z1vszklt;nat=yes;ngcplb=yes>
Via: SIP/2.0/UDP 192.168.10.62:2050
;received=41.177.66.139;branch=z9hG4bK-ttlvkk77nm7u;rport=14297
From: "AA Office" <sip:user403 at domain.domain.com>;tag=v0z1vszklt
To: <sip:400 at domain.domain.com
;user=phone>;tag=6E486C78-513717CB00030190-2E40F700
Call-ID: 513717c8ba36-755kmq45j2i5
CSeq: 3 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
SUBSCRIBE
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper rev.11619
Allow-Events: presence, kpml
P-NGCP-Src-Ip: 41.177.66.116
P-NGCP-Src-Port: 22106
P-NGCP-Src-Proto: udp
P-NGCP-Src-Af: 4
Content-Type: application/sdp
Content-Length: 254
Contact: <sip:ngcp-lb at 196.41.123.113:5060;ngcpct='sip:127.0.0.1:5080'>

v=0
o=Z 0 3 IN IP4 196.41.123.113
s=Z
c=IN IP4 196.41.123.113
t=0 0
m=audio 37342 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=inactive
a=direction:active
a=oldmediaip:192.168.10.88
a=nortpproxy:yes
Sent to udp:196.41.123.113:5060 at 6/3/2013 12:18:03:255 (741 bytes):

ACK sip:ngcp-lb at 196.41.123.113:5060;ngcpct='sip:127.0.0.1:5080' SIP/2.0
Via: SIP/2.0/UDP 192.168.10.62:2050;branch=z9hG4bK-cntszxfllm77;rport
Route: <sip:196.41.123.113;r2=on;lr=on;ftag=v0z1vszklt;nat=yes;ngcplb=yes>
Route: <sip:127.0.0.1;r2=on;lr=on;ftag=v0z1vszklt;nat=yes;ngcplb=yes>
Route: <sip:127.0.0.1:5062
;lr=on;ftag=v0z1vszklt;did=da1.e171;mpd=ii;rtpprx=yes;vsf=V3V3ZwIILyJUOU1ORmtBYW50Z2JXdXdnRUx4YVQ5TU5mS2E->
From: "AA Office" <sip:user403 at domain.domain.com>;tag=v0z1vszklt
To: <sip:400 at domain.domain.com
;user=phone>;tag=6E486C78-513717CB00030190-2E40F700
Call-ID: 513717c8ba36-755kmq45j2i5
CSeq: 3 ACK
Max-Forwards: 70
Contact: <sip:user403 at 192.168.10.62:2050;line=pxk7nlcg>;reg-id=1
Content-Length: 0


On Wed, Mar 6, 2013 at 11:41 AM, Theo <axessofficetheo at gmail.com> wrote:

> A little more now - I upgraded the firmware of a SNOM phone. Following the
> same process, upon pressing the transfer button I now get this:
>
> U 2013/03/06 11:36:22.236542 41.177.66.116:22106 -> 196.41.123.113:5060
> SIP/2.0 200 OK'
> Via: SIP/2.0/UDP
> 196.41.123.113;branch=z9hG4bKaa2.bc57c1db3d0853f982c750b38e5926b2.0'
> Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK~IoIoamR;rport=5080'
> Record-Route:
> <sip:196.41.123.113;lr;r2=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
> Record-Route:
> <sip:127.0.0.1;r2=on;lr=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
> Contact: <sip:user400 at 192.168.10.88:5060
> ;rinstance=7e064df24c43ccf0;transport=UDP>'
> To: <sip:user400 at domain.domain.com>;tag=d37f6717'
> From: <sip:2721403 at domain.domain.com
> >;tag=304E3AA0-51370E09000C6324-2EE19700'
> Call-ID: 51370e076928-0af4wphlmirc_b2b-1'
> CSeq: 11 INVITE'
> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
> SUBSCRIBE'
> Content-Type: application/sdp'
> Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri'
> User-Agent: Zoiper rev.11619'
> Allow-Events: presence, kpml'
> Content-Length: 185'
> '
> v=0'
> o=Z 0 3 IN IP4 192.168.10.88'
> s=Z'
> c=IN IP4 192.168.10.88'
> t=0 0'
> m=audio 8000 RTP/AVP 8 101'
> a=rtpmap:8 PCMA/8000'
> a=rtpmap:101 telephone-event/8000'
> a=fmtp:101 0-15'
> a=inactive'
>
> #
> U 2013/03/06 11:36:22.236896 127.0.0.1:5060 -> 127.0.0.1:5080
> SIP/2.0 200 OK'
> Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK~IoIoamR;rport=5080'
> Record-Route:
> <sip:196.41.123.113;lr;r2=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
> Record-Route:
> <sip:127.0.0.1;r2=on;lr=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
> Contact: <sip:user400 at 192.168.10.88:5060
> ;alias=41.177.66.116~22106~1;rinstance=7e064df24c43ccf0;transport=UDP>'
> To: <sip:user400 at domain.domain.com>;tag=d37f6717'
> From: <sip:2721403 at domain.domain.com
> >;tag=304E3AA0-51370E09000C6324-2EE19700'
> Call-ID: 51370e076928-0af4wphlmirc_b2b-1'
> CSeq: 11 INVITE'
> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
> SUBSCRIBE'
> Content-Type: application/sdp'
> Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri'
> User-Agent: Zoiper rev.11619'
> Allow-Events: presence, kpml'
> Content-Length: 233'
> P-NGCP-Src-Ip: 41.177.66.116'
> P-NGCP-Src-Port: 22106'
> P-NGCP-Src-Proto: udp'
> P-NGCP-Src-Af: 4'
> '
> v=0'
> o=Z 0 3 IN IP4 192.168.10.88'
> s=Z'
> c=IN IP4 41.177.66.116'
> t=0 0'
> m=audio 8000 RTP/AVP 8 101'
> a=rtpmap:8 PCMA/8000'
> a=rtpmap:101 telephone-event/8000'
> a=fmtp:101 0-15'
> a=inactive'
> a=direction:active'
> a=oldmediaip:192.168.10.88'
>
> #
> U 2013/03/06 11:36:22.236968 127.0.0.1:5080 -> 127.0.0.1:5060
> ACK sip:user400 at 192.168.10.88:5060;alias=41.177.66.116~22106~1;rinstance=7e064df24c43ccf0;transport=UDP
> SIP/2.0'
> Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bKTbHrca0H;rport'
> From: <sip:2721403 at domain.domain.com
> >;tag=304E3AA0-51370E09000C6324-2EE19700'
> To: <sip:user400 at domain.domain.com>;tag=d37f6717'
> CSeq: 11 ACK'
> Call-ID: 51370e076928-0af4wphlmirc_b2b-1'
> Contact: <sip:127.0.0.1:5080>'
> Route:
> <sip:127.0.0.1;r2=on;lr=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>,
> <sip:196.41.123.113;lr;r2=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
> Max-Forwards: 68'
> Content-Length: 0'
> '
>
> ##
> U 2013/03/06 11:36:22.237048 196.41.123.113:5060 -> 41.177.66.116:22106
> ACK sip:user400 at 192.168.10.88:5060;rinstance=7e064df24c43ccf0;transport=UDP
> SIP/2.0'
> Record-Route:
> <sip:196.41.123.113;r2=on;lr=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
> Record-Route:
> <sip:127.0.0.1;r2=on;lr=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
> Via: SIP/2.0/UDP
> 196.41.123.113;branch=z9hG4bKaa2.04ec964b43561d47cbdd4d22812c93a8.0'
> Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bKTbHrca0H;rport=5080'
> From: <sip:2721403 at domain.domain.com
> >;tag=304E3AA0-51370E09000C6324-2EE19700'
> To: <sip:user400 at domain.domain.com>;tag=d37f6717'
> CSeq: 11 ACK'
> Call-ID: 51370e076928-0af4wphlmirc_b2b-1'
> Max-Forwards: 67'
> Content-Length: 0'
> Contact: <sip:ngcp-lb at 196.41.123.113:5060;ngcpct='sip:127.0.0.1:5080'>'
>
>
> On Wed, Mar 6, 2013 at 9:41 AM, Theo <axessofficetheo at gmail.com> wrote:
>
>> Hi
>>
>> ngrep-sip gives nothing - Here's the process on a SNOM phone.
>>
>> Call comes in. You put it on hold. You dial another extension and
>> announce the call. All that shows up with ngrep-sip. Once the call is
>> announced you press the transfer button. Nothing is recorded for that with
>> ngrep sip whatsoever, which is obviously why it doesn't work :-(.
>>
>> Any ideas on that one?
>>
>>
>> On Tue, Mar 5, 2013 at 8:59 AM, Jon Bonilla <jbonilla at sipwise.com> wrote:
>>
>>> El Mon, 4 Mar 2013 08:59:03 +0200
>>> Theo <axessofficetheo at gmail.com> escribi?:
>>>
>>> > Hi
>>> >
>>> > Not sure if I am breaking protocol by bumping - if I did - apologies.
>>> I am
>>> > almost ready to put this in as our SBC upper registration solution
>>> (more to
>>> > follow after that I hope :-) - but need to resolve the attended
>>> transfer
>>> > problem. Any ideas on this? Sip Header filtering is off altogether
>>> (does
>>> > that have any caveats?). Transfer working fine now for all phones
>>> tested,
>>> > but attended transfer not.
>>> >
>>>
>>> ngrep-sip?
>>>
>>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130306/7d4f7ea9/attachment.html>

From admartin at nixus.es  Wed Mar  6 06:35:45 2013
From: admartin at nixus.es (Antonio David Martin)
Date: Wed, 6 Mar 2013 11:35:45 +0000
Subject: [Spce-user] live monitoring from shell
Message-ID: <70A7DFAD7F7A5A4EB0F34A47468371284E41C432@ACR01WIN004.grupoacr.com>

Is there any way to get the active calls from the command line? Something similar as in Asterisk (Asterisk 1.8.8.0)

#core show channels verbose
pbx*CLI> core show channels verbose
Channel              Context              Extension        Prio State   Application  Data                      CallerID        Duration Accountcode PeerAccount BridgedTo
SIP/nova105-00005 from-nova         102                 1 Up      Dial         SIP/nova102,,xXt       nova105      00:00:16                         SIP/nova102-00005
SIP/nova101-00005 from-nova         968777777           1 Up      Congestion   (Empty)                   34902000000     00:00:06                         (None)
SIP/oco103-000059c from-oco           1                   1 Up      AppQueue     (Outgoing Line)                           00:01:30                         SIP/asterisk-000059c
SIP/nova102-00005 from-nova                             1 Up      AppDial      (Outgoing Line)           102             00:00:15                         SIP/nova105-00005
SIP/asterisk-000059c incoming-oco       1                   2 Up      Queue        oco-atencionalcliente,t 607000000       00:01:40                         SIP/oco103-000059c
5 active channels
3 active calls
10785 calls processed

I have seen this command   "ngcp-kamctl proxy fifo get_statistics :dialog::all"  but It doesn't show the subscribers, source, destination, etc

Thanks in advance

David Mart?n



-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130306/b2f11a61/attachment.html>

From apogrebennyk at sipwise.com  Wed Mar  6 11:09:31 2013
From: apogrebennyk at sipwise.com (Andrew Pogrebennyk)
Date: Wed, 06 Mar 2013 17:09:31 +0100
Subject: [Spce-user] live monitoring from shell
In-Reply-To: <70A7DFAD7F7A5A4EB0F34A47468371284E41C432@ACR01WIN004.grupoacr.com>
References: <70A7DFAD7F7A5A4EB0F34A47468371284E41C432@ACR01WIN004.grupoacr.com>
Message-ID: <51376A3B.6030301@sipwise.com>

Hi David,

just do:
ngcp-kamctl proxy fifo dlg_list

Also you may want to check the sems project for call monitoring tools.

HTH.
Andrew

On 03/06/2013 12:35 PM, Antonio David Martin wrote:
> Is there any way to get the active calls from the command line?
> Something similar as in Asterisk (Asterisk 1.8.8.0)
> 
>  
> 
> #core show channels verbose
> 
> pbx*CLI> core show channels verbose
> 
> Channel              Context              Extension        Prio State  
> Application  Data                      CallerID        Duration
> Accountcode PeerAccount BridgedTo          
> 
> SIP/nova105-00005 from-nova         102                 1 Up     
> Dial         SIP/nova102,,xXt       nova105     
> 00:00:16                         SIP/nova102-00005
> 
> SIP/nova101-00005 from-nova         968777777           1 Up     
> Congestion   (Empty)                   34902000000    
> 00:00:06                         (None)             
> 
> SIP/oco103-000059c from-oco           1                   1 Up     
> AppQueue     (Outgoing Line)                          
> 00:01:30                         SIP/asterisk-000059c
> 
> SIP/nova102-00005 from-nova                             1 Up     
> AppDial      (Outgoing Line)           102            
> 00:00:15                         SIP/nova105-00005
> 
> SIP/asterisk-000059c incoming-oco       1                   2 Up     
> Queue        oco-atencionalcliente,t 607000000      
> 00:01:40                         SIP/oco103-000059c
> 
> 5 active channels
> 
> 3 active calls
> 
> 10785 calls processed
> 
>  
> 
> I have seen this command   ?ngcp-kamctl proxy fifo get_statistics
> :dialog::all?  but It doesn?t show the subscribers, source, destination, etc
> 
>  
> 
> Thanks in advance
> 
>  
> 
> *David Mart?n **
> 
> 
> ***
> 
>  
> 
> 
> 
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
> 




From axessofficetheo at gmail.com  Wed Mar  6 11:28:59 2013
From: axessofficetheo at gmail.com (Theo)
Date: Wed, 6 Mar 2013 18:28:59 +0200
Subject: [Spce-user] Upper Registration to Asterisk - transfers not
	working
In-Reply-To: <CA+9+E=X3bfBdtmTSy=TycC=veshUU8Wgb8H6HCyEzjvW1RpSJg@mail.gmail.com>
References: <CA+9+E=X_qiRReX=c_XubTD89HEV2HApxFw1yRDgDa2nW0o_SZg@mail.gmail.com>
	<20130215213405.58b3a93a@quenya>
	<CA+9+E=Wnubt_Wy+xHjk0C6kw0XQ8tc2-agbFvkevcTpYJEqg4Q@mail.gmail.com>
	<720785848-1361030221-cardhu_decombobulator_blackberry.rim.net-863913820-@b16.c11.bise7.blackberry>
	<CA+9+E=Xh_1QHALsUzOhnUOK7q4uD8ZyoLwHoEs2R-zHh=7Ox5g@mail.gmail.com>
	<CA+9+E=U+N0GJY7QOmvTYgW5fVsetEZ_e2rYTOtp_EWeH5U7HTg@mail.gmail.com>
	<CA+9+E=U215fg8=y_ff46U2pyUxB4imXi1-i1Sx3bMbtECZ+28A@mail.gmail.com>
	<51224FFD.4020408@sipwise.com> <51225420.5000607@sipwise.com>
	<CA+9+E=U-t1paf0KW+H-vHkOwGFBkbn1=8NFd5YUPdEwm_xyZsg@mail.gmail.com>
	<5122598B.1060703@sipwise.com>
	<CA+9+E=XeYZRHuBQkhsHBTMCarq2bOdX9c+93suqyCCb-Gj+3Ag@mail.gmail.com>
	<20130226152313.2d977e1d@quenya>
	<CA+9+E=WQUrn1HbeekF2fchdZ+7kX54Snphe0QY6W_x4oXN6yyA@mail.gmail.com>
	<CA+9+E=XRB-hPxK1ZTBU5ebhnaq-y0wS0=B=8ZaeVCheBC6+tWg@mail.gmail.com>
	<CA+9+E=WU=Sa9NTZ=HJrzLXZ+VnTON_--+9fBaHtivs+Lfh7Vwg@mail.gmail.com>
	<20130305075953.50b44f18@multivac>
	<CA+9+E=UKknSzZGHFp5qi2cLx=NdaZk879jQPQi_BvEY7ce-jAg@mail.gmail.com>
	<CA+9+E=UfxgtRPVjC_8BhcEm0kh2HSKnErhro7WrFJ-eLu8SzHA@mail.gmail.com>
	<CA+9+E=X3bfBdtmTSy=TycC=veshUU8Wgb8H6HCyEzjvW1RpSJg@mail.gmail.com>
Message-ID: <CA+9+E=V-nzXP5CeQMCBKYkyp-oJ3xKiVmKgj5D9UpBoh_363pg@mail.gmail.com>

Tried with some other phones and it worked, so it seems specific to the
SNOM phones. Of course we have over a thousand SNOM phones out there so for
us to start using this we'd have to find a fix. My guess is that SNOM needs
to fix it, but don't have the required ammo nor knowledge to take them to
task on it (unless I am completely wrong to start off with). Aside from
that, I doubt whether they would fix that in a matter of days and I had
hoped to start implementing this as a solution.

Anything that can be done on the scpe to help with this? Any words of
wisdom I can share with SNOM?

On Wed, Mar 6, 2013 at 12:45 PM, Theo <axessofficetheo at gmail.com> wrote:

> And the SIP trace from the SNOM logs. I don't see any refer-to being sent
> from the phone....
>
>
> Sent to udp:196.41.123.113:5060 at 6/3/2013 12:18:02:571 (1617 bytes):
>
> INVITE sip:ngcp-lb at 196.41.123.113:5060;ngcpct='sip:127.0.0.1:5080' SIP/2.0
> Via: SIP/2.0/UDP 192.168.10.62:2050;branch=z9hG4bK-ttlvkk77nm7u;rport
> Route: <sip:196.41.123.113;r2=on;lr=on;ftag=v0z1vszklt;nat=yes;ngcplb=yes>
> Route: <sip:127.0.0.1;r2=on;lr=on;ftag=v0z1vszklt;nat=yes;ngcplb=yes>
> Route: <sip:127.0.0.1:5062
> ;lr=on;ftag=v0z1vszklt;did=da1.e171;mpd=ii;rtpprx=yes;vsf=V3V3ZwIILyJUOU1ORmtBYW50Z2JXdXdnRUx4YVQ5TU5mS2E->
> From: "AA Office" <sip:user403 at domain.domain.com>;tag=v0z1vszklt
> To: <sip:400 at domain.domain.com
> ;user=phone>;tag=6E486C78-513717CB00030190-2E40F700
> Call-ID: 513717c8ba36-755kmq45j2i5
> CSeq: 3 INVITE
> Max-Forwards: 70
> Contact: <sip:user403 at 192.168.10.62:2050;line=pxk7nlcg>;reg-id=1
> X-Serialnumber: 0004132FAC49
> P-Key-Flags: keys="3"
> User-Agent: snom300/8.7.3.19
> Accept: application/sdp
> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
> MESSAGE, INFO, UPDATE
> Allow-Events: talk, hold, refer, call-info
> Supported: timer, 100rel, replaces, from-change
> Session-Expires: 3600;refresher=uas
> Min-SE: 90
> Content-Type: application/sdp
> Content-Length: 489
>
> v=0
> o=root 1854415615 1854415616 IN IP4 192.168.10.62
> s=call
> c=IN IP4 192.168.10.62
> t=0 0
> m=audio 64268 RTP/AVP 9 0 8 3 99 108 18 101
> a=crypto:1 AES_CM_128_HMAC_SHA1_32
> inline:fhqyKsE9KbQmap0NxeGKIFRZ3Yj2so3mw/32wC5U
> a=rtpmap:9 G722/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:99 G726-32/8000
> a=rtpmap:108 AAL2-G726-32/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
> a=sendonly
> Received from udp:196.41.123.113:5060 at 6/3/2013 12:18:02:719 (386
> bytes):
>
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 192.168.10.62:2050
> ;received=41.177.66.139;branch=z9hG4bK-ttlvkk77nm7u;rport=14297
> From: "AA Office" <sip:user403 at domain.domain.com>;tag=v0z1vszklt
> To: <sip:400 at domain.domain.com
> ;user=phone>;tag=6E486C78-513717CB00030190-2E40F700
> Call-ID: 513717c8ba36-755kmq45j2i5
> CSeq: 3 INVITE
> Server: Sipwise NGCP Proxy 2.X
> Content-Length: 0
>
> Received from udp:196.41.123.113:5060 at 6/3/2013 12:18:03:231 (1253
> bytes):
>
> SIP/2.0 200 OK
> Record-Route: <sip:127.0.0.1:5062;lr=on;ftag=v0z1vszklt;rtpprx=yes;mpd=ii>
> Record-Route:
> <sip:127.0.0.1;r2=on;lr=on;ftag=v0z1vszklt;nat=yes;ngcplb=yes>
> Record-Route:
> <sip:196.41.123.113;r2=on;lr=on;ftag=v0z1vszklt;nat=yes;ngcplb=yes>
> Via: SIP/2.0/UDP 192.168.10.62:2050
> ;received=41.177.66.139;branch=z9hG4bK-ttlvkk77nm7u;rport=14297
> From: "AA Office" <sip:user403 at domain.domain.com>;tag=v0z1vszklt
> To: <sip:400 at domain.domain.com
> ;user=phone>;tag=6E486C78-513717CB00030190-2E40F700
> Call-ID: 513717c8ba36-755kmq45j2i5
> CSeq: 3 INVITE
> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
> SUBSCRIBE
> Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
> User-Agent: Zoiper rev.11619
> Allow-Events: presence, kpml
> P-NGCP-Src-Ip: 41.177.66.116
> P-NGCP-Src-Port: 22106
> P-NGCP-Src-Proto: udp
> P-NGCP-Src-Af: 4
> Content-Type: application/sdp
> Content-Length: 254
> Contact: <sip:ngcp-lb at 196.41.123.113:5060;ngcpct='sip:127.0.0.1:5080'>
>
> v=0
> o=Z 0 3 IN IP4 196.41.123.113
> s=Z
> c=IN IP4 196.41.123.113
> t=0 0
> m=audio 37342 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=inactive
> a=direction:active
> a=oldmediaip:192.168.10.88
> a=nortpproxy:yes
> Sent to udp:196.41.123.113:5060 at 6/3/2013 12:18:03:255 (741 bytes):
>
> ACK sip:ngcp-lb at 196.41.123.113:5060;ngcpct='sip:127.0.0.1:5080' SIP/2.0
> Via: SIP/2.0/UDP 192.168.10.62:2050;branch=z9hG4bK-cntszxfllm77;rport
> Route: <sip:196.41.123.113;r2=on;lr=on;ftag=v0z1vszklt;nat=yes;ngcplb=yes>
> Route: <sip:127.0.0.1;r2=on;lr=on;ftag=v0z1vszklt;nat=yes;ngcplb=yes>
> Route: <sip:127.0.0.1:5062
> ;lr=on;ftag=v0z1vszklt;did=da1.e171;mpd=ii;rtpprx=yes;vsf=V3V3ZwIILyJUOU1ORmtBYW50Z2JXdXdnRUx4YVQ5TU5mS2E->
> From: "AA Office" <sip:user403 at domain.domain.com>;tag=v0z1vszklt
> To: <sip:400 at domain.domain.com
> ;user=phone>;tag=6E486C78-513717CB00030190-2E40F700
> Call-ID: 513717c8ba36-755kmq45j2i5
> CSeq: 3 ACK
> Max-Forwards: 70
> Contact: <sip:user403 at 192.168.10.62:2050;line=pxk7nlcg>;reg-id=1
> Content-Length: 0
>
>
> On Wed, Mar 6, 2013 at 11:41 AM, Theo <axessofficetheo at gmail.com> wrote:
>
>> A little more now - I upgraded the firmware of a SNOM phone. Following
>> the same process, upon pressing the transfer button I now get this:
>>
>> U 2013/03/06 11:36:22.236542 41.177.66.116:22106 -> 196.41.123.113:5060
>> SIP/2.0 200 OK'
>> Via: SIP/2.0/UDP
>> 196.41.123.113;branch=z9hG4bKaa2.bc57c1db3d0853f982c750b38e5926b2.0'
>> Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK~IoIoamR;rport=5080'
>> Record-Route:
>> <sip:196.41.123.113;lr;r2=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
>> Record-Route:
>> <sip:127.0.0.1;r2=on;lr=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
>> Contact: <sip:user400 at 192.168.10.88:5060
>> ;rinstance=7e064df24c43ccf0;transport=UDP>'
>> To: <sip:user400 at domain.domain.com>;tag=d37f6717'
>> From: <sip:2721403 at domain.domain.com
>> >;tag=304E3AA0-51370E09000C6324-2EE19700'
>> Call-ID: 51370e076928-0af4wphlmirc_b2b-1'
>> CSeq: 11 INVITE'
>> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
>> SUBSCRIBE'
>> Content-Type: application/sdp'
>> Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri'
>> User-Agent: Zoiper rev.11619'
>> Allow-Events: presence, kpml'
>> Content-Length: 185'
>> '
>> v=0'
>> o=Z 0 3 IN IP4 192.168.10.88'
>> s=Z'
>> c=IN IP4 192.168.10.88'
>> t=0 0'
>> m=audio 8000 RTP/AVP 8 101'
>> a=rtpmap:8 PCMA/8000'
>> a=rtpmap:101 telephone-event/8000'
>> a=fmtp:101 0-15'
>> a=inactive'
>>
>> #
>> U 2013/03/06 11:36:22.236896 127.0.0.1:5060 -> 127.0.0.1:5080
>> SIP/2.0 200 OK'
>> Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK~IoIoamR;rport=5080'
>> Record-Route:
>> <sip:196.41.123.113;lr;r2=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
>> Record-Route:
>> <sip:127.0.0.1;r2=on;lr=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
>> Contact: <sip:user400 at 192.168.10.88:5060
>> ;alias=41.177.66.116~22106~1;rinstance=7e064df24c43ccf0;transport=UDP>'
>> To: <sip:user400 at domain.domain.com>;tag=d37f6717'
>> From: <sip:2721403 at domain.domain.com
>> >;tag=304E3AA0-51370E09000C6324-2EE19700'
>> Call-ID: 51370e076928-0af4wphlmirc_b2b-1'
>> CSeq: 11 INVITE'
>> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
>> SUBSCRIBE'
>> Content-Type: application/sdp'
>> Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri'
>> User-Agent: Zoiper rev.11619'
>> Allow-Events: presence, kpml'
>> Content-Length: 233'
>> P-NGCP-Src-Ip: 41.177.66.116'
>> P-NGCP-Src-Port: 22106'
>> P-NGCP-Src-Proto: udp'
>> P-NGCP-Src-Af: 4'
>> '
>> v=0'
>> o=Z 0 3 IN IP4 192.168.10.88'
>> s=Z'
>> c=IN IP4 41.177.66.116'
>> t=0 0'
>> m=audio 8000 RTP/AVP 8 101'
>> a=rtpmap:8 PCMA/8000'
>> a=rtpmap:101 telephone-event/8000'
>> a=fmtp:101 0-15'
>> a=inactive'
>> a=direction:active'
>> a=oldmediaip:192.168.10.88'
>>
>> #
>> U 2013/03/06 11:36:22.236968 127.0.0.1:5080 -> 127.0.0.1:5060
>>  ACK sip:user400 at 192.168.10.88:5060;alias=41.177.66.116~22106~1;rinstance=7e064df24c43ccf0;transport=UDP
>> SIP/2.0'
>> Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bKTbHrca0H;rport'
>> From: <sip:2721403 at domain.domain.com
>> >;tag=304E3AA0-51370E09000C6324-2EE19700'
>> To: <sip:user400 at domain.domain.com>;tag=d37f6717'
>> CSeq: 11 ACK'
>> Call-ID: 51370e076928-0af4wphlmirc_b2b-1'
>> Contact: <sip:127.0.0.1:5080>'
>> Route:
>> <sip:127.0.0.1;r2=on;lr=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>,
>> <sip:196.41.123.113;lr;r2=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
>> Max-Forwards: 68'
>> Content-Length: 0'
>> '
>>
>> ##
>> U 2013/03/06 11:36:22.237048 196.41.123.113:5060 -> 41.177.66.116:22106
>> ACK sip:user400 at 192.168.10.88:5060;rinstance=7e064df24c43ccf0;transport=UDP
>> SIP/2.0'
>> Record-Route:
>> <sip:196.41.123.113;r2=on;lr=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
>> Record-Route:
>> <sip:127.0.0.1;r2=on;lr=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
>> Via: SIP/2.0/UDP
>> 196.41.123.113;branch=z9hG4bKaa2.04ec964b43561d47cbdd4d22812c93a8.0'
>> Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bKTbHrca0H;rport=5080'
>> From: <sip:2721403 at domain.domain.com
>> >;tag=304E3AA0-51370E09000C6324-2EE19700'
>> To: <sip:user400 at domain.domain.com>;tag=d37f6717'
>> CSeq: 11 ACK'
>> Call-ID: 51370e076928-0af4wphlmirc_b2b-1'
>> Max-Forwards: 67'
>> Content-Length: 0'
>> Contact: <sip:ngcp-lb at 196.41.123.113:5060;ngcpct='sip:127.0.0.1:5080'>'
>>
>>
>> On Wed, Mar 6, 2013 at 9:41 AM, Theo <axessofficetheo at gmail.com> wrote:
>>
>>> Hi
>>>
>>> ngrep-sip gives nothing - Here's the process on a SNOM phone.
>>>
>>> Call comes in. You put it on hold. You dial another extension and
>>> announce the call. All that shows up with ngrep-sip. Once the call is
>>> announced you press the transfer button. Nothing is recorded for that with
>>> ngrep sip whatsoever, which is obviously why it doesn't work :-(.
>>>
>>> Any ideas on that one?
>>>
>>>
>>> On Tue, Mar 5, 2013 at 8:59 AM, Jon Bonilla <jbonilla at sipwise.com>wrote:
>>>
>>>> El Mon, 4 Mar 2013 08:59:03 +0200
>>>> Theo <axessofficetheo at gmail.com> escribi?:
>>>>
>>>> > Hi
>>>> >
>>>> > Not sure if I am breaking protocol by bumping - if I did - apologies.
>>>> I am
>>>> > almost ready to put this in as our SBC upper registration solution
>>>> (more to
>>>> > follow after that I hope :-) - but need to resolve the attended
>>>> transfer
>>>> > problem. Any ideas on this? Sip Header filtering is off altogether
>>>> (does
>>>> > that have any caveats?). Transfer working fine now for all phones
>>>> tested,
>>>> > but attended transfer not.
>>>> >
>>>>
>>>> ngrep-sip?
>>>>
>>>>
>>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130306/c9b4b465/attachment.html>

From matthew at tenacit.net  Thu Mar  7 10:04:56 2013
From: matthew at tenacit.net (Matthew Ogden)
Date: Thu, 7 Mar 2013 17:04:56 +0200
Subject: [Spce-user] peering rules priority
In-Reply-To: <20121106141253.1b5b604c@multivac>
References: <354a10abf6bdbe7436c2943328ed1441@mail.gmail.com>
	<20121106141253.1b5b604c@multivac>
Message-ID: <CAB4Z0mKtmJZ6dvwOUGKqp_1OUHL_etOg4mXgWP0zhjJmoVJA0Q@mail.gmail.com>

Now,

How to make one get twice as much traffic as the other for a set numbers of
same prefix length?
Just add another group with the same peer IP and prefixes?



On Tue, Nov 6, 2012 at 3:12 PM, Jon Bonilla <jbonilla at sipwise.com> wrote:

> El Tue, 6 Nov 2012 12:13:49 +0200
> Matthew Ogden <matthew at tenacit.net> escribi?:
>
> > How would I achieve this:
> >
> >
> >
> > I have 3 peers, (currently as 3 separate peer groups)
> >
> >
> >
> > For certain number prefixes, I would like equal load balancing if
> possible.
> >
> >
> >
> > For everything else, I want them in a specific order of priority.
> >
>
> The gw selection priority is like this:
>
> - First match length
> - Second group priority
> - Third gw weight
>
> So for balancing between you 3 groupd you could have same match length,
> priority and weight in all three nodes.
>
> For others where you want strict priority you can add the preffixes with
> different match lenght if possible. An idea would be to use callee pattern.
> something like this (example spain):
>
> ^34
> ^34[0-9]
> ^34[0-9][0-9]
>
> There's another option if this is not possible but requires kamailio's lcr
> module patching. I've just tested it in my last week's training in Madrid
> and
> maybe could fit:
>
> - Make gws not unique between peer groups
> - Make gw weight behave as priority
> - Make group priority third option, after gw weight
>
> I could provide you with the patch if the default solution does not fit
> your
> needs.
>
>
> cheers,
>
> Jon
>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130307/4913bbaa/attachment.html>

From john.murray at skyracktelecom.com  Thu Mar  7 17:00:44 2013
From: john.murray at skyracktelecom.com (John Murray)
Date: Thu, 7 Mar 2013 22:00:44 -0000
Subject: [Spce-user] inbound PSTN peer routing to local sip uri
In-Reply-To: <51308CD2.7040802@sipwise.com>
References: <013501ce15b3$90327c00$b0977400$@murray@skyracktelecom.com>
	<20130301103021.4a56afaf@quenya>
	<51307a6a.8665b40a.4f88.ffffa26e@mx.google.com>
	<51308CD2.7040802@sipwise.com>
Message-ID: <51390e0d.0a08b40a.33f2.ffff98a0@mx.google.com>

Andreas hi,

That worked great. I actually did:
	if ($si == "1.2.1.2"){
		$rd = $td;
        }

Which is perfect. Thanks very much for that answer.

Regards

John



-----Original Message-----
From: spce-user-bounces at lists.sipwise.com
[mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of Andreas Granig
Sent: 01 March 2013 11:11
To: spce-user at lists.sipwise.com
Subject: Re: [Spce-user] inbound PSTN peer routing to local sip uri

Hi,
On 03/01/2013 10:52 AM, John Murray wrote:
> SIP URI: a001002 at skyrack.net

Well, the problem here is that the call comes in as
sip:a001002 at 100.100.100.182, but the subscriber is sip:a001002 at skyrack.net.

There is currently no way to also change the domain part via rewrite rules,
so you'd have to do that hardcoded in the kamailio-lb config, like this:

# after the whole loose-route block:
if($rd == "100.100.100.182") {
   $rd = "skyrack.net";
}

You can use it more elegantly of course by using the template values of
network.yml, but that's basically the idea for now.

Andreas

_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com
http://lists.sipwise.com/listinfo/spce-user




From john.murray at skyracktelecom.com  Thu Mar  7 17:09:44 2013
From: john.murray at skyracktelecom.com (John Murray)
Date: Thu, 7 Mar 2013 22:09:44 -0000
Subject: [Spce-user] Send peer inbound call straight to voicemail
Message-ID: <021301ce1b80$79b93db0$6d2bb910$@murray@skyracktelecom.com>

Hi,

 

I want to send a specific call type to voicemail.

 

I send a call out to a PBX peer who has no voicemail. If the handset is not
answered I get a call from the peer with RURI 500 which is the voicemail
number, From is the original caller and a Divert: header with the subscriber
ddi.

 

I know how to take off the Divert header and get the subscriber ddi but what
would I put in the RURI to send straight to the subscribers voicemail?

I've tried prefixing 'vmu' to the ddi but that didn't work.

 

Thanks for your help.

 

Regards

 

John

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130307/db122a52/attachment.html>

From jbonilla at sipwise.com  Thu Mar  7 19:22:02 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Fri, 8 Mar 2013 01:22:02 +0100
Subject: [Spce-user] peering rules priority
In-Reply-To: <CAB4Z0mKtmJZ6dvwOUGKqp_1OUHL_etOg4mXgWP0zhjJmoVJA0Q@mail.gmail.com>
References: <354a10abf6bdbe7436c2943328ed1441@mail.gmail.com>
	<20121106141253.1b5b604c@multivac>
	<CAB4Z0mKtmJZ6dvwOUGKqp_1OUHL_etOg4mXgWP0zhjJmoVJA0Q@mail.gmail.com>
Message-ID: <20130308012202.5b79c012@multivac>

El Thu, 7 Mar 2013 17:04:56 +0200
Matthew Ogden <matthew at tenacit.net> escribi?:

> Now,
> 
> How to make one get twice as much traffic as the other for a set numbers of
> same prefix length?
> Just add another group with the same peer IP and prefixes?
> 
> 

At the moment, due to the billing system, a gateway (same ip) can't be created
in two different peer groups.







From skchopperguy at gmail.com  Sat Mar  9 04:10:26 2013
From: skchopperguy at gmail.com (Skyler)
Date: Sat, 9 Mar 2013 01:10:26 -0800
Subject: [Spce-user] SMS
In-Reply-To: <51316C6E.2060201@gmail.com>
References: <EBB6CE73-9366-45FD-8E9E-8585458FA2B9@gmail.com>
	<20130301224615.1255f743@quenya>
	<CAG-pSShJ3mf8LdKKfwYjTj3=NyZ0gm_NkOBBBvmSin08MEYFrQ@mail.gmail.com>
	<513158C2.1000900@gmail.com>
	<9EF84F26-E19D-42E7-B77A-0FE3EB44BB21@gmail.com>
	<51316C6E.2060201@gmail.com>
Message-ID: <CAOUeCaQJA4bPkNfdH3k6EP5yxfahqPgAue929T3-tL0We9QQQg@mail.gmail.com>

Just for closure, the final response:yes SPCE does support SMS in this case.

On Fri, Mar 1, 2013 at 7:05 PM, Skyler <skchopperguy at gmail.com> wrote:

> That's very similar to what I was about to start work on. IM between
> multiple SPCE boxes. What the link describes is basically an IM being
> passed along to another system if not found locally. Can definitely be done
> but will take more thought into the handling and testing though.
>
> Skyler
>
>
> On 3/1/2013 5:50 PM, Jeremie Chism wrote:
>
>> Here is a link to the wiki our carrier has
>>
>> http://wiki.voipinnovations.**com/SMS.ashx<http://wiki.voipinnovations.com/SMS.ashx>
>>
>> Sent from my iPhone
>>
>> On Mar 1, 2013, at 7:41 PM, Skyler <skchopperguy at gmail.com
>> <mailto:skchopperguy at gmail.com**>> wrote:
>>
>>  In theory, if your SMSC can translate MESSAGE it should not be too
>>> hard to treat your SMSC in the same way peers are handled. ie: lookup
>>> local, if not found then go to smsc gateway. Mind you I'm sure it
>>> would be more involved than that.
>>>
>>> what are you using as an smsc? I'd be into tinkering for fun to see if
>>> a stable solution could come of it.
>>>
>>> Skyler
>>>
>>> On 3/1/2013 3:11 PM, Jeremie Chism wrote:
>>>
>>>> Yes it is sent by sip message.
>>>>
>>>> On Friday, March 1, 2013, Jon Bonilla wrote:
>>>>
>>>>    El Fri, 1 Mar 2013 15:37:41 -0600
>>>>    Jeremie Chism <jchism2 at gmail.com <mailto:jchism2 at gmail.com>
>>>>
>>>> <javascript:;>> escribi?:
>>>>
>>>>     > Does sipwise support SMS and if so is there something that needs
>>>>    to be
>>>>     > changed in the config to make it work.
>>>>
>>>>
>>>>    Could you tell us what exactly you mean by supporting sms? I guess
>>>>    the answer
>>>>    is no, but sending an http request on events to a sms gateway
>>>>    shouldn't be too
>>>>    difficult.
>>>>
>>>>
>>>>
>>>> --
>>>> Jeremie Chism
>>>> Triton Communications
>>>>
>>>>
>>>> ______________________________**_________________
>>>> Spce-user mailing list
>>>> Spce-user at lists.sipwise.com <mailto:Spce-user at lists.**sipwise.com<Spce-user at lists.sipwise.com>
>>>> >
>>>> http://lists.sipwise.com/**listinfo/spce-user<http://lists.sipwise.com/listinfo/spce-user>
>>>>
>>>>
>>> ______________________________**_________________
>>> Spce-user mailing list
>>> Spce-user at lists.sipwise.com <mailto:Spce-user at lists.**sipwise.com<Spce-user at lists.sipwise.com>
>>> >
>>> http://lists.sipwise.com/**listinfo/spce-user<http://lists.sipwise.com/listinfo/spce-user>
>>>
>>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130309/8a26345d/attachment.html>

From jchism2 at gmail.com  Sun Mar 10 20:21:00 2013
From: jchism2 at gmail.com (Jeremie Chism)
Date: Sun, 10 Mar 2013 19:21:00 -0500
Subject: [Spce-user] Voicemail box limit
Message-ID: <9E6CB8F5-123E-408B-A1BD-975096191BD6@gmail.com>

Is there a limit to the number of voicemails or time limit for voicemail box. A customer notified me that their mailbox was full. If there is a limit can it be changed. 

Thanks

Sent from my iPhone


From jirka.jirout at mac.com  Mon Mar 11 04:31:23 2013
From: jirka.jirout at mac.com (Jirka Jirout)
Date: Mon, 11 Mar 2013 09:31:23 +0100
Subject: [Spce-user] Fwd: TLS problem
References: <DBDE31C2-0278-4CBC-920B-4DA2B18D54EC@me.com>
Message-ID: <3182F19C-DA78-4627-B891-4E430230C651@mac.com>

Hello,

I have now spent almost a week trying to figure out some very weird TLS behavior. I am putting together a new client app and I am experiencing strange things here. The new app can connect in about 10 % of all cases. Since SPCE is still working fine with the existing ones, I naturally lamed the new application any me code. 

HOWEVER:
Today I decided to start from scratch and wrote a simple commmand line application that does only three things: opens a TLS connection, writes the REGISTER SIP message to the socket and prints the response (which should be something like 407 Unauthorized or something, but that would be fine).

The first two steps are fine - ssl_connect() and ssl_write() return success. But then the strange thins start to happen:

- No answer ever comes back
- When I run ngrep-sip on the server, I do not see the message anywhere, although tcpdump shows it on the external interface

I first though this might be a certificate problem or something, but running my app against the spce administration interface on port 1433 returns the data just fine, although the interface uses the same SSL certificates.

The weirdest thing of course is that in about 10 % of cases, my new client connects just fine and works for hours without any problem. And that the older clients work fine.

my kamailio.cfg for lb looks like this in the tls section:

#!ifdef ENABLE_TLS
loadmodule "tls.so"
#!endif
....
#!ifdef ENABLE_TLS
modparam("tls", "certificate", "/crypto-keys/certificate.pem")
modparam("tls", "private_key", "/crypto-keys/private.key")
modparam("tls", "tls_method", "SSLv23")
#!endif

SPCE is v 2.7, openssl 0.9.8.o

Any help would be greatly appreciated as this is slowly starting to drive me nuts ;-)

regards, jj




From raul.alonso at tpartner.net  Mon Mar 11 04:57:42 2013
From: raul.alonso at tpartner.net (Raul Alonso)
Date: Mon, 11 Mar 2013 09:57:42 +0100
Subject: [Spce-user] Voicemail box limit
In-Reply-To: <9E6CB8F5-123E-408B-A1BD-975096191BD6@gmail.com>
References: <9E6CB8F5-123E-408B-A1BD-975096191BD6@gmail.com>
Message-ID: <006d01ce1e36$7ea36db0$7bea4910$@tpartner.net>

Hi,

You can change these settings in the config.yml.

    max_msg_length: 180
    maxgreet: 60
    maxmsg: 30
    min_msg_length: 3

Regards,
Raul.

-----Mensaje original-----
De: spce-user-bounces at lists.sipwise.com
[mailto:spce-user-bounces at lists.sipwise.com] En nombre de Jeremie Chism
Enviado el: lunes, 11 de marzo de 2013 1:21
Para: spce-user at lists.sipwise.com
Asunto: [Spce-user] Voicemail box limit

Is there a limit to the number of voicemails or time limit for voicemail
box. A customer notified me that their mailbox was full. If there is a limit
can it be changed. 

Thanks

Sent from my iPhone
_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com
http://lists.sipwise.com/listinfo/spce-user




From jirka.jirout at me.com  Sat Mar  9 16:45:02 2013
From: jirka.jirout at me.com (Jirka Jirout)
Date: Sat, 09 Mar 2013 22:45:02 +0100
Subject: [Spce-user] TLS problem
Message-ID: <DBDE31C2-0278-4CBC-920B-4DA2B18D54EC@me.com>

Hello,

I have now spent almost a week trying to figure out some very weird TLS behavior. I am putting together a new client app and I am experiencing strange things here. The new app can connect in about 10 % of all cases. Since SPCE is still working fine with the existing ones, I naturally lamed the new application any me code. 

HOWEVER:
 Today I decided to start from scratch and wrote a simple commmand line application that does only three things: opens a TLS connection, writes the REGISTER SIP message to the socket and prints the response (which should be something like 407 Unauthorized or something, but that would be fine).

The first two steps are fine - ssl_connect() and ssl_write() return success. But then the strange thins start to happen:

- No answer ever comes back
- When I run ngrep-sip on the server, I do not see the message anywhere, although tcpdump shows it on the external interface

I first though this might be a certificate problem or something, but running my app against the spce administration interface on port 1433 returns the data just fine, although the interface uses the same SSL certificates.

Next I tried openssl s_client -connect server:port - The connection is established fine. When I type some nonsense, i. e. GET\r\n\r\n, I end up with "read:errno=0" and a closed socket. But when I try to send the message (copy & paste), I end up with " RENEGOTIATING 11504:error:1409E0E5:SSL routines:SSL3_WRITE_BYTES:ssl handshake failure:s3_pkt.c:545:" which is the same error that my new client is getting. To be sure, I tried from several computers with openssl 0.9.8o (also used on the server) and 1.0.1c and got the same results.

The weirdest thing of course is that in about 10 % of cases, my new client connects just fine and works for hours without any problem. And that the older clients work fine.

my kamailio.cfg for lb looks like this in the tls section:

#!ifdef ENABLE_TLS
loadmodule "tls.so"
#!endif
....
#!ifdef ENABLE_TLS
modparam("tls", "certificate", "/crypto-keys/certificate.pem")
modparam("tls", "private_key", "/crypto-keys/private.key")
modparam("tls", "tls_method", "SSLv23")
#!endif

SPCE is v 2.7, openssl 0.9.8.o

Any help would be greatly appreciated as this is slowly starting to drive me nuts ;-)

regards, jj


From jbonilla at sipwise.com  Mon Mar 11 07:29:59 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Mon, 11 Mar 2013 12:29:59 +0100
Subject: [Spce-user] RTCP, Media Monitoring, Media Stats, MOS, etc..
In-Reply-To: <CAMXUi2TBTRXLeYm=x=rUOSdBVzvS2dbWDyymsiLrCWi6RbLcAA@mail.gmail.com>
References: <CAMXUi2TBTRXLeYm=x=rUOSdBVzvS2dbWDyymsiLrCWi6RbLcAA@mail.gmail.com>
Message-ID: <20130311122959.2e5f00e7@multivac>

El Fri, 1 Mar 2013 09:34:31 -0700
Gary Nieboer <gary at netdiverse.com> escribi?:

> This is a request for an update on media monitoring / performance stats,
> RTCP, etc...
> 
> Although it would be great to have a reporting engine to capture
> information on each call and provide summary stats and details per call, it
> will still be good if SPCE stores jitter packet loss, etc.. for each leg of
> a call that can be found in a log somewhere. If there is a way to see real
> time stats on multiple legs like asterisk >sip show channelstats, that
> would be great too.
> 
> What is available with SPCE?
> 
> Thanks,
> 

At this moment, all the kamailio-mediaproxy-ng protocol is being rewritten to
provide the next generation of our media proxy relay server. It will be able to
provide/gather more information than the current release so providing acurate
and more complete stats of the media traffic will be possible in the future.
When? I don't know as the first version will be released in 2.8 and we'll
continue working on the new branch for future versions.

You can keep a eye on it following our github account. Patches and suggestions
are also welcomed!

https://github.com/sipwise/


cheers,

Jon



From jbonilla at sipwise.com  Mon Mar 11 07:37:02 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Mon, 11 Mar 2013 12:37:02 +0100
Subject: [Spce-user] Fwd: TLS problem
In-Reply-To: <3182F19C-DA78-4627-B891-4E430230C651@mac.com>
References: <DBDE31C2-0278-4CBC-920B-4DA2B18D54EC@me.com>
	<3182F19C-DA78-4627-B891-4E430230C651@mac.com>
Message-ID: <20130311123702.213c7d7b@multivac>

El Mon, 11 Mar 2013 09:31:23 +0100
Jirka Jirout <jirka.jirout at mac.com> escribi?:

> 
> HOWEVER:
> Today I decided to start from scratch and wrote a simple commmand line
> application that does only three things: opens a TLS connection, writes the
> REGISTER SIP message to the socket and prints the response (which should be
> something like 407 Unauthorized or something, but that would be fine).
> 
> The first two steps are fine - ssl_connect() and ssl_write() return success.
> But then the strange thins start to happen:
> 
> - No answer ever comes back
> - When I run ngrep-sip on the server, I do not see the message anywhere,
> although tcpdump shows it on the external interface

are you listening tcp port 5061?
> 
> The weirdest thing of course is that in about 10 % of cases, my new client
> connects just fine and works for hours without any problem. And that the
> older clients work fine.
> 
> my kamailio.cfg for lb looks like this in the tls section:
> 
> #!ifdef ENABLE_TLS
> loadmodule "tls.so"
> #!endif
> ....
> #!ifdef ENABLE_TLS
> modparam("tls", "certificate", "/crypto-keys/certificate.pem")
> modparam("tls", "private_key", "/crypto-keys/private.key")
> modparam("tls", "tls_method", "SSLv23")
> #!endif
> 
> SPCE is v 2.7, openssl 0.9.8.o
> 
> Any help would be greatly appreciated as this is slowly starting to drive me
> nuts ;-)
> 

I'll try to take a look when I have some more time but meanwhile I'd suggest
you to crosspost this to the kamailio-users mailing list as this issue could
match there and you might also get an answer there.




From jbonilla at sipwise.com  Mon Mar 11 07:43:34 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Mon, 11 Mar 2013 12:43:34 +0100
Subject: [Spce-user] Upper Registration to Asterisk - transfers not
 working
In-Reply-To: <CA+9+E=V-nzXP5CeQMCBKYkyp-oJ3xKiVmKgj5D9UpBoh_363pg@mail.gmail.com>
References: <CA+9+E=X_qiRReX=c_XubTD89HEV2HApxFw1yRDgDa2nW0o_SZg@mail.gmail.com>
	<CA+9+E=U215fg8=y_ff46U2pyUxB4imXi1-i1Sx3bMbtECZ+28A@mail.gmail.com>
	<51224FFD.4020408@sipwise.com> <51225420.5000607@sipwise.com>
	<CA+9+E=U-t1paf0KW+H-vHkOwGFBkbn1=8NFd5YUPdEwm_xyZsg@mail.gmail.com>
	<5122598B.1060703@sipwise.com>
	<CA+9+E=XeYZRHuBQkhsHBTMCarq2bOdX9c+93suqyCCb-Gj+3Ag@mail.gmail.com>
	<20130226152313.2d977e1d@quenya>
	<CA+9+E=WQUrn1HbeekF2fchdZ+7kX54Snphe0QY6W_x4oXN6yyA@mail.gmail.com>
	<CA+9+E=XRB-hPxK1ZTBU5ebhnaq-y0wS0=B=8ZaeVCheBC6+tWg@mail.gmail.com>
	<CA+9+E=WU=Sa9NTZ=HJrzLXZ+VnTON_--+9fBaHtivs+Lfh7Vwg@mail.gmail.com>
	<20130305075953.50b44f18@multivac>
	<CA+9+E=UKknSzZGHFp5qi2cLx=NdaZk879jQPQi_BvEY7ce-jAg@mail.gmail.com>
	<CA+9+E=UfxgtRPVjC_8BhcEm0kh2HSKnErhro7WrFJ-eLu8SzHA@mail.gmail.com>
	<CA+9+E=X3bfBdtmTSy=TycC=veshUU8Wgb8H6HCyEzjvW1RpSJg@mail.gmail.com>
	<CA+9+E=V-nzXP5CeQMCBKYkyp-oJ3xKiVmKgj5D9UpBoh_363pg@mail.gmail.com>
Message-ID: <20130311124334.7c2c0c06@multivac>

El Wed, 6 Mar 2013 18:28:59 +0200
Theo <axessofficetheo at gmail.com> escribi?:

> Tried with some other phones and it worked, so it seems specific to the
> SNOM phones. Of course we have over a thousand SNOM phones out there so for
> us to start using this we'd have to find a fix. My guess is that SNOM needs
> to fix it, but don't have the required ammo nor knowledge to take them to
> task on it (unless I am completely wrong to start off with). Aside from
> that, I doubt whether they would fix that in a matter of days and I had
> hoped to start implementing this as a solution.
> 
> Anything that can be done on the scpe to help with this? Any words of
> wisdom I can share with SNOM?
> 


Sorry Theo I'm away form office and can't follow all threads. Please let me
know:

- Does the REFER sent from the snom phone to the spce and the REFER sent from
  the SPCE to the Asterisk look the same? Any headers missing?

- Is the REFER working fine in case you directly register the snom against the
  asterisk box? 

- If so could you capture both traces and put them attached (not inline as the
  mail clients break the lines)




From jbonilla at sipwise.com  Mon Mar 11 07:50:29 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Mon, 11 Mar 2013 12:50:29 +0100
Subject: [Spce-user] SPCE and NAT
In-Reply-To: <20130225101944.74670dc3@quenya>
References: <0A699BDB-8395-454D-8FFA-81A20B8B5C13@optionsdsl.ca>
	<511B9CA5.5010406@sipwise.com>
	<CAOUeCaSCXbR3fJo8nyy9nORdOtuB_7QcuqajyVKw3mn+mQJ+kQ@mail.gmail.com>
	<20130220231132.588a2f58@quenya>
	<CAOUeCaSS=Dqxk-Du0h_NXwKTYvvHwb9=m9Pov9zieiZYbJOSNw@mail.gmail.com>
	<20130225101944.74670dc3@quenya>
Message-ID: <20130311125029.70f1ca36@multivac>

El Mon, 25 Feb 2013 10:19:44 +0100
Jon Bonilla (Manwe) <jbonilla at sipwise.com> escribi?:


> We don't have NOTIFY keepalive support in the released versions. OPTIONS
> keepalive support isn't the best neither. I'd say that we could backport this
> to released versions and I'll discuss it internally. But what do you (our
> beloved community) say about this?
> 


Just as information, this patch was released for versions 2.5, 2.6 and 2.7.

Aditionally some small fixes were also backported. I'd suggest running the apt
+ ngcpcfg combo.

Those of you running 2.4 or lower versions of the spce should consider
upgrading to 2.7 (or 2.6 at least) as those older versions won't be
mantained much longer.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130311/6566afe3/attachment.asc>

From jirka.jirout at mac.com  Mon Mar 11 12:27:03 2013
From: jirka.jirout at mac.com (Jirka Jirout)
Date: Mon, 11 Mar 2013 17:27:03 +0100
Subject: [Spce-user] TLS problem
Message-ID: <FB145375-D2D0-4DAC-AF84-090544413297@mac.com>

Hello,


>>  - No answer ever comes back
>>  - When I run ngrep-sip on the server, I do not see the message anywhere,
>>  although tcpdump shows it on the external interface
>> 
> 

> are you listening tcp port 5061?
I am not, we use a different port. But I do try to connect to the port on which TLS is enabled.

> I'll try to take a look when I have some more time but meanwhile I'd suggest
> you to crosspost this to the kamailio-users mailing list as this issue could
> match there and you might also get an answer there.
Thanks for suggestion, I will do that. If I do not manage to fix this soon, I will open a support ticket for this with you as well.

regards, jj


From mario.contreras at innovasur.es  Mon Mar 11 14:39:21 2013
From: mario.contreras at innovasur.es (Mario Contreras)
Date: Mon, 11 Mar 2013 19:39:21 +0100
Subject: [Spce-user] Avoid 302 response
Message-ID: <513E24D9.1060400@innovasur.es>

Hi all,

One of my subscribers has suffered an attack, and some calls have been 
accepted by sipwise because the subscriber sent an 302 message.

These are the messages:
SIP PEERING            SPCE


SPCE           SUBSCRIBER    SIP PEERING


Those are the two conversations, one between my sip peering and 
spce(it's an incoming call), and the other from spce to subscriber, but 
it redirects the call to another number with a 302.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130311/df4cfb9b/attachment.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: heabffjf.png
Type: image/png
Size: 3666 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130311/df4cfb9b/attachment.png>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: cdgfdfae.png
Type: image/png
Size: 6168 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130311/df4cfb9b/attachment-0001.png>

From mario.contreras at innovasur.es  Mon Mar 11 14:45:53 2013
From: mario.contreras at innovasur.es (Mario Contreras)
Date: Mon, 11 Mar 2013 19:45:53 +0100
Subject: [Spce-user] Avoid 302 response
In-Reply-To: <513E24D9.1060400@innovasur.es>
References: <513E24D9.1060400@innovasur.es>
Message-ID: <513E2661.7080804@innovasur.es>

Sorry, I didn't want to send... I hope you can help me. I've been 
looking for in kamailio proxy files, but I haven't find anything 
helpful(I imagine it's there but I don't know where...).
Thanks for all!

El 11/03/2013 19:39, Mario Contreras escribi?:
> Hi all,
>
> One of my subscribers has suffered an attack, and some calls have been 
> accepted by sipwise because the subscriber sent an 302 message.
>
> These are the messages:
> SIP PEERING            SPCE
>
>
> SPCE           SUBSCRIBER    SIP PEERING
>
>
> Those are the two conversations, one between my sip peering and 
> spce(it's an incoming call), and the other from spce to subscriber, 
> but it redirects the call to another number with a 302.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130311/4d5c3923/attachment.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: image/png
Size: 3666 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130311/4d5c3923/attachment.png>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: image/png
Size: 6168 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130311/4d5c3923/attachment-0001.png>

From mario.contreras at innovasur.es  Tue Mar 12 05:48:46 2013
From: mario.contreras at innovasur.es (Mario Contreras)
Date: Tue, 12 Mar 2013 10:48:46 +0100
Subject: [Spce-user] Avoid 302 response
In-Reply-To: <513E2F5B.4090707@sipwise.com>
References: <513E24D9.1060400@innovasur.es> <513E2661.7080804@innovasur.es>
	<513E2F5B.4090707@sipwise.com>
Message-ID: <513EF9FE.4060908@innovasur.es>

Thanks for your answer.

This is what I've done:

if(t_check_status("301|302"))
{
     xlog("L_ERROR", "Failed to fetch contact '$ct' from 301/302 - [% 
logreq -%]\n");
     acc_db_request("480", "acc");
     $var(announce_handle) = "callee_tmp_unavailable";
     $var(announce_set) = $avp(s:callee_sound_set);
     $(avp(s:announce_code)[*]) = 403;
     $(avp(s:announce_reason)[*]) = "Forbidden";
     route(ROUTE_EARLY_REJECT);
}

I think it will work. You think I can improve it?

Anyway, my spce treats the calls like they were incoming. Is that 
normal? Here I show an entry from cdr table:

*************************** 1. row ***************************
                                 id: 285697
                        update_time: 2013-03-10 05:09:27
                     source_user_id: 0
                 source_provider_id: 3
      source_external_subscriber_id:
        source_external_contract_id:
                  source_account_id: 0
                        source_user: myuser
                      source_domain: example.com
                         source_cli: CCCCCCCCCC
                        source_clir: 1
                destination_user_id: 0
            destination_provider_id: 3
destination_external_subscriber_id:
   destination_external_contract_id:
             destination_account_id: 0
                   destination_user: XXXXXXXXXX
                 destination_domain: 1.1.1.1
            destination_user_dialed: XXXXXXXXXX
                destination_user_in: XXXXXXXXXX
              destination_domain_in: example.com
                     peer_auth_user:
                    peer_auth_realm:
                          call_type: cfb
                        call_status: ok
                          call_code: 200
                          init_time: 1362888551.234
                         start_time: 1362888554.040
                           duration: 4.690
                            call_id: 
kif13x12kf0axb0kf1mc41ck102e3dkm at SoftX3000
                       carrier_cost: 0.00
                      reseller_cost: 0.00
                      customer_cost: 0.00
                  carrier_free_time: 0
                 reseller_free_time: 0
                 customer_free_time: 0
             carrier_billing_fee_id: NULL
            reseller_billing_fee_id: NULL
            customer_billing_fee_id: NULL
            carrier_billing_zone_id: NULL
           reseller_billing_zone_id: NULL
           customer_billing_zone_id: NULL
                frag_carrier_onpeak: NULL
               frag_reseller_onpeak: NULL
               frag_customer_onpeak: NULL
                      is_fragmented: NULL
                           rated_at: 2013-03-10 05:09:27
                      rating_status: ok
***************************************************************

Thank you!

El 11/03/2013 20:24, Andreas Granig escribi?:
> Hi,
>
> Search for "302" in your
> /etc/ngcp-config/templates/etc/kamailio/proxy/proxy.cfg and comment
> parts you don't want to be handled as is (preferably copy it to
> proxy.cfg.customtt.tt2 first and do your changes there to not get them
> overwritten with the next update).
>
> Andreas
>
> On 03/11/2013 07:45 PM, Mario Contreras wrote:
>> Sorry, I didn't want to send... I hope you can help me. I've been
>> looking for in kamailio proxy files, but I haven't find anything
>> helpful(I imagine it's there but I don't know where...).
>> Thanks for all!
>>
>> El 11/03/2013 19:39, Mario Contreras escribi?:
>>> Hi all,
>>>
>>> One of my subscribers has suffered an attack, and some calls have been
>>> accepted by sipwise because the subscriber sent an 302 message.
>>>
>>> These are the messages:
>>> SIP PEERING            SPCE
>>>
>>>
>>> SPCE           SUBSCRIBER    SIP PEERING
>>>
>>>
>>> Those are the two conversations, one between my sip peering and
>>> spce(it's an incoming call), and the other from spce to subscriber,
>>> but it redirects the call to another number with a 302.
>>
>>
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/listinfo/spce-user
>>




From agranig at sipwise.com  Tue Mar 12 06:28:55 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Tue, 12 Mar 2013 11:28:55 +0100
Subject: [Spce-user] Avoid 302 response
In-Reply-To: <513EF9FE.4060908@innovasur.es>
References: <513E24D9.1060400@innovasur.es> <513E2661.7080804@innovasur.es>
	<513E2F5B.4090707@sipwise.com> <513EF9FE.4060908@innovasur.es>
Message-ID: <513F0367.1060504@sipwise.com>

Hi,
On 03/12/2013 10:48 AM, Mario Contreras wrote:
> if(t_check_status("301|302"))
> {
>      xlog("L_ERROR", "Failed to fetch contact '$ct' from 301/302 - [%
> logreq -%]\n");
>      acc_db_request("480", "acc");
>      $var(announce_handle) = "callee_tmp_unavailable";
>      $var(announce_set) = $avp(s:callee_sound_set);
>      $(avp(s:announce_code)[*]) = 403;
>      $(avp(s:announce_reason)[*]) = "Forbidden";
>      route(ROUTE_EARLY_REJECT);
> }

Could work fine at first sight. Best is to try it out :)

> Anyway, my spce treats the calls like they were incoming. Is that
> normal? Here I show an entry from cdr table:
>
> *************************** 1. row ***************************
>                                  id: 285697
>                         update_time: 2013-03-10 05:09:27
>                      source_user_id: 0
>                  source_provider_id: 3
>       source_external_subscriber_id:
>         source_external_contract_id:
>                   source_account_id: 0
>                         source_user: myuser
>                       source_domain: example.com
>                          source_cli: CCCCCCCCCC
>                         source_clir: 1
>                 destination_user_id: 0
>             destination_provider_id: 3
> destination_external_subscriber_id:
>    destination_external_contract_id:
>              destination_account_id: 0
>                    destination_user: XXXXXXXXXX
>                  destination_domain: 1.1.1.1
>             destination_user_dialed: XXXXXXXXXX
>                 destination_user_in: XXXXXXXXXX
>               destination_domain_in: example.com
>                      peer_auth_user:
>                     peer_auth_realm:
>                           call_type: cfb

As you can see, the SPCE treats such call deflection as call-forward 
(call_type set to "cfb" instead of "call"). This is what you usually 
want, because if your subscriber configures his device to do a 
call-forward, you want this subscriber to pay for this call-forward, 
instead of charging it to the caller, because the caller doesn't know 
about this redirect.

It's the responsibility of the owner of the end device (if it's a 
bring-your-own-device service) or the platform operator (if you manage 
the devices of your customers) to keep it secure and not allow attackers 
to set it to arbitrary numbers.

Andreas



From mario.contreras at innovasur.es  Tue Mar 12 07:22:28 2013
From: mario.contreras at innovasur.es (Mario Contreras)
Date: Tue, 12 Mar 2013 12:22:28 +0100
Subject: [Spce-user] Avoid 302 response
In-Reply-To: <513F0367.1060504@sipwise.com>
References: <513E24D9.1060400@innovasur.es> <513E2661.7080804@innovasur.es>
	<513E2F5B.4090707@sipwise.com> <513EF9FE.4060908@innovasur.es>
	<513F0367.1060504@sipwise.com>
Message-ID: <513F0FF4.2080506@innovasur.es>

El 12/03/2013 11:28, Andreas Granig escribi?:
> As you can see, the SPCE treats such call deflection as call-forward
> (call_type set to "cfb" instead of "call"). This is what you usually
> want, because if your subscriber configures his device to do a
> call-forward, you want this subscriber to pay for this call-forward,
> instead of charging it to the caller, because the caller doesn't know
> about this redirect.
That's what I mean. Shouldn't the spce charge to my user? I think it 
should, but this call has almost five seconds of duration and the 
customer cost is zero.

> It's the responsibility of the owner of the end device (if it's a
> bring-your-own-device service) or the platform operator (if you manage
> the devices of your customers) to keep it secure and not allow attackers
> to set it to arbitrary numbers.
>
> Andreas

That's right of course. Thanks for your help! :)



From agranig at sipwise.com  Tue Mar 12 07:49:46 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Tue, 12 Mar 2013 12:49:46 +0100
Subject: [Spce-user] Avoid 302 response
In-Reply-To: <513F0FF4.2080506@innovasur.es>
References: <513E24D9.1060400@innovasur.es> <513E2661.7080804@innovasur.es>
	<513E2F5B.4090707@sipwise.com> <513EF9FE.4060908@innovasur.es>
	<513F0367.1060504@sipwise.com> <513F0FF4.2080506@innovasur.es>
Message-ID: <513F165A.8020204@sipwise.com>

Hi,

On 03/12/2013 12:22 PM, Mario Contreras wrote:
> El 12/03/2013 11:28, Andreas Granig escribi?:
>> As you can see, the SPCE treats such call deflection as call-forward
>> (call_type set to "cfb" instead of "call"). This is what you usually
>> want, because if your subscriber configures his device to do a
>> call-forward, you want this subscriber to pay for this call-forward,
>> instead of charging it to the caller, because the caller doesn't know
>> about this redirect.
> That's what I mean. Shouldn't the spce charge to my user? I think it
> should, but this call has almost five seconds of duration and the
> customer cost is zero.

You're right. Which version are you running? If 2.7, can you do an 
"apt-get update; apt-get upgrade; ngcpcfg apply" (might result in a 
short service down-time due to kamailio restart)? We've applied a fix 
for 2.7 for properly setting the account_id fields on 3xx around a month 
ago.

Andreas



From axessofficetheo at gmail.com  Tue Mar 12 10:30:23 2013
From: axessofficetheo at gmail.com (Theo)
Date: Tue, 12 Mar 2013 16:30:23 +0200
Subject: [Spce-user] Upper Registration to Asterisk - transfers not
	working
In-Reply-To: <20130311124334.7c2c0c06@multivac>
References: <CA+9+E=X_qiRReX=c_XubTD89HEV2HApxFw1yRDgDa2nW0o_SZg@mail.gmail.com>
	<CA+9+E=U215fg8=y_ff46U2pyUxB4imXi1-i1Sx3bMbtECZ+28A@mail.gmail.com>
	<51224FFD.4020408@sipwise.com> <51225420.5000607@sipwise.com>
	<CA+9+E=U-t1paf0KW+H-vHkOwGFBkbn1=8NFd5YUPdEwm_xyZsg@mail.gmail.com>
	<5122598B.1060703@sipwise.com>
	<CA+9+E=XeYZRHuBQkhsHBTMCarq2bOdX9c+93suqyCCb-Gj+3Ag@mail.gmail.com>
	<20130226152313.2d977e1d@quenya>
	<CA+9+E=WQUrn1HbeekF2fchdZ+7kX54Snphe0QY6W_x4oXN6yyA@mail.gmail.com>
	<CA+9+E=XRB-hPxK1ZTBU5ebhnaq-y0wS0=B=8ZaeVCheBC6+tWg@mail.gmail.com>
	<CA+9+E=WU=Sa9NTZ=HJrzLXZ+VnTON_--+9fBaHtivs+Lfh7Vwg@mail.gmail.com>
	<20130305075953.50b44f18@multivac>
	<CA+9+E=UKknSzZGHFp5qi2cLx=NdaZk879jQPQi_BvEY7ce-jAg@mail.gmail.com>
	<CA+9+E=UfxgtRPVjC_8BhcEm0kh2HSKnErhro7WrFJ-eLu8SzHA@mail.gmail.com>
	<CA+9+E=X3bfBdtmTSy=TycC=veshUU8Wgb8H6HCyEzjvW1RpSJg@mail.gmail.com>
	<CA+9+E=V-nzXP5CeQMCBKYkyp-oJ3xKiVmKgj5D9UpBoh_363pg@mail.gmail.com>
	<20130311124334.7c2c0c06@multivac>
Message-ID: <CA+9+E=XydUFsz0V85v4kGsZpcDZXs-asTzfetqKqmtUYegGCbQ@mail.gmail.com>

Hi Jon

I am away myself but as soon as I am back I will send the info.

thanks

Theo

On Mon, Mar 11, 2013 at 1:43 PM, Jon Bonilla <jbonilla at sipwise.com> wrote:

> El Wed, 6 Mar 2013 18:28:59 +0200
> Theo <axessofficetheo at gmail.com> escribi?:
>
> > Tried with some other phones and it worked, so it seems specific to the
> > SNOM phones. Of course we have over a thousand SNOM phones out there so
> for
> > us to start using this we'd have to find a fix. My guess is that SNOM
> needs
> > to fix it, but don't have the required ammo nor knowledge to take them to
> > task on it (unless I am completely wrong to start off with). Aside from
> > that, I doubt whether they would fix that in a matter of days and I had
> > hoped to start implementing this as a solution.
> >
> > Anything that can be done on the scpe to help with this? Any words of
> > wisdom I can share with SNOM?
> >
>
>
> Sorry Theo I'm away form office and can't follow all threads. Please let me
> know:
>
> - Does the REFER sent from the snom phone to the spce and the REFER sent
> from
>   the SPCE to the Asterisk look the same? Any headers missing?
>
> - Is the REFER working fine in case you directly register the snom against
> the
>   asterisk box?
>
> - If so could you capture both traces and put them attached (not inline as
> the
>   mail clients break the lines)
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130312/1c0045ee/attachment.html>

From pwilbert at gmail.com  Tue Mar 12 15:24:20 2013
From: pwilbert at gmail.com (Penn Wilbert)
Date: Tue, 12 Mar 2013 15:24:20 -0400
Subject: [Spce-user] Voicemail not answering on fresh SIP provider CE
	installation
Message-ID: <CABSTHcoXc=cAGKrndSF4P+innKYZUdEDayp5gy9-XabtccO1JQ@mail.gmail.com>

I loaded the sip provider CE VirtualBox VM on my server. I went through the
motions to add the system. I have added a few SIP devices to the system and
can make calls back and forth. However I cannot get voicemail to work. I
set one number to forward to Voicebox all the time and when I call the
number I get a fast beep tone and a eventual call failed.

I can log in to the web interfaces with the users and see the missed calls.
I cannot find the PIN for the user when they dial 2000 to get access to the
voicemail system, even though the system answers and asks for a phone and
PIN.

The asterisk server is running and the config on the server does state that
voicemail is yes.

Thank you for your time!

-- 
Penn Wilbert
pwilbert at gmail.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130312/d0aa28d0/attachment.html>

From agranig at sipwise.com  Tue Mar 12 15:40:22 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Tue, 12 Mar 2013 20:40:22 +0100
Subject: [Spce-user] WSDL without documentation tags
Message-ID: <513F84A6.4010901@sipwise.com>

Hi,

We've backported a change to the SOAP interface to 2.7 which allows you 
to fetch the WSDL without the inline documentation tags.

Some clients are known to choke on them (e.g. the stock SOAP client for 
PHP, where we recommended nusoap in the past, and also Visual Studio/C# 
and probably some more), so we added an URL parameter called "plain" to 
fetch the plain WSDL without a path to the stylesheet and without the 
documentation tags.

The previous URL

	https://<ip>:2443/SOAP/Provisioning.wsdl

works as usual and renders the online documentation in your browsers and 
can be used with most SOAP clients.

If this doesn't work, try

	https://<ip>:2443/SOAP/Provisioning.wsdl?plain

for the version without documentation in your SOAP client.

Do an "apt-get update && apt-get upgrade && ngcpcfg apply" to install 
this patch.

Andreas



From john.murray at skyracktelecom.com  Wed Mar 13 07:40:02 2013
From: john.murray at skyracktelecom.com (John Murray)
Date: Wed, 13 Mar 2013 11:40:02 -0000
Subject: [Spce-user] Voicemail config bug?
Message-ID: <026701ce1fdf$801e98e0$805bcaa0$@murray@skyracktelecom.com>

Hello,

 

If a subscriber is setup with diverts to voicemail and the subscribers DDI
is subsequently changed. This does not get reflected in the voicemail
setting and calls are forwarded to vmu-OLD-DDI- at voicebox.local which is
dropped:

 

asterisk[2961]: WARNING[18132]: app_voicemail.c:3972 in leave_voicemail: No
entry in voicemail config file for '-OLD-DDI-'

 

I can fix this by deleting the voicemail destination set and re-adding it.

 

Can this be fixed?

 

Best regards

 

John

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130313/0ce82ab9/attachment.html>

From dgrotti at sipwise.com  Wed Mar 13 07:45:39 2013
From: dgrotti at sipwise.com (Daniel Grotti)
Date: Wed, 13 Mar 2013 12:45:39 +0100
Subject: [Spce-user] Voicemail not answering on fresh SIP provider CE
 installation
In-Reply-To: <CABSTHcoXc=cAGKrndSF4P+innKYZUdEDayp5gy9-XabtccO1JQ@mail.gmail.com>
References: <CABSTHcoXc=cAGKrndSF4P+innKYZUdEDayp5gy9-XabtccO1JQ@mail.gmail.com>
Message-ID: <514066E3.2050602@sipwise.com>

Hi Penn,
Can you see the call arrives in Asterisk ?
Try to connect to Asterisk (asterisk -vvvvvr) and then make a call and
see what happens.

Could you also provide us with some logs ?

- /var/log/ngcp/kamailio-proxy.log


Thanks,
Daniel


On 03/12/2013 08:24 PM, Penn Wilbert wrote:
> I loaded the sip provider CE VirtualBox VM on my server. I went
> through the motions to add the system. I have added a few SIP devices
> to the system and can make calls back and forth. However I cannot get
> voicemail to work. I set one number to forward to Voicebox all the
> time and when I call the number I get a fast beep tone and a eventual
> call failed.
>
> I can log in to the web interfaces with the users and see the missed
> calls. I cannot find the PIN for the user when they dial 2000 to get
> access to the voicemail system, even though the system answers and
> asks for a phone and PIN.
>
> The asterisk server is running and the config on the server does state
> that voicemail is yes.
>
> Thank you for your time!
>
> -- 
> Penn Wilbert
> pwilbert at gmail.com <mailto:pwilbert at gmail.com>
>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130313/9520f885/attachment.html>

From pwilbert at gmail.com  Wed Mar 13 07:49:25 2013
From: pwilbert at gmail.com (Penn Wilbert)
Date: Wed, 13 Mar 2013 07:49:25 -0400
Subject: [Spce-user] Voicemail not answering on fresh SIP provider CE
	installation
In-Reply-To: <514066E3.2050602@sipwise.com>
References: <CABSTHcoXc=cAGKrndSF4P+innKYZUdEDayp5gy9-XabtccO1JQ@mail.gmail.com>
	<514066E3.2050602@sipwise.com>
Message-ID: <CABSTHcrRPFmk5PUh-OHjU8PmzvzFi+DeWgRiLOZAsT_hh0AASA@mail.gmail.com>

Daniel,

I'll update in a few hours about the status of this issue. I found the
archives after I had sent the email. In the archives I found a similar
issue happening to a user using a fresh install of 2.7 CE. We had not
updated the VM yet so I'll be testing shortly as to whether the issue still
happens or not after the update. If it doesn't work I'll send logs.


On Wed, Mar 13, 2013 at 7:45 AM, Daniel Grotti <dgrotti at sipwise.com> wrote:

>  Hi Penn,
> Can you see the call arrives in Asterisk ?
> Try to connect to Asterisk (asterisk -vvvvvr) and then make a call and see
> what happens.
>
> Could you also provide us with some logs ?
>
> - /var/log/ngcp/kamailio-proxy.log
>
>
> Thanks,
> Daniel
>
>
>
> On 03/12/2013 08:24 PM, Penn Wilbert wrote:
>
>  I loaded the sip provider CE VirtualBox VM on my server. I went through
> the motions to add the system. I have added a few SIP devices to the system
> and can make calls back and forth. However I cannot get voicemail to work.
> I set one number to forward to Voicebox all the time and when I call the
> number I get a fast beep tone and a eventual call failed.
>
> I can log in to the web interfaces with the users and see the missed
> calls. I cannot find the PIN for the user when they dial 2000 to get access
> to the voicemail system, even though the system answers and asks for a
> phone and PIN.
>
>  The asterisk server is running and the config on the server does state
> that voicemail is yes.
>
>  Thank you for your time!
>
> --
> Penn Wilbert
> pwilbert at gmail.com
>
>
> _______________________________________________
> Spce-user mailing listSpce-user at lists.sipwise.comhttp://lists.sipwise.com/listinfo/spce-user
>
>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
>
>


-- 
Penn Wilbert
pwilbert at gmail.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130313/6bfe2bc5/attachment.html>

From john.murray at skyracktelecom.com  Wed Mar 13 07:54:04 2013
From: john.murray at skyracktelecom.com (John Murray)
Date: Wed, 13 Mar 2013 11:54:04 -0000
Subject: [Spce-user] Domain prefs for subscribers with 'Trusted Source'
Message-ID: <027e01ce1fe1$75bf5860$613e0920$@murray@skyracktelecom.com>

Hello,

 

Subscribers making calls from devices connected via 'Trusted Source' don't
seem to load the domain preferences of the subscriber but atempt to load
domain preferences of the incomming 'From: <domain part>' header.

Is this correct?

 

Regards

 

John

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130313/aad7b607/attachment.html>

From apogrebennyk at sipwise.com  Wed Mar 13 08:00:58 2013
From: apogrebennyk at sipwise.com (Andrew Pogrebennyk)
Date: Wed, 13 Mar 2013 13:00:58 +0100
Subject: [Spce-user] Voicemail config bug?
In-Reply-To: <026701ce1fdf$801e98e0$805bcaa0$@murray@skyracktelecom.com>
References: <026701ce1fdf$801e98e0$805bcaa0$@murray@skyracktelecom.com>
Message-ID: <51406A7A.2090807@sipwise.com>

Hi John,

On 03/13/2013 12:40 PM, John Murray wrote:
> If a subscriber is setup with diverts to voicemail and the subscribers
> DDI is subsequently changed. This does not get reflected in the
> voicemail setting and calls are forwarded to vmu-OLD-DDI- at voicebox.local
> <mailto:vmu-OLD-DDI- at voicebox.local> which is dropped:

This is already in our bugtracker. I will notify you when this gets fixed.

Regards,
Andrew



From john.murray at skyracktelecom.com  Wed Mar 13 08:03:49 2013
From: john.murray at skyracktelecom.com (John Murray)
Date: Wed, 13 Mar 2013 12:03:49 -0000
Subject: [Spce-user] Voicemail config bug?
In-Reply-To: <51406A7A.2090807@sipwise.com>
References: <026701ce1fdf$801e98e0$805bcaa0$@murray@skyracktelecom.com>
	<51406A7A.2090807@sipwise.com>
Message-ID: <51406b26.8f4bb40a.2fee.ffffb73d@mx.google.com>

Many thanks.

-----Original Message-----
From: Andrew Pogrebennyk [mailto:apogrebennyk at sipwise.com] 
Sent: 13 March 2013 12:01
To: John Murray
Cc: spce-user at lists.sipwise.com
Subject: Re: [Spce-user] Voicemail config bug?

Hi John,

On 03/13/2013 12:40 PM, John Murray wrote:
> If a subscriber is setup with diverts to voicemail and the subscribers 
> DDI is subsequently changed. This does not get reflected in the 
> voicemail setting and calls are forwarded to 
> vmu-OLD-DDI- at voicebox.local <mailto:vmu-OLD-DDI- at voicebox.local> which is
dropped:

This is already in our bugtracker. I will notify you when this gets fixed.

Regards,
Andrew




From csaroli at isysp.com  Wed Mar 13 11:52:10 2013
From: csaroli at isysp.com (Christopher Saroli)
Date: Wed, 13 Mar 2013 15:52:10 +0000
Subject: [Spce-user]  Billing Profiles/Fees
Message-ID: <EE7149FDD60B3045A4377F05AAFFF3F118342A46@ISS-Server.ISS.local>

Little confused on this, every call I make/receive does not get a fee applied to it.  Just trying right now to apply a fee to all calls.  I might be totally wrong here, any info would be appreciated.

This is my config:

Billing Profile:
Interval Charge:  0.02
Freetime: 	  0
Free Cash: 	  0
Monthly Lim	  0
Lock		  none
Currency: 	 $


Bililng Fees
Destination:  	.*
Zone:      	All
Init Rate:	1
Init interval:	1
Follow Rate: 	1
Follow interval: 1


Also I have no outgoing calls recorded in the CDR under the admin interface but in the customer portal they are there. Is this by design or is something wrong.

Thanks




From Stephen at newbreakcommunications.com  Wed Mar 13 12:25:05 2013
From: Stephen at newbreakcommunications.com (Stephen Donovan)
Date: Wed, 13 Mar 2013 16:25:05 +0000
Subject: [Spce-user] random registration dropping
Message-ID: <2CDACF784B10A14EAAFAB6140DAE0D132409FC@EXCHANGE1.newbreakmanagement.local>

We have a few customers that are complaining of random dropped calls and no dialtone for short periods of time.  It has not happened for a long enough period of time to even nail it down.  Are there any known issues ?  I'm not certain these are the only ones it was happening to but the customers  that called were both using Linksys SPA8000 ATAs registered directly to spce.

I can provide any logs needed from the affected period of time.

Stephen

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130313/f43eaa3a/attachment.html>

From jcifre at ib-red.com  Wed Mar 13 12:36:34 2013
From: jcifre at ib-red.com (Joan Cifre (ibred))
Date: Wed, 13 Mar 2013 17:36:34 +0100
Subject: [Spce-user] random registration dropping
In-Reply-To: <2CDACF784B10A14EAAFAB6140DAE0D132409FC@EXCHANGE1.newbreakmanagement.local>
References: <2CDACF784B10A14EAAFAB6140DAE0D132409FC@EXCHANGE1.newbreakmanagement.local>
Message-ID: <5140AB12.60301@ib-red.com>

In Linksys configuration try to disable keepalives for nat.


El 2013-03-13 17:25, Stephen Donovan escribi?:
>
> We have a few customers that are complaining of random dropped calls 
> and no dialtone for short periods of time.  It has not happened for a 
> long enough period of time to even nail it down.  Are there any known 
> issues ? I'm not certain these are the only ones it was happening to 
> but the customers  that called were both using Linksys SPA8000 ATAs 
> registered directly to spce.
>
> I can provide any logs needed from the affected period of time.
>
> Stephen
>
>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130313/6ce1d8d7/attachment.html>

From Stephen at newbreakcommunications.com  Wed Mar 13 13:05:45 2013
From: Stephen at newbreakcommunications.com (Stephen Donovan)
Date: Wed, 13 Mar 2013 17:05:45 +0000
Subject: [Spce-user] random registration dropping
In-Reply-To: <5140AB12.60301@ib-red.com>
References: <2CDACF784B10A14EAAFAB6140DAE0D132409FC@EXCHANGE1.newbreakmanagement.local>
	<5140AB12.60301@ib-red.com>
Message-ID: <2CDACF784B10A14EAAFAB6140DAE0D13240B62@EXCHANGE1.newbreakmanagement.local>

Just verified no NAT keepalive enabled on any line.

Stephen


From: Joan Cifre (ibred) [mailto:jcifre at ib-red.com]
Sent: Wednesday, March 13, 2013 11:37 AM
To: Stephen Donovan
Cc: 'spce-user at lists.sipwise.com'
Subject: Re: [Spce-user] random registration dropping

In Linksys configuration try to disable keepalives for nat.

El 2013-03-13 17:25, Stephen Donovan escribi?:
We have a few customers that are complaining of random dropped calls and no dialtone for short periods of time.  It has not happened for a long enough period of time to even nail it down.  Are there any known issues ?  I'm not certain these are the only ones it was happening to but the customers  that called were both using Linksys SPA8000 ATAs registered directly to spce.

I can provide any logs needed from the affected period of time.

Stephen





_______________________________________________

Spce-user mailing list

Spce-user at lists.sipwise.com<mailto:Spce-user at lists.sipwise.com>

http://lists.sipwise.com/listinfo/spce-user

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130313/6ffee1e6/attachment.html>

From yyicur4me at gmail.com  Wed Mar 13 10:41:34 2013
From: yyicur4me at gmail.com (V)
Date: Wed, 13 Mar 2013 10:41:34 -0400
Subject: [Spce-user] Initial registration fails W/malformed sip
Message-ID: <CALwLLOC0sDheu-5U-Kiv3gQzeTsp2yfjCScjgmtDMGL-WMt5_Q@mail.gmail.com>

Endpoint fails to register.

My logs show the domain coming through incorrect
I suspect something is setup incorrect at server

sip user ID: ryandavis
domain: telific.com
ngcp server ip: 98.103.105.117
ngcp server DNS name ash-butler
ngcp server FQDN: ash-butler.telific.com

It seems to be appending the DNS name to the user ID but I have no idea why

Any help will be greatly appreciated.
Please advise
Thanks
Vince


Mar 13 10:28:15 ash-bulter /usr/sbin/kamailio[1606]: INFO: <script>: New
request - M=OPTIONS R=sip:nagios at voip.sipwise.local F=
sip:sipsak at 98.103.105.117:6988 T=sip:nagios at voip.sipwise.local IP=udp:
127.0.0.1:6988 ID=239285313 at 98.103.105.117
Mar 13 10:28:15 ash-bulter /usr/sbin/kamailio[1606]: INFO: <script>: NAT
check disabled by administrator for source IP '127.0.0.1' - M=OPTIONS
R=sip:nagios at voip.sipwise.local
F=sip:sipsak at 98.103.105.117:6988T=sip:nagios at voip.sipwise.localIP=
127.0.0.1:6988 ID=239285313 at 98.103.105.117
Mar 13 10:28:15 ash-bulter /usr/sbin/kamailio[1606]: INFO: <script>:
Relaying request, du='sip:127.0.0.1:5062', fs='udp:127.0.0.1:5060' -
R=sip:nagios at voip.sipwise.local ID=239285313 at 98.103.105.117
Mar 13 10:28:15 ash-bulter /usr/sbin/kamailio[1604]: INFO: <script>: Reply
from Inbound - S=100 - Trying M=OPTIONS IP=udp:127.0.0.1:5062
ID=239285313 at 98.103.105.117
Mar 13 10:28:15 ash-bulter /usr/sbin/kamailio[1604]: INFO: <script>:
Sending reply, fs='udp:127.0.0.1:5060' - ID=239285313 at 98.103.105.117
Mar 13 10:28:15 ash-bulter /usr/sbin/kamailio[1602]: INFO: <script>: Reply
from Inbound - S=200 - Alive M=OPTIONS IP=udp:127.0.0.1:5062
ID=239285313 at 98.103.105.117
Mar 13 10:28:15 ash-bulter /usr/sbin/kamailio[1602]: INFO: <script>:
Sending reply, fs='udp:127.0.0.1:5060' - ID=239285313 at 98.103.105.117
Mar 13 10:28:21 ash-bulter /usr/sbin/kamailio[1593]: WARNING: sanity
[sanity.c:800]: sanity_check(): check_parse_uris(): failed to parse From uri
Mar 13 10:28:21 ash-bulter /usr/sbin/kamailio[1593]: WARNING: <script>:
Malformed SIP message detected - M=REGISTER
R=sip:ash-butler.telific.comF=sip:ryandavis at telific.com
@ash-butler.telific.com
T=sip:ryandavis at telific.com@ash-butler.telific.comIP=udp:
173.89.168.17:5060 ID=1136074965-5060-1 at BJC.BGI.A.C
Mar 13 10:28:25 ash-bulter /usr/sbin/kamailio[1601]: INFO: <script>: New
request - M=OPTIONS R=sip:nagios at voip.sipwise.local F=
sip:sipsak at 98.103.105.117:8393 T=sip:nagios at voip.sipwise.local IP=udp:
127.0.0.1:8393 ID=1964588741 at 98.103.105.117
Mar 13 10:28:25 ash-bulter /usr/sbin/kamailio[1601]: INFO: <script>: NAT
check disabled by administrator for source IP '127.0.0.1' - M=OPTIONS
R=sip:nagios at voip.sipwise.local
F=sip:sipsak at 98.103.105.117:8393T=sip:nagios at voip.sipwise.localIP=
127.0.0.1:8393 ID=1964588741 at 98.103.105.117
Mar 13 10:28:25 ash-bulter /usr/sbin/kamailio[1601]: INFO: <script>:
Relaying request, du='sip:127.0.0.1:5062', fs='udp:127.0.0.1:5060' -
R=sip:nagios at voip.sipwise.local ID=1964588741 at 98.103.105.117
Mar 13 10:28:25 ash-bulter /usr/sbin/kamailio[1599]: INFO: <script>: Reply
from Inbound - S=100 - Trying M=OPTIONS IP=udp:127.0.0.1:5062
ID=1964588741 at 98.103.105.117
Mar 13 10:28:25 ash-bulter /usr/sbin/kamailio[1599]: INFO: <script>:
Sending reply, fs='udp:127.0.0.1:5060' - ID=1964588741 at 98.103.105.117
Mar 13 10:28:25 ash-bulter /usr/sbin/kamailio[1608]: INFO: <script>: Reply
from Inbound - S=200 - Alive M=OPTIONS IP=udp:127.0.0.1:5062
ID=1964588741 at 98.103.105.117
Mar 13 10:28:25 ash-bulter /usr/sbin/kamailio[1608]: INFO: <script>:
Sending reply, fs='udp:127.0.0.1:5060' - ID=1964588741 at 98.103.105.117
Mar 13 10:28:35 ash-bulter /usr/sbin/kamailio[1600]: INFO: <script>: New
request - M=OPTIONS R=sip:nagios at voip.sipwise.local F=
sip:sipsak at 98.103.105.117:6584 T=sip:nagios at voip.sipwise.local IP=udp:
127.0.0.1:6584 ID=1724643046 at 98.103.105.117
Mar 13 10:28:35 ash-bulter /usr/sbin/kamailio[1600]: INFO: <script>: NAT
check disabled by administrator for source IP '127.0.0.1' - M=OPTIONS
R=sip:nagios at voip.sipwise.local
F=sip:sipsak at 98.103.105.117:6584T=sip:nagios at voip.sipwise.localIP=
127.0.0.1:6584 ID=1724643046 at 98.103.105.117
Mar 13 10:28:35 ash-bulter /usr/sbin/kamailio[1600]: INFO: <script>:
Relaying request, du='sip:127.0.0.1:5062', fs='udp:127.0.0.1:5060' -
R=sip:nagios at voip.sipwise.local ID=1724643046 at 98.103.105.117
Mar 13 10:28:35 ash-bulter /usr/sbin/kamailio[1603]: INFO: <script>: Reply
from Inbound - S=100 - Trying M=OPTIONS IP=udp:127.0.0.1:5062
ID=1724643046 at 98.103.105.117
Mar 13 10:28:35 ash-bulter /usr/sbin/kamailio[1603]: INFO: <script>:
Sending reply, fs='udp:127.0.0.1:5060' - ID=1724643046 at 98.103.105.117
Mar 13 10:28:35 ash-bulter /usr/sbin/kamailio[1603]: INFO: <script>: Reply
from Inbound - S=200 - Alive M=OPTIONS IP=udp:127.0.0.1:5062
ID=1724643046 at 98.103.105.117
Mar 13 10:28:35 ash-bulter /usr/sbin/kamailio[1603]: INFO: <script>:
Sending reply, fs='udp:127.0.0.1:5060' - ID=1724643046 at 98.103.105.117


-- 

*mY iRQs aRe cOnfLicTiNg ?*
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130313/001aa2f3/attachment.html>

From agranig at sipwise.com  Wed Mar 13 14:59:05 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Wed, 13 Mar 2013 19:59:05 +0100
Subject: [Spce-user] Initial registration fails W/malformed sip
In-Reply-To: <CALwLLOC0sDheu-5U-Kiv3gQzeTsp2yfjCScjgmtDMGL-WMt5_Q@mail.gmail.com>
References: <CALwLLOC0sDheu-5U-Kiv3gQzeTsp2yfjCScjgmtDMGL-WMt5_Q@mail.gmail.com>
Message-ID: <5140CC79.9080309@sipwise.com>

Hi,

On 03/13/2013 03:41 PM, V wrote:
> Endpoint fails to register.
> My logs show the domain coming through incorrect
> I suspect something is setup incorrect at server

No, it's a misconfiguration of your client. Probably you're using a 
client with a username and domain field, and you put 
ryandavis at telific.com into the username, because your client is sending 
sip:ryandavis at telific.com@ash-butler.telific.com.

In any case, if you use telific.com as domain, but the server is 
reachable only via ash-butler.telific.com, you need to use ryndavis as 
username, telific.com as domain, and ash-butler.telific.com as 
outbound-proxy/registrar.

Andreas



From john.murray at skyracktelecom.com  Wed Mar 13 16:15:49 2013
From: john.murray at skyracktelecom.com (John Murray)
Date: Wed, 13 Mar 2013 20:15:49 -0000
Subject: [Spce-user] Delete subscriber does not delete 'trusted' entry
Message-ID: <035601ce2027$8e13aab0$aa3b0010$@murray@skyracktelecom.com>

Hello,

 

In v2.7 when I subscriber is deleted the entry in the trusted table is not
deleted therefore when adding the same pattern to a new subscriber the
original (deleted) subscriber is matched.

 

Regards

 

John

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130313/a0a6d65e/attachment.html>

From pwilbert at gmail.com  Wed Mar 13 16:18:54 2013
From: pwilbert at gmail.com (Penn Wilbert)
Date: Wed, 13 Mar 2013 16:18:54 -0400
Subject: [Spce-user] Voicemail not answering on fresh SIP provider CE
	installation
In-Reply-To: <CABSTHcrRPFmk5PUh-OHjU8PmzvzFi+DeWgRiLOZAsT_hh0AASA@mail.gmail.com>
References: <CABSTHcoXc=cAGKrndSF4P+innKYZUdEDayp5gy9-XabtccO1JQ@mail.gmail.com>
	<514066E3.2050602@sipwise.com>
	<CABSTHcrRPFmk5PUh-OHjU8PmzvzFi+DeWgRiLOZAsT_hh0AASA@mail.gmail.com>
Message-ID: <CABSTHcpCp_LqdJfwnjZk2k6UPZxWBGbprLDy447FGg6GjEnMEA@mail.gmail.com>

Running updates against the server resolved the issue


On Wed, Mar 13, 2013 at 7:49 AM, Penn Wilbert <pwilbert at gmail.com> wrote:

> Daniel,
>
> I'll update in a few hours about the status of this issue. I found the
> archives after I had sent the email. In the archives I found a similar
> issue happening to a user using a fresh install of 2.7 CE. We had not
> updated the VM yet so I'll be testing shortly as to whether the issue still
> happens or not after the update. If it doesn't work I'll send logs.
>
>
> On Wed, Mar 13, 2013 at 7:45 AM, Daniel Grotti <dgrotti at sipwise.com>wrote:
>
>>  Hi Penn,
>> Can you see the call arrives in Asterisk ?
>> Try to connect to Asterisk (asterisk -vvvvvr) and then make a call and
>> see what happens.
>>
>> Could you also provide us with some logs ?
>>
>> - /var/log/ngcp/kamailio-proxy.log
>>
>>
>> Thanks,
>> Daniel
>>
>>
>>
>> On 03/12/2013 08:24 PM, Penn Wilbert wrote:
>>
>>  I loaded the sip provider CE VirtualBox VM on my server. I went through
>> the motions to add the system. I have added a few SIP devices to the system
>> and can make calls back and forth. However I cannot get voicemail to work.
>> I set one number to forward to Voicebox all the time and when I call the
>> number I get a fast beep tone and a eventual call failed.
>>
>> I can log in to the web interfaces with the users and see the missed
>> calls. I cannot find the PIN for the user when they dial 2000 to get access
>> to the voicemail system, even though the system answers and asks for a
>> phone and PIN.
>>
>>  The asterisk server is running and the config on the server does state
>> that voicemail is yes.
>>
>>  Thank you for your time!
>>
>> --
>> Penn Wilbert
>> pwilbert at gmail.com
>>
>>
>> _______________________________________________
>> Spce-user mailing listSpce-user at lists.sipwise.comhttp://lists.sipwise.com/listinfo/spce-user
>>
>>
>>
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/listinfo/spce-user
>>
>>
>
>
> --
> Penn Wilbert
> pwilbert at gmail.com
>



-- 
Penn Wilbert
pwilbert at gmail.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130313/f1ca7968/attachment.html>

From jbonilla at sipwise.com  Wed Mar 13 18:02:43 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Wed, 13 Mar 2013 23:02:43 +0100
Subject: [Spce-user] Domain prefs for subscribers with 'Trusted Source'
In-Reply-To: <027e01ce1fe1$75bf5860$613e0920$@murray@skyracktelecom.com>
References: <027e01ce1fe1$75bf5860$613e0920$@murray@skyracktelecom.com>
Message-ID: <20130313230243.470b266c@quenya>

El Wed, 13 Mar 2013 11:54:04 -0000
"John Murray" <john.murray at skyracktelecom.com> escribi?:

> Hello,
> 
>  
> 
> Subscribers making calls from devices connected via 'Trusted Source' don't
> seem to load the domain preferences of the subscriber but atempt to load
> domain preferences of the incomming 'From: <domain part>' header.
> 
> Is this correct?
> 

I'd say yes. When a source is trusted you don't check which domain it belongs to
as you would usually do with a "normal" subscriber. You just trust what it
sends. 

Maybe this could be reviewed. Is this a desirable behaviour? It wouldn't be
difficult to check and load the real domain.



-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130313/eb2a0393/attachment.asc>

From jbonilla at sipwise.com  Wed Mar 13 18:05:20 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Wed, 13 Mar 2013 23:05:20 +0100
Subject: [Spce-user] random registration dropping
In-Reply-To: <5140AB12.60301@ib-red.com>
References: <2CDACF784B10A14EAAFAB6140DAE0D132409FC@EXCHANGE1.newbreakmanagement.local>
	<5140AB12.60301@ib-red.com>
Message-ID: <20130313230520.5be292fb@quenya>

El Wed, 13 Mar 2013 17:36:34 +0100
"Joan Cifre (ibred)" <jcifre at ib-red.com> escribi?:

> In Linksys configuration try to disable keepalives for nat.
>

Nat keepalives sent by the clients are now handled by the spce without banning
the user. Unless they send too many and the ip is banned :)


-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130313/77fb5e2a/attachment.asc>

From jbonilla at sipwise.com  Wed Mar 13 18:06:57 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Wed, 13 Mar 2013 23:06:57 +0100
Subject: [Spce-user] random registration dropping
In-Reply-To: <2CDACF784B10A14EAAFAB6140DAE0D132409FC@EXCHANGE1.newbreakmanagement.local>
References: <2CDACF784B10A14EAAFAB6140DAE0D132409FC@EXCHANGE1.newbreakmanagement.local>
Message-ID: <20130313230657.7eb033ac@quenya>

El Wed, 13 Mar 2013 16:25:05 +0000
Stephen Donovan <Stephen at newbreakcommunications.com> escribi?:

> We have a few customers that are complaining of random dropped calls and no
> dialtone for short periods of time.  It has not happened for a long enough
> period of time to even nail it down.  Are there any known issues ?  I'm not
> certain these are the only ones it was happening to but the customers  that
> called were both using Linksys SPA8000 ATAs registered directly to spce.
> 
> I can provide any logs needed from the affected period of time.
> 
> Stephen
> 

no known issue that I know. We would need to see a log matching such a call.
Kamailio-proxy as starting point as usual and then we could check how/why/where
the call was dropped.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130313/c179e508/attachment.asc>

From gavin.sweet at skyracktelecom.com  Wed Mar 13 19:56:03 2013
From: gavin.sweet at skyracktelecom.com (Gavin Sweet)
Date: Wed, 13 Mar 2013 23:56:03 -0000
Subject: [Spce-user] Display of fee zone in subscriber CDR details GUI
Message-ID: <051801ce2046$52671550$f7353ff0$@sweet@skyracktelecom.com>

Hi all -

 

Trying to get description strings for zones/zone details rather than ID's in
the CDR details GUI..

 

Problem is that in the web admin interface, looking at an individual
subscriber, when I click the "Details" link next to a specific call, the
pop-over box displays carrier_billing_zone_id and customer_billing_zone_id
as integers, rather than pulling back the actual text. 

I can go look these up in the DB, but it makes the GUI much less useful for
doing quick admin or test checks.

Is this a bug, something fixable? Where can I find the code for this to see
what the issue might be?

 

Many thanks,

Gavin

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130313/f1838c15/attachment.html>

From nh.developer at gmail.com  Thu Mar 14 07:34:17 2013
From: nh.developer at gmail.com (Developing Tomorrow)
Date: Thu, 14 Mar 2013 12:34:17 +0100
Subject: [Spce-user] media packets not travelling properly
Message-ID: <00ba01ce20a7$dd924120$98b6c360$@gmail.com>

Hi.,

 

This is my first mail to this mailing list, so please forgive my erroneous
procedure if it might be.

 

I have installed sipwise CE, with some pains but it does the job.

If I use a provider connected over normal line (any normal sip provider)

The signalling is done properly I can see the messages, but then the audio
is not following.

 

By that I mean I cannot hear the ringtone nor can I hear the other party
talking, nor they can hear me.

 

I have tried over a vpn, and I have one way audio.

 

The system is installed on a public server with the default iptables.

I have created only 1 domain that receive the external connections on port
5060

 

On the other hand, I have a vpn conection to a foreign network.

Client-----XXXXX (public ip) ----------provider********************* no
audio nor A nor B leg.

Client-----XXXXX(public ip)----------VPN_start
(10.8.0.1)-------------VPN_end(10.8.0.6)-----------provider (172.26.10.2
[not real])******** 1 way audio, they do not hear me

 

Anybody can shade a light here?

 

Thx

herve

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130314/ce2315ab/attachment.html>

From nh.developer at gmail.com  Thu Mar 14 07:26:41 2013
From: nh.developer at gmail.com (Developing Tomorrow)
Date: Thu, 14 Mar 2013 12:26:41 +0100
Subject: [Spce-user] media packets not travelling properly
Message-ID: <00b501ce20a6$cdbdefc0$6939cf40$@gmail.com>

Hi.,

 

This is my first mail to this mailing list, so please forgive my erroneous
procedure if it might be.

 

I have installed sipwise CE, with some pains but it does the job.

If I use a provider connected over normal line (any normal sip provider)

The signalling is done properly I can see the messages, but then the audio
is not following.

 

By that I mean I cannot hear the ringtone nor can I hear the other party
talking, nor they can hear me.

 

I have tried over a vpn, and I have one way audio.

 

The system is installed on a public server with the default iptables.

I have created only 1 domain that receive the external connections on port
5060

 

On the other hand, I have a vpn conection to a foreign network.

Client-----XXXXX (public ip) ----------provider********************* no
audio nor A nor B leg.

Client-----XXXXX(public ip)----------VPN_start
(10.8.0.1)-------------VPN_end(10.8.0.6)-----------provider (172.26.10.2
[not real])******** 1 way audio, they do not hear me

 

Anybody can shade a light here?

 

Thx

herve

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130314/07c29be2/attachment.html>

From dgrotti at sipwise.com  Thu Mar 14 07:49:01 2013
From: dgrotti at sipwise.com (Daniel Grotti)
Date: Thu, 14 Mar 2013 12:49:01 +0100
Subject: [Spce-user] media packets not travelling properly
In-Reply-To: <00ba01ce20a7$dd924120$98b6c360$@gmail.com>
References: <00ba01ce20a7$dd924120$98b6c360$@gmail.com>
Message-ID: <5141B92D.7010105@sipwise.com>

On 03/14/2013 12:34 PM, Developing Tomorrow wrote:
>
> Hi.,
>
>  
>
> This is my first mail to this mailing list, so please forgive my
> erroneous procedure if it might be.
>
>  
>
> I have installed sipwise CE, with some pains but it does the job.
>
> If I use a provider connected over normal line (any normal sip provider)
>
> The signalling is done properly I can see the messages, but then the
> audio is not following.
>
>  
>
> By that I mean I cannot hear the ringtone nor can I hear the other
> party talking, nor they can hear me.
>
>  
>
> I have tried over a vpn, and I have one way audio.
>
>  
>
> The system is installed on a public server with the default iptables.
>
> I have created only 1 domain that receive the external connections on
> port 5060
>
>  
>
> On the other hand, I have a vpn conection to a foreign network.
>
> Client-----XXXXX (public ip) ----------provider*********************
> no audio nor A nor B leg.
>
> Client-----XXXXX(public ip)----------VPN_start
> (10.8.0.1)-------------VPN_end(10.8.0.6)-----------provider
> (172.26.10.2 [not real])******** 1 way audio, they do not hear me
>
>  
>
> Anybody can shade a light here?
>
>  
>
> Thx
>
> herve
>
>  
>
>  
>
>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user


Hi Herve,
please check your mediaproxy process, is it running properly (ps -ef |
grep mediaproxy) and listening on your public IP ?
If the client is behind nat, please make sure it's not using any kind on
STUN server to discover its own public IP, or ICE or whatever, NAT
stuffs are handled by NGCP.

Try to run ngrep-sip and check the SIP messages. SDP part of INVITE to
provider and of 200 OK to the client should contain the mediaproxy's
public IP.


Daniel
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130314/be920fb1/attachment.html>

From gavin.sweet at skyracktelecom.com  Thu Mar 14 08:51:15 2013
From: gavin.sweet at skyracktelecom.com (Gavin Sweet)
Date: Thu, 14 Mar 2013 12:51:15 -0000
Subject: [Spce-user] Problem (Bug?) in Billing Fee uploads
Message-ID: <05c801ce20b2$9d760cb0$d8622610$@sweet@skyracktelecom.com>

Hi - 
I have had some problems with destination regex support when uploading
Billing Fees using the batch Upload file function in the Admin GUI.
I have tested these problems with very small file uploads to ensure its not
a problem with file size/transfer etc.
I actually started off trying to ensure an exact match for the number
44118555, which should be simply ^44118555$ - however this doesn't upload.
I then investigated further and found some unexpected results as below. If
anyone can help, or let me know that this is a bug, that would be great.
Manual definition of these fees is not an option for workaround as we have
over 30000 records. An odd (inelegant) form of regex may force it to work,
but that's not what I'd prefer.

Other than case 7 and 8 which I tried for completeness, I think all the
below should work, but they don't.
----------------------------------------------------------------------------
---------------------
Case       Uploaded fee (csv record)                                  =>
Destination (in profile)
----------------------------------------------------------------------------
---------------------
1 (ok)     ^*.$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0                         =>
^*.$
2 (ok)     ^44118555.$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0                  =>
^44118555.$
3 (ok)     ^44118[1-9][0-9][0-9]$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0       =>
^44118[1-9][0-9][0-9]$
4 (ok)     ^4411855[5]$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0                 =>
^4411855[5]$
5 (fail)   ^44118555.*$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0                 =>
44118555
6 (fail)   ^44118555$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0                   =>
44118555
7 (fail)   '^44118555$',ABC,ABC,0,1,0.5,1,0,1,0.5,1,0                 =>
'^44118555$'
8 (fail)   "^44118555$",ABC,ABC,0,1,0.5,1,0,1,0.5,1,0                 =>
"^44118555$"
----------------------------------------------------------------------------
---------------------

Thanks,
Gavin




From gavin.sweet at skyracktelecom.com  Thu Mar 14 10:38:40 2013
From: gavin.sweet at skyracktelecom.com (Gavin Sweet)
Date: Thu, 14 Mar 2013 14:38:40 -0000
Subject: [Spce-user] Problem (Bug?) in Billing Fee uploads
In-Reply-To: <05c801ce20b2$9d760cb0$d8622610$@sweet@skyracktelecom.com>
References: <05c801ce20b2$9d760cb0$d8622610$@sweet@skyracktelecom.com>
Message-ID: <063801ce20c1$9f1da460$dd58ed20$@sweet@skyracktelecom.com>

My table of tests got a bit mangled.
Basically, this problem is when trying to upload fees from file. 
For example, for *exactly* 44118555, the destination should be: ^44118555$
But the ^ and $ get stripped for some reason during the upload from file.
Same with ^44118555.*$
But NOT with ^4411855[5]$ or  ^44118555.$

Cheers
Gavin

> -----Original Message-----
> From: spce-user-bounces at lists.sipwise.com [mailto:spce-user-
> bounces at lists.sipwise.com] On Behalf Of Gavin Sweet
> Sent: 14 March 2013 12:51
> To: spce-user at lists.sipwise.com
> Subject: [Spce-user] Problem (Bug?) in Billing Fee uploads
> 
> Hi -
> I have had some problems with destination regex support when uploading
> Billing Fees using the batch Upload file function in the Admin GUI.
> I have tested these problems with very small file uploads to ensure its
> not a problem with file size/transfer etc.
> I actually started off trying to ensure an exact match for the number
> 44118555, which should be simply ^44118555$ - however this doesn't
> upload.
> I then investigated further and found some unexpected results as below.
> If anyone can help, or let me know that this is a bug, that would be
> great.
> Manual definition of these fees is not an option for workaround as we
> have over 30000 records. An odd (inelegant) form of regex may force it
> to work, but that's not what I'd prefer.
> 
> Other than case 7 and 8 which I tried for completeness, I think all the
> below should work, but they don't.
> -----------------------------------------------------------------------
> -----
> ---------------------
> Case       Uploaded fee (csv record)
> =>
> Destination (in profile)
> -----------------------------------------------------------------------
> -----
> ---------------------
> 1 (ok)     ^*.$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> =>
> ^*.$
> 2 (ok)     ^44118555.$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> =>
> ^44118555.$
> 3 (ok)     ^44118[1-9][0-9][0-9]$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> =>
> ^44118[1-9][0-9][0-9]$
> 4 (ok)     ^4411855[5]$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> =>
> ^4411855[5]$
> 5 (fail)   ^44118555.*$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> =>
> 44118555
> 6 (fail)   ^44118555$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> =>
> 44118555
> 7 (fail)   '^44118555$',ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> =>
> '^44118555$'
> 8 (fail)   "^44118555$",ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> =>
> "^44118555$"
> -----------------------------------------------------------------------
> -----
> ---------------------
> 
> Thanks,
> Gavin
> 
> 
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
> -----
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2013.0.2904 / Virus Database: 2641/6166 - Release Date:
> 03/12/13




From jirka.jirout at mac.com  Thu Mar 14 11:16:01 2013
From: jirka.jirout at mac.com (Jirka Jirout)
Date: Thu, 14 Mar 2013 16:16:01 +0100
Subject: [Spce-user] BYE not correct when sent by called party
Message-ID: <672A65D6-6EDB-463D-8AE5-9EE631A9B076@mac.com>

Hello,

I have a problem with my client app. Both ends are connected via TLS on port 5889 of the LB. When the calling party terminates the call, it sends this and the call terminates correctly:

> BYE sip:127.0.0.1:5080 SIP/2.0'
> ...
> Record-Route: <sip:127.0.0.1:5888;ngcplb=yes;ftag=334359690;lr=on>'
> Record-Route: <sip:SERVER_PUBLIC_IP:5889;transport=tls;ftag=334359690;lr=on>'
> Via: SIP/2.0/UDP 127.0.0.1:5888;branch=z9hG4bKa6f1.654eced0f96edbfaf849c6c2ce831b08.0;i=d951'
> Via: SIP/2.0/TLS 192.168.178.23:54256;received=83.161.151.39;rport;branch=z9hG4bK685900316;rport=54256'
> From: <sip:611092 at OUR_DOMAIN>;tag=334359690'
> To: <sip:279208 at OUR_DOMAIN>;tag=46CD768A-5141E4A8000DAE1F-D32F2700'
> ...
> Route: <sip:127.0.0.1:5888;nat=yes;ngcplb=yes;ftag=334359690;lr=on>'
> Route: <sip:127.0.0.1:5062;lr=on;ftag=334359690;did=f08.4e81;mpd=ii;rtpprx=yes;vsf=dU5ZM1lleEZ6TkxNaUpYTlBkZk11TlkzWWV4Rno->'

When however the called party terminates the call, the packet looks like:

> BYE sip:127.0.0.1:5080 SIP/2.0'
> ...
> Record-Route: <sip:127.0.0.1:5888;ngcplb=yes;ftag=1501122437;lr=on>'
> Record-Route: <sip:SERVER_PUBLIC_IP:5889;transport=tls;ftag=1501122437;lr=on>'
> Via: SIP/2.0/UDP 127.0.0.1:5888;branch=z9hG4bKcc48.1dec6a8cec99aabfe552815890f9cfa7.0;i=8351'
> Via: SIP/2.0/TLS 192.168.178.46:60714;received=83.161.151.39;rport;branch=z9hG4bK1819339446;rport=60714'
> From: <sip:279208 at OUR_DOMAIN>;tag=1501122437'
> To: <sip:611092 at OUR_DOMAIN>;tag=0A95611E-5141E52C000C3E89-D81C3700'
> ...
> Route: <sip:127.0.0.1:5888;ftag=0A95611E-5141E52C000C3E89-D81C3700;lr=on>'

And the called party receives 403 Domain not served here

The kamailio log shows:

Mar 14 15:56:57 server /usr/sbin/kamailio[399]: WARNING: <core> [forward.c:268]: WARNING: get_send_socket: protocol/port mismatch (forced udp:127.0.0.1:5888, to tls:83.161.151.39:60714)

Now, I was under the impression, that the answers are always sent by using Via headers. Here it does not appear to be the case and . We will of course try to solve this in our client app's code, but I was wondering whether this could be somehow fixed on the server side only.

regards, jj


From nh.developer at gmail.com  Thu Mar 14 11:33:09 2013
From: nh.developer at gmail.com (Developing Tomorrow)
Date: Thu, 14 Mar 2013 16:33:09 +0100
Subject: [Spce-user] How to route the packet thru a VPN
Message-ID: <001401ce20c9$3be08bd0$b3a1a370$@gmail.com>

Hi

 

Sipwise routes the packets perfectly to standard operators

And to gateways that are connected directly thru a PPP connection.

 

How can we achieve the same when the VPN tunnel is created thru a tun+
device (ex: openvpn)

The tunnel works perfectly.

I can ping my gateway on the other side.

My routes are well, and can exchange files back and forth.

 

But sipwise can simply not find the gateway when put in sip peering..

When I dump the messages of the tun+ device, I absolutely see nothing
regarding sip messages, so it seems that it's not even trying.

 

Any idea would be appreciated.

 

thanks

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130314/a79c98d7/attachment.html>

From nh.developer at gmail.com  Thu Mar 14 11:35:25 2013
From: nh.developer at gmail.com (Developing Tomorrow)
Date: Thu, 14 Mar 2013 16:35:25 +0100
Subject: [Spce-user] Spce-user Digest, Vol 28, Issue 25
In-Reply-To: <mailman.1164.1363271921.2547.spce-user@lists.sipwise.com>
References: <mailman.1164.1363271921.2547.spce-user@lists.sipwise.com>
Message-ID: <001901ce20c9$8d357810$a7a06830$@gmail.com>

Well, issue is "resolved"

Apparently on a fresh install, if you do not define the prefix properly, 
Sipwise will not let you call an unformatted number.
I have tried to dial my mobile number directly , but it did not work.
But putting 00 is would recognize the format and go thru...

So lesson is that all small details count.

Thank you for the assistance



_________________________________________________________________________
Please consider the environment before printing this email.
?
Notice: Please do not include customer identifying information (i.e.,
account number, social security, etc.) in email communications.
?
Privacy Statement: This message may contain confidential information that is
protected by law. If you received this message in error, please notify the
sender by fax or email and delete this message. If you properly received
this message, you may use its contents only in strictaccordance with our
instructions and privacy policy.
If any person makes a false or misleading representation to obtain customer
information, that person may have committed a federal crime, and we may
report that person to the proper authorities.

-----Original Message-----
From: spce-user-bounces at lists.sipwise.com
[mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of
spce-user-request at lists.sipwise.com
Sent: 14 March 2013 15:39
To: spce-user at lists.sipwise.com
Subject: Spce-user Digest, Vol 28, Issue 25

Send Spce-user mailing list submissions to
	spce-user at lists.sipwise.com

To subscribe or unsubscribe via the World Wide Web, visit
	http://lists.sipwise.com/listinfo/spce-user
or, via email, send a message with subject or body 'help' to
	spce-user-request at lists.sipwise.com

You can reach the person managing the list at
	spce-user-owner at lists.sipwise.com

When replying, please edit your Subject line so it is more specific than
"Re: Contents of Spce-user digest..."


Today's Topics:

   1. media packets not travelling properly (Developing Tomorrow)
   2. media packets not travelling properly (Developing Tomorrow)
   3. Re: media packets not travelling properly (Daniel Grotti)
   4. Problem (Bug?) in Billing Fee uploads (Gavin Sweet)
   5. Re: Problem (Bug?) in Billing Fee uploads (Gavin Sweet)


----------------------------------------------------------------------

Message: 1
Date: Thu, 14 Mar 2013 12:34:17 +0100
From: "Developing Tomorrow" <nh.developer at gmail.com>
Subject: [Spce-user] media packets not travelling properly
To: <spce-user at lists.sipwise.com>
Message-ID: <00ba01ce20a7$dd924120$98b6c360$@gmail.com>
Content-Type: text/plain; charset="us-ascii"

Hi.,

 

This is my first mail to this mailing list, so please forgive my erroneous
procedure if it might be.

 

I have installed sipwise CE, with some pains but it does the job.

If I use a provider connected over normal line (any normal sip provider)

The signalling is done properly I can see the messages, but then the audio
is not following.

 

By that I mean I cannot hear the ringtone nor can I hear the other party
talking, nor they can hear me.

 

I have tried over a vpn, and I have one way audio.

 

The system is installed on a public server with the default iptables.

I have created only 1 domain that receive the external connections on port
5060

 

On the other hand, I have a vpn conection to a foreign network.

Client-----XXXXX (public ip) ----------provider********************* no
audio nor A nor B leg.

Client-----XXXXX(public ip)----------VPN_start
(10.8.0.1)-------------VPN_end(10.8.0.6)-----------provider (172.26.10.2
[not real])******** 1 way audio, they do not hear me

 

Anybody can shade a light here?

 

Thx

herve

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130314/ce2315ab/
attachment-0001.htm>

------------------------------

Message: 2
Date: Thu, 14 Mar 2013 12:26:41 +0100
From: "Developing Tomorrow" <nh.developer at gmail.com>
Subject: [Spce-user] media packets not travelling properly
To: <spce-user at lists.sipwise.com>
Message-ID: <00b501ce20a6$cdbdefc0$6939cf40$@gmail.com>
Content-Type: text/plain; charset="us-ascii"

Hi.,

 

This is my first mail to this mailing list, so please forgive my erroneous
procedure if it might be.

 

I have installed sipwise CE, with some pains but it does the job.

If I use a provider connected over normal line (any normal sip provider)

The signalling is done properly I can see the messages, but then the audio
is not following.

 

By that I mean I cannot hear the ringtone nor can I hear the other party
talking, nor they can hear me.

 

I have tried over a vpn, and I have one way audio.

 

The system is installed on a public server with the default iptables.

I have created only 1 domain that receive the external connections on port
5060

 

On the other hand, I have a vpn conection to a foreign network.

Client-----XXXXX (public ip) ----------provider********************* no
audio nor A nor B leg.

Client-----XXXXX(public ip)----------VPN_start
(10.8.0.1)-------------VPN_end(10.8.0.6)-----------provider (172.26.10.2
[not real])******** 1 way audio, they do not hear me

 

Anybody can shade a light here?

 

Thx

herve

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130314/07c29be2/
attachment-0001.htm>

------------------------------

Message: 3
Date: Thu, 14 Mar 2013 12:49:01 +0100
From: Daniel Grotti <dgrotti at sipwise.com>
Subject: Re: [Spce-user] media packets not travelling properly
To: spce-user at lists.sipwise.com
Message-ID: <5141B92D.7010105 at sipwise.com>
Content-Type: text/plain; charset="iso-8859-1"

On 03/14/2013 12:34 PM, Developing Tomorrow wrote:
>
> Hi.,
>
>  
>
> This is my first mail to this mailing list, so please forgive my 
> erroneous procedure if it might be.
>
>  
>
> I have installed sipwise CE, with some pains but it does the job.
>
> If I use a provider connected over normal line (any normal sip 
> provider)
>
> The signalling is done properly I can see the messages, but then the 
> audio is not following.
>
>  
>
> By that I mean I cannot hear the ringtone nor can I hear the other 
> party talking, nor they can hear me.
>
>  
>
> I have tried over a vpn, and I have one way audio.
>
>  
>
> The system is installed on a public server with the default iptables.
>
> I have created only 1 domain that receive the external connections on 
> port 5060
>
>  
>
> On the other hand, I have a vpn conection to a foreign network.
>
> Client-----XXXXX (public ip) ----------provider*********************
> no audio nor A nor B leg.
>
> Client-----XXXXX(public ip)----------VPN_start 
> (10.8.0.1)-------------VPN_end(10.8.0.6)-----------provider
> (172.26.10.2 [not real])******** 1 way audio, they do not hear me
>
>  
>
> Anybody can shade a light here?
>
>  
>
> Thx
>
> herve
>
>  
>
>  
>
>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user


Hi Herve,
please check your mediaproxy process, is it running properly (ps -ef | grep
mediaproxy) and listening on your public IP ?
If the client is behind nat, please make sure it's not using any kind on
STUN server to discover its own public IP, or ICE or whatever, NAT stuffs
are handled by NGCP.

Try to run ngrep-sip and check the SIP messages. SDP part of INVITE to
provider and of 200 OK to the client should contain the mediaproxy's public
IP.


Daniel
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130314/be920fb1/
attachment-0001.htm>

------------------------------

Message: 4
Date: Thu, 14 Mar 2013 12:51:15 -0000
From: "Gavin Sweet" <gavin.sweet at skyracktelecom.com>
Subject: [Spce-user] Problem (Bug?) in Billing Fee uploads
To: <spce-user at lists.sipwise.com>
Message-ID: <05c801ce20b2$9d760cb0$d8622610$@sweet at skyracktelecom.com>
Content-Type: text/plain;	charset="us-ascii"

Hi -
I have had some problems with destination regex support when uploading
Billing Fees using the batch Upload file function in the Admin GUI.
I have tested these problems with very small file uploads to ensure its not
a problem with file size/transfer etc.
I actually started off trying to ensure an exact match for the number
44118555, which should be simply ^44118555$ - however this doesn't upload.
I then investigated further and found some unexpected results as below. If
anyone can help, or let me know that this is a bug, that would be great.
Manual definition of these fees is not an option for workaround as we have
over 30000 records. An odd (inelegant) form of regex may force it to work,
but that's not what I'd prefer.

Other than case 7 and 8 which I tried for completeness, I think all the
below should work, but they don't.
----------------------------------------------------------------------------
---------------------
Case       Uploaded fee (csv record)                                  =>
Destination (in profile)
----------------------------------------------------------------------------
---------------------
1 (ok)     ^*.$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0                         =>
^*.$
2 (ok)     ^44118555.$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0                  =>
^44118555.$
3 (ok)     ^44118[1-9][0-9][0-9]$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0       =>
^44118[1-9][0-9][0-9]$
4 (ok)     ^4411855[5]$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0                 =>
^4411855[5]$
5 (fail)   ^44118555.*$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0                 =>
44118555
6 (fail)   ^44118555$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0                   =>
44118555
7 (fail)   '^44118555$',ABC,ABC,0,1,0.5,1,0,1,0.5,1,0                 =>
'^44118555$'
8 (fail)   "^44118555$",ABC,ABC,0,1,0.5,1,0,1,0.5,1,0                 =>
"^44118555$"
----------------------------------------------------------------------------
---------------------

Thanks,
Gavin




------------------------------

Message: 5
Date: Thu, 14 Mar 2013 14:38:40 -0000
From: "Gavin Sweet" <gavin.sweet at skyracktelecom.com>
Subject: Re: [Spce-user] Problem (Bug?) in Billing Fee uploads
To: <spce-user at lists.sipwise.com>
Message-ID: <063801ce20c1$9f1da460$dd58ed20$@sweet at skyracktelecom.com>
Content-Type: text/plain;	charset="us-ascii"

My table of tests got a bit mangled.
Basically, this problem is when trying to upload fees from file. 
For example, for *exactly* 44118555, the destination should be: ^44118555$
But the ^ and $ get stripped for some reason during the upload from file.
Same with ^44118555.*$
But NOT with ^4411855[5]$ or  ^44118555.$

Cheers
Gavin

> -----Original Message-----
> From: spce-user-bounces at lists.sipwise.com [mailto:spce-user- 
> bounces at lists.sipwise.com] On Behalf Of Gavin Sweet
> Sent: 14 March 2013 12:51
> To: spce-user at lists.sipwise.com
> Subject: [Spce-user] Problem (Bug?) in Billing Fee uploads
> 
> Hi -
> I have had some problems with destination regex support when uploading 
> Billing Fees using the batch Upload file function in the Admin GUI.
> I have tested these problems with very small file uploads to ensure 
> its not a problem with file size/transfer etc.
> I actually started off trying to ensure an exact match for the number 
> 44118555, which should be simply ^44118555$ - however this doesn't 
> upload.
> I then investigated further and found some unexpected results as below.
> If anyone can help, or let me know that this is a bug, that would be 
> great.
> Manual definition of these fees is not an option for workaround as we 
> have over 30000 records. An odd (inelegant) form of regex may force it 
> to work, but that's not what I'd prefer.
> 
> Other than case 7 and 8 which I tried for completeness, I think all 
> the below should work, but they don't.
> ----------------------------------------------------------------------
> -
> -----
> ---------------------
> Case       Uploaded fee (csv record)
> =>
> Destination (in profile)
> ----------------------------------------------------------------------
> -
> -----
> ---------------------
> 1 (ok)     ^*.$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> =>
> ^*.$
> 2 (ok)     ^44118555.$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> =>
> ^44118555.$
> 3 (ok)     ^44118[1-9][0-9][0-9]$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> =>
> ^44118[1-9][0-9][0-9]$
> 4 (ok)     ^4411855[5]$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> =>
> ^4411855[5]$
> 5 (fail)   ^44118555.*$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> =>
> 44118555
> 6 (fail)   ^44118555$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> =>
> 44118555
> 7 (fail)   '^44118555$',ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> =>
> '^44118555$'
> 8 (fail)   "^44118555$",ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> =>
> "^44118555$"
> ----------------------------------------------------------------------
> -
> -----
> ---------------------
> 
> Thanks,
> Gavin
> 
> 
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
> -----
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2013.0.2904 / Virus Database: 2641/6166 - Release Date:
> 03/12/13




------------------------------

_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com
http://lists.sipwise.com/listinfo/spce-user


End of Spce-user Digest, Vol 28, Issue 25
*****************************************




From agranig at sipwise.com  Thu Mar 14 12:16:50 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Thu, 14 Mar 2013 17:16:50 +0100
Subject: [Spce-user] Delete subscriber does not delete 'trusted' entry
In-Reply-To: <035601ce2027$8e13aab0$aa3b0010$@murray@skyracktelecom.com>
References: <035601ce2027$8e13aab0$aa3b0010$@murray@skyracktelecom.com>
Message-ID: <5141F7F2.9020006@sipwise.com>

Hi John,

On 03/13/2013 09:15 PM, John Murray wrote:
> In v2.7 when I subscriber is deleted the entry in the trusted table is
> not deleted therefore when adding the same pattern to a new subscriber
> the original (deleted) subscriber is matched.

Thanks, we're going to have a look at that!

Andreas



From Stephen at newbreakcommunications.com  Thu Mar 14 12:39:29 2013
From: Stephen at newbreakcommunications.com (Stephen Donovan)
Date: Thu, 14 Mar 2013 16:39:29 +0000
Subject: [Spce-user] random registration dropping
In-Reply-To: <20130314150404.6daf9c05@multivac>
References: <2CDACF784B10A14EAAFAB6140DAE0D132409FC@EXCHANGE1.newbreakmanagement.local>
	<20130313230657.7eb033ac@quenya>
	<AF650DB9-A3FD-44B3-9E52-A38B53103CAB@newbreakcommunications.com>
	<20130314150404.6daf9c05@multivac>
Message-ID: <2CDACF784B10A14EAAFAB6140DAE0D1324262B@EXCHANGE1.newbreakmanagement.local>

We are seeing authentication failed messages in kamailio-proxy.log on registration and outbound calls.  Sometimes as many as 2-3 failed attempts before registration succeeds.  On outbound calls, usually just one failure, try again it goes through.

It is the user getting banned due to registration attempts.

Stephen



-----Original Message-----
From: Jon Bonilla [mailto:jbonilla at sipwise.com] 
Sent: Thursday, March 14, 2013 9:04 AM
To: Stephen Donovan
Subject: Re: [Spce-user] random registration dropping

El Thu, 14 Mar 2013 13:47:10 +0000
Stephen Donovan <Stephen at newbreakcommunications.com> escribi?:

> I'm going to sit back and wait, everything seems to have settled down 
> after update last night. If we see any more dropped calls I will 
> forward relevant logs.
> 
> We were seeing the ip temporarily show up banned sometimes and almost 
> immediately drop from the ban list.
> 
>

Is it set to 50 requests in 5 seconds? Is the ip banned or the user is? You could raise the limit or switch the protection off just for testing.

Please keep the list in cc


From gavin.sweet at skyracktelecom.com  Thu Mar 14 14:03:16 2013
From: gavin.sweet at skyracktelecom.com (Gavin Sweet)
Date: Thu, 14 Mar 2013 18:03:16 -0000
Subject: [Spce-user] Problem (Bug?) in Billing Fee uploads
In-Reply-To: <063801ce20c1$9f1da460$dd58ed20$@sweet@skyracktelecom.com>
References: <05c801ce20b2$9d760cb0$d8622610$@sweet@skyracktelecom.com>
	<063801ce20c1$9f1da460$dd58ed20$@sweet@skyracktelecom.com>
Message-ID: <06e601ce20de$34361070$9ca23150$@sweet@skyracktelecom.com>

I re-read the handbook regarding this point, but the behaviour is not
consistent.
According to the handbook, the web interface removes the regex, but this
isn't the case - only for some formats of regex.
I'd like really to be able to see the full code (including regex) that's
being matched, because that's the only way you can be sure you have a fee
that's correct.
At the moment it's pretty tricky to manage because you're not quite sure
what you're looking at.
Thanks for any light anyone can shed...




> -----Original Message-----
> From: spce-user-bounces at lists.sipwise.com [mailto:spce-user-
> bounces at lists.sipwise.com] On Behalf Of Gavin Sweet
> Sent: 14 March 2013 14:39
> To: spce-user at lists.sipwise.com
> Subject: Re: [Spce-user] Problem (Bug?) in Billing Fee uploads
> 
> My table of tests got a bit mangled.
> Basically, this problem is when trying to upload fees from file.
> For example, for *exactly* 44118555, the destination should be:
> ^44118555$ But the ^ and $ get stripped for some reason during the
> upload from file.
> Same with ^44118555.*$
> But NOT with ^4411855[5]$ or  ^44118555.$
> 
> Cheers
> Gavin
> 
> > -----Original Message-----
> > From: spce-user-bounces at lists.sipwise.com [mailto:spce-user-
> > bounces at lists.sipwise.com] On Behalf Of Gavin Sweet
> > Sent: 14 March 2013 12:51
> > To: spce-user at lists.sipwise.com
> > Subject: [Spce-user] Problem (Bug?) in Billing Fee uploads
> >
> > Hi -
> > I have had some problems with destination regex support when
> uploading
> > Billing Fees using the batch Upload file function in the Admin GUI.
> > I have tested these problems with very small file uploads to ensure
> > its not a problem with file size/transfer etc.
> > I actually started off trying to ensure an exact match for the number
> > 44118555, which should be simply ^44118555$ - however this doesn't
> > upload.
> > I then investigated further and found some unexpected results as
> below.
> > If anyone can help, or let me know that this is a bug, that would be
> > great.
> > Manual definition of these fees is not an option for workaround as we
> > have over 30000 records. An odd (inelegant) form of regex may force
> it
> > to work, but that's not what I'd prefer.
> >
> > Other than case 7 and 8 which I tried for completeness, I think all
> > the below should work, but they don't.
> > ---------------------------------------------------------------------
> -
> > -
> > -----
> > ---------------------
> > Case       Uploaded fee (csv record)
> > =>
> > Destination (in profile)
> > ---------------------------------------------------------------------
> -
> > -
> > -----
> > ---------------------
> > 1 (ok)     ^*.$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> > =>
> > ^*.$
> > 2 (ok)     ^44118555.$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> > =>
> > ^44118555.$
> > 3 (ok)     ^44118[1-9][0-9][0-9]$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> > =>
> > ^44118[1-9][0-9][0-9]$
> > 4 (ok)     ^4411855[5]$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> > =>
> > ^4411855[5]$
> > 5 (fail)   ^44118555.*$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> > =>
> > 44118555
> > 6 (fail)   ^44118555$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> > =>
> > 44118555
> > 7 (fail)   '^44118555$',ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> > =>
> > '^44118555$'
> > 8 (fail)   "^44118555$",ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> > =>
> > "^44118555$"
> > ---------------------------------------------------------------------
> -
> > -
> > -----
> > ---------------------
> >
> > Thanks,
> > Gavin
> >
> >
> > _______________________________________________
> > Spce-user mailing list
> > Spce-user at lists.sipwise.com
> > http://lists.sipwise.com/listinfo/spce-user
> > -----
> > No virus found in this message.
> > Checked by AVG - www.avg.com
> > Version: 2013.0.2904 / Virus Database: 2641/6166 - Release Date:
> > 03/12/13
> 
> 
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
> -----
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2013.0.2904 / Virus Database: 2641/6166 - Release Date:
> 03/12/13




From jbonilla at sipwise.com  Thu Mar 14 14:09:26 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Thu, 14 Mar 2013 19:09:26 +0100
Subject: [Spce-user] random registration dropping
In-Reply-To: <2CDACF784B10A14EAAFAB6140DAE0D1324262B@EXCHANGE1.newbreakmanagement.local>
References: <2CDACF784B10A14EAAFAB6140DAE0D132409FC@EXCHANGE1.newbreakmanagement.local>
	<20130313230657.7eb033ac@quenya>
	<AF650DB9-A3FD-44B3-9E52-A38B53103CAB@newbreakcommunications.com>
	<20130314150404.6daf9c05@multivac>
	<2CDACF784B10A14EAAFAB6140DAE0D1324262B@EXCHANGE1.newbreakmanagement.local>
Message-ID: <20130314190926.3dac1292@quenya>

El Thu, 14 Mar 2013 16:39:29 +0000
Stephen Donovan <Stephen at newbreakcommunications.com> escribi?:

> We are seeing authentication failed messages in kamailio-proxy.log on
> registration and outbound calls.  Sometimes as many as 2-3 failed attempts
> before registration succeeds.  On outbound calls, usually just one failure,
> try again it goes through.
> 
> It is the user getting banned due to registration attempts.
> 

If you're not provisioning the devices and the users might have wrong
credentials in their systems maybe you should disable the ddos protection in
config.yml and let the dos protection enabled.

btw, are you running the latest versions of the templates?

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130314/4122821f/attachment.asc>

From agranig at sipwise.com  Thu Mar 14 15:19:48 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Thu, 14 Mar 2013 20:19:48 +0100
Subject: [Spce-user] Problem (Bug?) in Billing Fee uploads
In-Reply-To: <063801ce20c1$9f1da460$dd58ed20$@sweet@skyracktelecom.com>
References: <05c801ce20b2$9d760cb0$d8622610$@sweet@skyracktelecom.com>
	<063801ce20c1$9f1da460$dd58ed20$@sweet@skyracktelecom.com>
Message-ID: <514222D4.1070308@sipwise.com>

Hi,

On 03/14/2013 03:38 PM, Gavin Sweet wrote:
> My table of tests got a bit mangled.
> Basically, this problem is when trying to upload fees from file.
> For example, for *exactly* 44118555, the destination should be: ^44118555$
> But the ^ and $ get stripped for some reason during the upload from file.
> Same with ^44118555.*$
> But NOT with ^4411855[5]$ or  ^44118555.$

It's actually a bit of a consistency problem for historical reasons, 
mixed with an issue when displaying it, and it will be changed in the 
near future.

Currently it works like this:

If you put in a plain number like "441234", it internally gets extended 
to "^441234.*$". This is what people expected us to do in the past, as 
they only wanted to do prefix-based destination matching. When the value 
is displayed again on the admin panel, it is displayed as "441234" 
again. Although, if you check the billing.billing_fees table in the DB, 
you'll the the full regex there.

The same applies to IP addresses, where "1.2.3.4" gets 
"^.*@1\.2\.3\.4$", but are displayed in the admin panel as "1.2.3.4" 
again, and the same for SIP URIs like "user at example.org", which gets 
"^user at example\.org$".

Everything else will be passed to the DB as is. However, something like 
"^441234$" will be saved as is, but displayed as "441234", which is an 
inconsistency and is confusing, as I admit (I'd consider it as a bug).

Now since regex are already used quite heavily in rewrite rules, we have 
already decided to entirely drop this implicit and confusing "magic" in 
the next release, so people will be required to enter exactly the regex 
they want to match. It will break the logic of people who rely on this 
prefix-based thing, but properly documented (also on the admin panel 
itself) it should be manageable. For everyone else, it will be more obvious.

Andreas




From Stephen at newbreakcommunications.com  Thu Mar 14 15:57:28 2013
From: Stephen at newbreakcommunications.com (Stephen Donovan)
Date: Thu, 14 Mar 2013 19:57:28 +0000
Subject: [Spce-user] random registration dropping
In-Reply-To: <20130314190926.3dac1292@quenya>
References: <2CDACF784B10A14EAAFAB6140DAE0D132409FC@EXCHANGE1.newbreakmanagement.local>
	<20130313230657.7eb033ac@quenya>
	<AF650DB9-A3FD-44B3-9E52-A38B53103CAB@newbreakcommunications.com>
	<20130314150404.6daf9c05@multivac>
	<2CDACF784B10A14EAAFAB6140DAE0D1324262B@EXCHANGE1.newbreakmanagement.local>
	<20130314190926.3dac1292@quenya>
Message-ID: <2CDACF784B10A14EAAFAB6140DAE0D13242C28@EXCHANGE1.newbreakmanagement.local>

If I'm understanding correctly, the templates are for provisioning ?  I'm not currently using any provisioning, actually, we haven't figured out how to make it work.  Each ATA is manually configured and managed by us, we do not let the end user touch anything.  Credentials are correct, we will see one failed attempt and a couple seconds later a successful registration with no action on our part.

Stephen


-----Original Message-----
From: Jon Bonilla [mailto:jbonilla at sipwise.com] 
Sent: Thursday, March 14, 2013 1:09 PM
To: spce-user@
Cc: Stephen Donovan
Subject: Re: [Spce-user] random registration dropping

El Thu, 14 Mar 2013 16:39:29 +0000
Stephen Donovan <Stephen at newbreakcommunications.com> escribi?:

> We are seeing authentication failed messages in kamailio-proxy.log on 
> registration and outbound calls.  Sometimes as many as 2-3 failed 
> attempts before registration succeeds.  On outbound calls, usually 
> just one failure, try again it goes through.
> 
> It is the user getting banned due to registration attempts.
> 

If you're not provisioning the devices and the users might have wrong credentials in their systems maybe you should disable the ddos protection in config.yml and let the dos protection enabled.

btw, are you running the latest versions of the templates?


From lorenzo.mangani at gmail.com  Thu Mar 14 16:20:58 2013
From: lorenzo.mangani at gmail.com (Lorenzo Mangani)
Date: Thu, 14 Mar 2013 21:20:58 +0100
Subject: [Spce-user] random registration dropping
In-Reply-To: <2CDACF784B10A14EAAFAB6140DAE0D13242C28@EXCHANGE1.newbreakmanagement.local>
References: <2CDACF784B10A14EAAFAB6140DAE0D132409FC@EXCHANGE1.newbreakmanagement.local>
	<20130313230657.7eb033ac@quenya>
	<AF650DB9-A3FD-44B3-9E52-A38B53103CAB@newbreakcommunications.com>
	<20130314150404.6daf9c05@multivac>
	<2CDACF784B10A14EAAFAB6140DAE0D1324262B@EXCHANGE1.newbreakmanagement.local>
	<20130314190926.3dac1292@quenya>
	<2CDACF784B10A14EAAFAB6140DAE0D13242C28@EXCHANGE1.newbreakmanagement.local>
Message-ID: <CAKN+pHa9Z6Z3RXdGQ6ezNhCkHRHmcyg1JTVWw77pQYxyygLEsg@mail.gmail.com>

Stephen,

Can you share a signaling session for one such devices/events to see if the
device is sending anything unusual? (including the sequential failures
until the success)


Best,

Lorenzo Mangani

HOMER DEV TEAM
QXIP - Network Engineering


On Thu, Mar 14, 2013 at 8:57 PM, Stephen Donovan <
Stephen at newbreakcommunications.com> wrote:

> If I'm understanding correctly, the templates are for provisioning ?  I'm
> not currently using any provisioning, actually, we haven't figured out how
> to make it work.  Each ATA is manually configured and managed by us, we do
> not let the end user touch anything.  Credentials are correct, we will see
> one failed attempt and a couple seconds later a successful registration
> with no action on our part.
>
> Stephen
>
>
> -----Original Message-----
> From: Jon Bonilla [mailto:jbonilla at sipwise.com]
> Sent: Thursday, March 14, 2013 1:09 PM
> To: spce-user@
> Cc: Stephen Donovan
> Subject: Re: [Spce-user] random registration dropping
>
> El Thu, 14 Mar 2013 16:39:29 +0000
> Stephen Donovan <Stephen at newbreakcommunications.com> escribi?:
>
> > We are seeing authentication failed messages in kamailio-proxy.log on
> > registration and outbound calls.  Sometimes as many as 2-3 failed
> > attempts before registration succeeds.  On outbound calls, usually
> > just one failure, try again it goes through.
> >
> > It is the user getting banned due to registration attempts.
> >
>
> If you're not provisioning the devices and the users might have wrong
> credentials in their systems maybe you should disable the ddos protection
> in config.yml and let the dos protection enabled.
>
> btw, are you running the latest versions of the templates?
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
>



--
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130314/ad5816ea/attachment.html>

From jbonilla at sipwise.com  Thu Mar 14 16:49:41 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Thu, 14 Mar 2013 21:49:41 +0100
Subject: [Spce-user] random registration dropping
In-Reply-To: <2CDACF784B10A14EAAFAB6140DAE0D13242C28@EXCHANGE1.newbreakmanagement.local>
References: <2CDACF784B10A14EAAFAB6140DAE0D132409FC@EXCHANGE1.newbreakmanagement.local>
	<20130313230657.7eb033ac@quenya>
	<AF650DB9-A3FD-44B3-9E52-A38B53103CAB@newbreakcommunications.com>
	<20130314150404.6daf9c05@multivac>
	<2CDACF784B10A14EAAFAB6140DAE0D1324262B@EXCHANGE1.newbreakmanagement.local>
	<20130314190926.3dac1292@quenya>
	<2CDACF784B10A14EAAFAB6140DAE0D13242C28@EXCHANGE1.newbreakmanagement.local>
Message-ID: <20130314214941.236f9d6f@quenya>

El Thu, 14 Mar 2013 19:57:28 +0000
Stephen Donovan <Stephen at newbreakcommunications.com> escribi?:

> If I'm understanding correctly, the templates are for provisioning ?

No. Templates are related to the configuration files. Just to be sure that
you're running latest versions. You can run "apt-get update && apt-get upgrade
&& ngcpcfg apply"

>  I'm not
> currently using any provisioning, actually, we haven't figured out how to
> make it work.  Each ATA is manually configured and managed by us, we do not
> let the end user touch anything.  Credentials are correct, we will see one
> failed attempt and a couple seconds later a successful registration with no
> action on our part.
> 

This is correct as the first attempt will receive a 401 and the second one will
receive the 200ok. That way the user is not banned as you need 3 attempts to be
banned with the default settings.


-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130314/68cf23b0/attachment.asc>

From thilo.bangert at gmail.com  Thu Mar 14 17:32:21 2013
From: thilo.bangert at gmail.com (Thilo Bangert)
Date: Thu, 14 Mar 2013 22:32:21 +0100
Subject: [Spce-user] Private IP space without NAT
Message-ID: <7749345.FK6hup84n9@marsupilami>

Hi,

we have spce running in private rfc1918 IP space - all clients and peers are 
in either on the same subnet or in another, directly routed subnet, which is 
also in private IP space.

spce detects the clients being on private IP space and enables NAT support.

However, as outlined above, there is no NAT in our setup. I havent experienced 
any problem yet, but should i be careful? i already have set 
natping_interval: 0
is there anything else i should/could do?

ideally i would like to disable NAT processing (nathelper et. al)

thanks

kind regards
Thilo



From jbonilla at sipwise.com  Fri Mar 15 05:14:31 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Fri, 15 Mar 2013 10:14:31 +0100
Subject: [Spce-user] Private IP space without NAT
In-Reply-To: <7749345.FK6hup84n9@marsupilami>
References: <7749345.FK6hup84n9@marsupilami>
Message-ID: <20130315101431.2d153fe4@quenya>

El Thu, 14 Mar 2013 22:32:21 +0100
Thilo Bangert <thilo.bangert at gmail.com> escribi?:

> Hi,
> 
> we have spce running in private rfc1918 IP space - all clients and peers are 
> in either on the same subnet or in another, directly routed subnet, which is 
> also in private IP space.
> 
> spce detects the clients being on private IP space and enables NAT support.
> 
> However, as outlined above, there is no NAT in our setup. I havent
> experienced any problem yet, but should i be careful? i already have set 
> natping_interval: 0
> is there anything else i should/could do?
> 
> ideally i would like to disable NAT processing (nathelper et. al)
> 

There's no way that I know to easily disable NAT support in the spce. Usually,
there shouldn't be any problem in your nonat scenario. The only reason I could
imagine for disabling NAT is for performance reasons and it should be a quite
big install if one server doesn't handle all your traffic (you would need a
sip:carrier :P)

regarding performance, the NAT keepalive traffic, which you already disabled
and the rtp traffic, which can be disabled via the "never_use_rtpproxy" in the
admin interface are the options to disable.

If the reason is for disabling ip mangling in SIP, you have an array in
config.yml called  "kamailio.lb.nattest_exception_ips" you could use as guide
to check where you can create a custom template to disable the contact mangling.

Maybe someone of my colleagues at sipwise has a cleaner solution, but you don't
have a problem IMHO.


cheers,

Jon

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130315/f118f532/attachment.asc>

From jbonilla at sipwise.com  Fri Mar 15 05:16:18 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Fri, 15 Mar 2013 10:16:18 +0100
Subject: [Spce-user] Spce-user Digest, Vol 28, Issue 25
In-Reply-To: <001901ce20c9$8d357810$a7a06830$@gmail.com>
References: <mailman.1164.1363271921.2547.spce-user@lists.sipwise.com>
	<001901ce20c9$8d357810$a7a06830$@gmail.com>
Message-ID: <20130315101618.20f05c26@quenya>

El Thu, 14 Mar 2013 16:35:25 +0100
"Developing Tomorrow" <nh.developer at gmail.com> escribi?:

> Well, issue is "resolved"
> 
> Apparently on a fresh install, if you do not define the prefix properly, 
> Sipwise will not let you call an unformatted number.
> I have tried to dial my mobile number directly , but it did not work.
> But putting 00 is would recognize the format and go thru...
> 
> So lesson is that all small details count.
> 
> Thank you for the assistance
> 
>

Hi

Sorry, but with this topic (daily Digest) and without any quotation I can't see
what issue you are talking about. 

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130315/b055e2d5/attachment.asc>

From jbonilla at sipwise.com  Fri Mar 15 05:18:15 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Fri, 15 Mar 2013 10:18:15 +0100
Subject: [Spce-user] How to route the packet thru a VPN
In-Reply-To: <001401ce20c9$3be08bd0$b3a1a370$@gmail.com>
References: <001401ce20c9$3be08bd0$b3a1a370$@gmail.com>
Message-ID: <20130315101815.49382ef7@quenya>

El Thu, 14 Mar 2013 16:33:09 +0100
"Developing Tomorrow" <nh.developer at gmail.com> escribi?:

> Hi
> 
>  
> 
> Sipwise routes the packets perfectly to standard operators
> 
> And to gateways that are connected directly thru a PPP connection.
> 
>  
> 
> How can we achieve the same when the VPN tunnel is created thru a tun+
> device (ex: openvpn)
> 
> The tunnel works perfectly.
> 
> I can ping my gateway on the other side.
> 
> My routes are well, and can exchange files back and forth.
> 
>  
> 
> But sipwise can simply not find the gateway when put in sip peering..
> 
> When I dump the messages of the tun+ device, I absolutely see nothing
> regarding sip messages, so it seems that it's not even trying.
> 
>  
> 
> Any idea would be appreciated.
> 
>  
> 
> thanks
> 


What does a network capture say? And a ngrep-sip capture? There's another
thread for this issue where a colleague asks for a network capture AFAIKS

The thread is called ""[Spce-user] media packets not travelling properly""
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130315/c4fa27f7/attachment.asc>

From jbonilla at sipwise.com  Fri Mar 15 05:21:12 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Fri, 15 Mar 2013 10:21:12 +0100
Subject: [Spce-user] Billing Profiles/Fees
In-Reply-To: <EE7149FDD60B3045A4377F05AAFFF3F118342A46@ISS-Server.ISS.local>
References: <EE7149FDD60B3045A4377F05AAFFF3F118342A46@ISS-Server.ISS.local>
Message-ID: <20130315102112.37d2c298@quenya>

El Wed, 13 Mar 2013 15:52:10 +0000
Christopher Saroli <csaroli at isysp.com> escribi?:

> Little confused on this, every call I make/receive does not get a fee applied
> to it.  Just trying right now to apply a fee to all calls.  I might be
> totally wrong here, any info would be appreciated.
> 
> This is my config:
> 
> Billing Profile:
> Interval Charge:  0.02
> Freetime: 	  0
> Free Cash: 	  0
> Monthly Lim	  0
> Lock		  none
> Currency: 	 $
> 
> 
> Bililng Fees
> Destination:  	.*
> Zone:      	All
> Init Rate:	1
> Init interval:	1
> Follow Rate: 	1
> Follow interval: 1
> 
> 
> Also I have no outgoing calls recorded in the CDR under the admin interface
> but in the customer portal they are there. Is this by design or is something
> wrong.
>

Well, he cdrs should be shown in both interfaces. We can go there later on. But
anyways, to check if your calls are rated fine you should check the exported
cdrs (if any).

Is rate-o-mat running in your system? If not, your calls won't be rated and you
won't get exported cdrs.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130315/a3e945f0/attachment.asc>

From jbonilla at sipwise.com  Fri Mar 15 05:22:07 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Fri, 15 Mar 2013 10:22:07 +0100
Subject: [Spce-user] TLS problem
In-Reply-To: <FB145375-D2D0-4DAC-AF84-090544413297@mac.com>
References: <FB145375-D2D0-4DAC-AF84-090544413297@mac.com>
Message-ID: <20130315102207.7bb93f0a@quenya>

El Mon, 11 Mar 2013 17:27:03 +0100
Jirka Jirout <jirka.jirout at mac.com> escribi?:

> Hello,
> 
> 
> >>  - No answer ever comes back
> >>  - When I run ngrep-sip on the server, I do not see the message anywhere,
> >>  although tcpdump shows it on the external interface
> >> 
> > 
> 
> > are you listening tcp port 5061?
> I am not, we use a different port. But I do try to connect to the port on
> which TLS is enabled.
> 
> > I'll try to take a look when I have some more time but meanwhile I'd suggest
> > you to crosspost this to the kamailio-users mailing list as this issue could
> > match there and you might also get an answer there.
> Thanks for suggestion, I will do that. If I do not manage to fix this soon, I
> will open a support ticket for this with you as well.
> 
> 

I'm following this issue in the kamailio mailing list. Looks like it fits
better out there. Please let us know any updates.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130315/01a1ba6b/attachment.asc>

From jbonilla at sipwise.com  Fri Mar 15 05:26:30 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Fri, 15 Mar 2013 10:26:30 +0100
Subject: [Spce-user] Send peer inbound call straight to voicemail
In-Reply-To: <021301ce1b80$79b93db0$6d2bb910$@murray@skyracktelecom.com>
References: <021301ce1b80$79b93db0$6d2bb910$@murray@skyracktelecom.com>
Message-ID: <20130315102630.32c7a024@quenya>

El Thu, 7 Mar 2013 22:09:44 -0000
"John Murray" <john.murray at skyracktelecom.com> escribi?:

> Hi,
> 
>  
> 
> I want to send a specific call type to voicemail.
> 
>  
> 
> I send a call out to a PBX peer who has no voicemail. If the handset is not
> answered I get a call from the peer with RURI 500 which is the voicemail
> number, From is the original caller and a Divert: header with the subscriber
> ddi.
> 
>  
> 
> I know how to take off the Divert header and get the subscriber ddi but what
> would I put in the RURI to send straight to the subscribers voicemail?
> 
> I've tried prefixing 'vmu' to the ddi but that didn't work.
> 
>  

what was the ngrep-sip (including port 5062), the kamailio-proxy log and the
asterisk console output of that call? 

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130315/14f6a6d5/attachment.asc>

From jirka.jirout at mac.com  Fri Mar 15 06:00:54 2013
From: jirka.jirout at mac.com (Jirka Jirout)
Date: Fri, 15 Mar 2013 11:00:54 +0100
Subject: [Spce-user] BYE not correct when sent by called party
In-Reply-To: <mailman.1167.1363279166.2547.spce-user@lists.sipwise.com>
References: <mailman.1167.1363279166.2547.spce-user@lists.sipwise.com>
Message-ID: <E56D2EA8-47FE-4504-9A1C-90013B633FC4@mac.com>

Hello,

I have tried to "fix" this by using session timers, but apparently without much of success. My understanding is that the B2BUA will send UPDATE packets to both parties and if one of them fails to answer, it will send BYE to the other party and drop the call on the proxy.

I want the timeout to be 5 seconds, so I set the following in the subscriber preferences (using the admini interface):
	sst_expires: 10 
	sst_min_timer: 9
	sst_max_timer: 90
	sst_refresh_method: tried all three :-)

I have restarted sems, but the UPDATE or re-INVITE messages are still not being sent to the client. I looked into sems.log and I can see:

[#7f725e02c720] [readFromConfiguration, SBCCallProfile.cpp:229] INFO: SBC:      SST disabled

Why is the B2BUA ignoring my config? Are the timeouts perhaps too short?

regards, jj


From jbonilla at sipwise.com  Fri Mar 15 06:21:38 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Fri, 15 Mar 2013 11:21:38 +0100
Subject: [Spce-user] BYE not correct when sent by called party
In-Reply-To: <E56D2EA8-47FE-4504-9A1C-90013B633FC4@mac.com>
References: <mailman.1167.1363279166.2547.spce-user@lists.sipwise.com>
	<E56D2EA8-47FE-4504-9A1C-90013B633FC4@mac.com>
Message-ID: <20130315112138.3c7aec3f@quenya>

El Fri, 15 Mar 2013 11:00:54 +0100
Jirka Jirout <jirka.jirout at mac.com> escribi?:

> Hello,
> 
> I have tried to "fix" this by using session timers, but apparently without
> much of success. My understanding is that the B2BUA will send UPDATE packets
> to both parties and if one of them fails to answer, it will send BYE to the
> other party and drop the call on the proxy.
> 
> I want the timeout to be 5 seconds, so I set the following in the subscriber
> preferences (using the admini interface): sst_expires: 10 
> 	sst_min_timer: 9
> 	sst_max_timer: 90
> 	sst_refresh_method: tried all three :-)
> 
> I have restarted sems, but the UPDATE or re-INVITE messages are still not
> being sent to the client. I looked into sems.log and I can see:
> 
> [#7f725e02c720] [readFromConfiguration, SBCCallProfile.cpp:229] INFO:
> SBC:      SST disabled
> 
> Why is the B2BUA ignoring my config? Are the timeouts perhaps too short?
> 


Do you have the sst_enable option enabled? By default it's no.

If so, you'll need to set the sems debug to 3 to check the logs. You can do it
via "sems-stats -c "set_loglevel 3""
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130315/2d895f1e/attachment.asc>

From thilo.bangert at gmail.com  Fri Mar 15 07:36:52 2013
From: thilo.bangert at gmail.com (Thilo Bangert)
Date: Fri, 15 Mar 2013 12:36:52 +0100
Subject: [Spce-user] Private IP space without NAT
In-Reply-To: <20130315101431.2d153fe4@quenya>
References: <7749345.FK6hup84n9@marsupilami> <20130315101431.2d153fe4@quenya>
Message-ID: <3202687.qLYUjX0zSM@marsupilami>

On Friday 15 March 2013 10:14:31 Jon Bonilla wrote:
> El Thu, 14 Mar 2013 22:32:21 +0100
> 
> Thilo Bangert <thilo.bangert at gmail.com> escribi?:
> > Hi,
> > 
> > we have spce running in private rfc1918 IP space - all clients and peers
> > are in either on the same subnet or in another, directly routed subnet,
> > which is also in private IP space.
> > 
> > spce detects the clients being on private IP space and enables NAT
> > support.
> > 
> > However, as outlined above, there is no NAT in our setup. I havent
> > experienced any problem yet, but should i be careful? i already have set
> > natping_interval: 0
> > is there anything else i should/could do?
> > 
> > ideally i would like to disable NAT processing (nathelper et. al)
> 
> There's no way that I know to easily disable NAT support in the spce.
> Usually, there shouldn't be any problem in your nonat scenario. The only
> reason I could imagine for disabling NAT is for performance reasons and it
> should be a quite big install if one server doesn't handle all your traffic
> (you would need a sip:carrier :P)

ok - we're not there (yet :)

> 
> regarding performance, the NAT keepalive traffic, which you already disabled
> and the rtp traffic, which can be disabled via the "never_use_rtpproxy" in
> the admin interface are the options to disable.

ah - yes. that one i also already have disabled.

> 
> If the reason is for disabling ip mangling in SIP, you have an array in
> config.yml called  "kamailio.lb.nattest_exception_ips" you could use as
> guide to check where you can create a custom template to disable the
> contact mangling.

i dont have that option in my 2.7 config - 2.8 feature? however, there is no 
mangling going on as far as i can tell, and currently i dont mind spending a 
few cpu cycles on doing nothing...

> 
> Maybe someone of my colleagues at sipwise has a cleaner solution, but you
> don't have a problem IMHO.

thanks for confirming this.

kind regards
Thilo

> 
> 
> cheers,
> 
> Jon



From jirka.jirout at mac.com  Fri Mar 15 08:56:23 2013
From: jirka.jirout at mac.com (Jirka Jirout)
Date: Fri, 15 Mar 2013 13:56:23 +0100
Subject: [Spce-user] BYE not correct when sent by called party
References: <1D5937E1-61DE-487E-B8DB-A3281360619A@me.com>
Message-ID: <CBAC996A-4FD4-4F79-B43F-0765063DAA01@mac.com>

Hello,

> Do you have the sst_enable option enabled? By default it's no.

I did enable it for the domain using administration interface. Does it need to be enabled globally somehow?

I have also tried to enable RTP timeout by setting dead_rtp_time, but that does not terminate the connection either.

Just for the record: I have restarted SEMS and both lb and proxy kamailios.

regards, jj




From nh.developer at gmail.com  Fri Mar 15 09:22:54 2013
From: nh.developer at gmail.com (Developing Tomorrow)
Date: Fri, 15 Mar 2013 14:22:54 +0100
Subject: [Spce-user]  media packets not travelling properly
Message-ID: <004301ce2180$34912120$9db36360$@gmail.com>

Hi.,

 

Spce is a powerful platform. My original problems seems to be with the
router used on the other side of the VPN. A TPLINK 1043RN using DD-WRT
firmware.

After a few reboots, it seems that the modem finally let the info pass thru.

I have added some new machines on the same VPN tunnel but with a different
router (same make and firware) but am having the strange effect. Not
working.

 

Am still analysing the traffic thru the vpn and will provide some captures
if somebody can help.

The spce server is running three different VPN tunnels (10.8.0.1, 10.9.0.1,
10.10.0.1.)

10.8.0.1 is working perfectly..

 

 

 

        [|sip]

13:15:48.621427 IP (tos 0x0, ttl 63, id 0, offset 0, flags [DF], proto UDP
(17), length 31)

    10.8.0.12.sip > 10.8.0.1.sip: SIP, length: 3

 

        [|sip]

13:15:48.621476 IP (tos 0xc0, ttl 64, id 54459, offset 0, flags [none],
proto ICMP (1), length 59)

    10.8.0.1 > 10.8.0.12: ICMP 10.8.0.1 udp port sip unreachable, length 39

        IP (tos 0x0, ttl 63, id 0, offset 0, flags [DF], proto UDP (17),
length 31)

    10.8.0.12.sip > 10.8.0.1.sip: SIP, length: 3

 

        [|sip]

13:15:58.603201 IP (tos 0x0, ttl 63, id 0, offset 0, flags [DF], proto UDP
(17), length 31)

    10.8.0.12.sip > 10.8.0.1.sip: SIP, length: 3

 

        [|sip]

13:15:58.603235 IP (tos 0xc0, ttl 64, id 54460, offset 0, flags [none],
proto ICMP (1), length 59)

    10.8.0.1 > 10.8.0.12: ICMP 10.8.0.1 udp port sip unreachable, length 39

        IP (tos 0x0, ttl 63, id 0, offset 0, flags [DF], proto UDP (17),
length 31)

    10.8.0.12.sip > 10.8.0.1.sip: SIP, length: 3

 

        [|sip]

13:16:08.606366 IP (tos 0x0, ttl 63, id 0, offset 0, flags [DF], proto UDP
(17), length 31)

    10.8.0.12.sip > 10.8.0.1.sip: SIP, length: 3

 

        [|sip]

13:16:08.606420 IP (tos 0xc0, ttl 64, id 54461, offset 0, flags [none],
proto ICMP (1), length 59)

    10.8.0.1 > 10.8.0.12: ICMP 10.8.0.1 udp port sip unreachable, length 39

        IP (tos 0x0, ttl 63, id 0, offset 0, flags [DF], proto UDP (17),
length 31)

    10.8.0.12.sip > 10.8.0.1.sip: SIP, length: 3

 

This is the view of the tunnel

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130315/da98090b/attachment.html>

From gavin.sweet at skyracktelecom.com  Fri Mar 15 10:20:43 2013
From: gavin.sweet at skyracktelecom.com (Gavin Sweet)
Date: Fri, 15 Mar 2013 14:20:43 -0000
Subject: [Spce-user] Problem (Bug?) in Billing Fee uploads
In-Reply-To: <514222D4.1070308@sipwise.com>
References: <05c801ce20b2$9d760cb0$d8622610$@sweet@skyracktelecom.com>	<063801ce20c1$9f1da460$dd58ed20$@sweet@skyracktelecom.com>
	<514222D4.1070308@sipwise.com>
Message-ID: <07d901ce2188$475450e0$d5fcf2a0$@sweet@skyracktelecom.com>

Thanks Andreas, very clear.

In order to preserve regex consistency, we propose until the new release to
use this format for a full number (e.g.):
^4411855[5]$   (=UK dir enquiries)
which is functionally the same as ^4411855$ for but allows us to have the
same in UI and DB.

Or for a prefix (e.g.):
^3360[5]    (=France mobile)
Which, again, should match ^33605.*$

If you think this will fail or break something else, please shout.

Kind regards,
Gavin


> -----Original Message-----
> From: spce-user-bounces at lists.sipwise.com [mailto:spce-user-
> bounces at lists.sipwise.com] On Behalf Of Andreas Granig
> Sent: 14 March 2013 19:20
> To: spce-user at lists.sipwise.com
> Subject: Re: [Spce-user] Problem (Bug?) in Billing Fee uploads
> 
> Hi,
> 
> On 03/14/2013 03:38 PM, Gavin Sweet wrote:
> > My table of tests got a bit mangled.
> > Basically, this problem is when trying to upload fees from file.
> > For example, for *exactly* 44118555, the destination should be:
> > ^44118555$ But the ^ and $ get stripped for some reason during the
> upload from file.
> > Same with ^44118555.*$
> > But NOT with ^4411855[5]$ or  ^44118555.$
> 
> It's actually a bit of a consistency problem for historical reasons,
> mixed with an issue when displaying it, and it will be changed in the
> near future.
> 
> Currently it works like this:
> 
> If you put in a plain number like "441234", it internally gets extended
> to "^441234.*$". This is what people expected us to do in the past, as
> they only wanted to do prefix-based destination matching. When the
> value is displayed again on the admin panel, it is displayed as
> "441234"
> again. Although, if you check the billing.billing_fees table in the DB,
> you'll the the full regex there.
> 
> The same applies to IP addresses, where "1.2.3.4" gets
> "^.*@1\.2\.3\.4$", but are displayed in the admin panel as "1.2.3.4"
> again, and the same for SIP URIs like "user at example.org", which gets
> "^user at example\.org$".
> 
> Everything else will be passed to the DB as is. However, something like
> "^441234$" will be saved as is, but displayed as "441234", which is an
> inconsistency and is confusing, as I admit (I'd consider it as a bug).
> 
> Now since regex are already used quite heavily in rewrite rules, we
> have already decided to entirely drop this implicit and confusing
> "magic" in the next release, so people will be required to enter
> exactly the regex they want to match. It will break the logic of people
> who rely on this prefix-based thing, but properly documented (also on
> the admin panel
> itself) it should be manageable. For everyone else, it will be more
> obvious.
> 
> Andreas
> 
> 
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
> -----
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2013.0.2904 / Virus Database: 2641/6166 - Release Date:
> 03/12/13




From jbonilla at sipwise.com  Fri Mar 15 11:03:23 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Fri, 15 Mar 2013 16:03:23 +0100
Subject: [Spce-user] BYE not correct when sent by called party
In-Reply-To: <CBAC996A-4FD4-4F79-B43F-0765063DAA01@mac.com>
References: <1D5937E1-61DE-487E-B8DB-A3281360619A@me.com>
	<CBAC996A-4FD4-4F79-B43F-0765063DAA01@mac.com>
Message-ID: <20130315160323.7eca65ec@multivac>

El Fri, 15 Mar 2013 13:56:23 +0100
Jirka Jirout <jirka.jirout at mac.com> escribi?:

> Hello,
> 
> > Do you have the sst_enable option enabled? By default it's no.
> 
> I did enable it for the domain using administration interface. Does it need
> to be enabled globally somehow?
> 

No. If you have enabled for the domain it should be enabled for all
subscribers. But as you were not listing the option...

	sst_expires: 10 
	sst_min_timer: 9
	sst_max_timer: 90

	*** sst_enable: yes  ***

> I have also tried to enable RTP timeout by setting dead_rtp_time, but that
> does not terminate the connection either.

That's for the rtp going to sems. In a normal call scenario the rtp doesn't
traverse sems. You have the same feature in our mediaproxy-ng.

> 
> Just for the record: I have restarted SEMS and both lb and proxy kamailios.
> 
> 


If you have it enabled, please send us the invite sent by the proxy to sems.
sems is controlled by custom headers sent by the proxy (be careful as they
contain sensitive info). 

If the headers are right, enable sems debug and let's see what it says.




From Stephen at newbreakcommunications.com  Fri Mar 15 11:21:28 2013
From: Stephen at newbreakcommunications.com (Stephen Donovan)
Date: Fri, 15 Mar 2013 15:21:28 +0000
Subject: [Spce-user] random registration dropping
In-Reply-To: <20130314214941.236f9d6f@quenya>
References: <2CDACF784B10A14EAAFAB6140DAE0D132409FC@EXCHANGE1.newbreakmanagement.local>
	<20130313230657.7eb033ac@quenya>
	<AF650DB9-A3FD-44B3-9E52-A38B53103CAB@newbreakcommunications.com>
	<20130314150404.6daf9c05@multivac>
	<2CDACF784B10A14EAAFAB6140DAE0D1324262B@EXCHANGE1.newbreakmanagement.local>
	<20130314190926.3dac1292@quenya>
	<2CDACF784B10A14EAAFAB6140DAE0D13242C28@EXCHANGE1.newbreakmanagement.local>,
	<20130314214941.236f9d6f@quenya>
Message-ID: <2CDACF784B10A14EAAFAB6140DAE0D13243D59@EXCHANGE1.newbreakmanagement.local>

I have done the update, 2 days ago.

I'm finding this issue only with Linksys SPA8000 ATAs.  I swapped a client yesterday to a grandstream gxw4024 I had on the shelf, all problems are gone.  Other Linksys models are functioning well.  Jeremie Chism may be able to chime in with his experiences with the Linksys units as well, his customers are on our server as well.  I see the 401 on first attempt, I understand that to be the case, what I'm seeing is multiple attempts at every registration renewal.  I have the limit increased to 10 before banning so they do not get banned now but this still causes a 20 sec to 1 min interruption in service at most registration renewals but not all.

Again, only seeing this on SPA8000 units, nothing else.  I'm deploying all new grandstream units going forward and have had no issues with them.

Stephen



________________________________________
From: Jon Bonilla [jbonilla at sipwise.com]
Sent: Thursday, March 14, 2013 3:49 PM
To: spce-user@
Cc: Stephen Donovan
Subject: Re: [Spce-user] random registration dropping

El Thu, 14 Mar 2013 19:57:28 +0000
Stephen Donovan <Stephen at newbreakcommunications.com> escribi?:

> If I'm understanding correctly, the templates are for provisioning ?

No. Templates are related to the configuration files. Just to be sure that
you're running latest versions. You can run "apt-get update && apt-get upgrade
&& ngcpcfg apply"

>  I'm not
> currently using any provisioning, actually, we haven't figured out how to
> make it work.  Each ATA is manually configured and managed by us, we do not
> let the end user touch anything.  Credentials are correct, we will see one
> failed attempt and a couple seconds later a successful registration with no
> action on our part.
>

This is correct as the first attempt will receive a 401 and the second one will
receive the 200ok. That way the user is not banned as you need 3 attempts to be
banned with the default settings.





From jbonilla at sipwise.com  Fri Mar 15 11:57:06 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Fri, 15 Mar 2013 16:57:06 +0100
Subject: [Spce-user] random registration dropping
In-Reply-To: <2CDACF784B10A14EAAFAB6140DAE0D13243D59@EXCHANGE1.newbreakmanagement.local>
References: <2CDACF784B10A14EAAFAB6140DAE0D132409FC@EXCHANGE1.newbreakmanagement.local>
	<20130313230657.7eb033ac@quenya>
	<AF650DB9-A3FD-44B3-9E52-A38B53103CAB@newbreakcommunications.com>
	<20130314150404.6daf9c05@multivac>
	<2CDACF784B10A14EAAFAB6140DAE0D1324262B@EXCHANGE1.newbreakmanagement.local>
	<20130314190926.3dac1292@quenya>
	<2CDACF784B10A14EAAFAB6140DAE0D13242C28@EXCHANGE1.newbreakmanagement.local>
	<20130314214941.236f9d6f@quenya>
	<2CDACF784B10A14EAAFAB6140DAE0D13243D59@EXCHANGE1.newbreakmanagement.local>
Message-ID: <20130315165706.4854673f@multivac>

El Fri, 15 Mar 2013 15:21:28 +0000
Stephen Donovan <Stephen at newbreakcommunications.com> escribi?:

> I have done the update, 2 days ago.
> 
> I'm finding this issue only with Linksys SPA8000 ATAs.  I swapped a client
> yesterday to a grandstream gxw4024 I had on the shelf, all problems are
> gone.  Other Linksys models are functioning well.  Jeremie Chism may be able
> to chime in with his experiences with the Linksys units as well, his
> customers are on our server as well.  I see the 401 on first attempt, I
> understand that to be the case, what I'm seeing is multiple attempts at every
> registration renewal.  I have the limit increased to 10 before banning so
> they do not get banned now but this still causes a 20 sec to 1 min
> interruption in service at most registration renewals but not all.
> 
> Again, only seeing this on SPA8000 units, nothing else.  I'm deploying all
> new grandstream units going forward and have had no issues with them.
>

Thanks Stephen. Please, keep us updated. In case you don't find anything maybe
you could try to register one of your 8000 to one of my test servers so I could
investigate by myself. Let's wait/see how other devices behave.




From jirka.jirout at mac.com  Fri Mar 15 12:30:51 2013
From: jirka.jirout at mac.com (Jirka Jirout)
Date: Fri, 15 Mar 2013 17:30:51 +0100
Subject: [Spce-user] BYE not correct when sent by called party
In-Reply-To: <mailman.1176.1363363037.2547.spce-user@lists.sipwise.com>
References: <mailman.1176.1363363037.2547.spce-user@lists.sipwise.com>
Message-ID: <7BA44D46-0ECA-4012-8B14-406B5295A4E9@mac.com>

Hello,

> If you have it enabled, please send us the invite sent by the proxy to sems.
> sems is controlled by custom headers sent by the proxy (be careful as they
> contain sensitive info). 
> 
> If the headers are right, enable sems debug and let's see what it says.

here is the INVITE. As far as I can tell, the sst headers are there in P-App-Param:

U 2013/03/15 17:19:37.828094 127.0.0.1:5062 -> 127.0.0.1:5080
INVITE sip:611092 at 10.66.80.74:51522;transport=tls SIP/2.0'
Record-Route: <sip:127.0.0.1:5062;lr=on;ftag=827177332;did=243.d691;mpd=ii;rtpprx=yes;vsf=dU5ZM1lleEZ6TkxNaUpYTlBkZk11TlkzWWV4Rno->'
Record-Route: <sip:127.0.0.1:5888;nat=yes;ngcplb=yes;ftag=827177332;lr=on>'
Record-Route: <sip:SERVER_PUBLIC_IP:5888;nat=yes;ftag=827177332;lr=on>'
Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bK1ec4.a83e4ec5cd343f7fecf75bdf773646e4.0'
Route: <sip:lb at 127.0.0.1:5888;lr;received='sip:62.140.132.74:16292;transport=tls'>'
Via: SIP/2.0/UDP 127.0.0.1:5888;branch=z9hG4bK1ec4.76b4567dd1a238b177841633dc8db86d.0;i=d1'
Via: SIP/2.0/TCP 192.168.178.54:50796;received=CLIENT_PUBLIC_IP;branch=z9hG4bK64699119;rport=50796'
From: <sip:800169 at server.ourdomain.cz>;tag=827177332'
To: <sip:611092 at server.ourdomain.cz>'
Contact: <sip:800169 at 192.168.178.54:50796;alias=CLIENT_PUBLIC_IP~50796~2;transport=tcp>;+g.oma.sip-im;language="en,fr";+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"'
Call-ID: 26257500-8dda-7445-4475-daaf33515075'
CSeq: 847134761 INVITE'
Content-Type: application/sdp'
Content-Length: 734'
Max-Forwards: 31'
Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"'
P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel'
Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER'
P-Access-Network-Info: ADSL;utran-cell-id-3gpp=00000000'
User-Agent: CLIENT_APPLICATION
Supported: 100rel'
P-Called-Party-ID: <sip:611092 at server.ourdomain.cz>'
P-App-Param: ;aleg_sst_enable=yes;aleg_sst_expires=10;aleg_sst_min_timer=9;aleg_sst_max_timer=90;aleg_sst_refresh_method=UPDATE_FALLBACK_INVITE;sst_enable=yes;sst_expires=10;sst_min_timer=9;sst_max_timer=90;sst_refresh_method=UPDATE_FALLBACK_INVITE'
P-App-Name: sbc'
P-D-Uri: sip:lb at 127.0.0.1:5888;lr;received='sip:62.140.132.74:16292;transport=tls'
P-Caller-UUID: 0d7be19d-0565-4c80-995e-4aabca6f93b4'
P-Callee-UUID: 1358f8cc-c4dc-4f59-83dd-caa71267bfb7'

I do not see anything in the log. As a matter of fact, the logging seems to have stopped since i set the log level to 3 like you advised me :-)

regards, jj


From adriavidal at gmail.com  Sun Mar 17 04:18:50 2013
From: adriavidal at gmail.com (=?UTF-8?B?QWRyacOgIFZpZGFs?=)
Date: Sun, 17 Mar 2013 09:18:50 +0100
Subject: [Spce-user] login error Sipwise NGCP Administration Interface
Message-ID: <CAFYw3-duvK2yiPn_WaLOGzQoE58fNO65wu04LO8D3hcZd1yEOA@mail.gmail.com>

I've just installed NGCP Ce on a clean debian server, after install when i
try to login at https://myip:1443
administrator/administrator login is not accepted.

Can i reset or modify it from the console, config file or something else?

-- 
--
Adri? Vidal
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130317/d625a655/attachment.html>

From agranig at sipwise.com  Sun Mar 17 04:42:55 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Sun, 17 Mar 2013 09:42:55 +0100
Subject: [Spce-user] login error Sipwise NGCP Administration Interface
In-Reply-To: <CAFYw3-duvK2yiPn_WaLOGzQoE58fNO65wu04LO8D3hcZd1yEOA@mail.gmail.com>
References: <CAFYw3-duvK2yiPn_WaLOGzQoE58fNO65wu04LO8D3hcZd1yEOA@mail.gmail.com>
Message-ID: <5145820F.9010700@sipwise.com>

Hi,

On 03/17/2013 09:18 AM, Adri? Vidal wrote:
> I've just installed NGCP Ce on a clean debian server, after install when
> i try to login at https://myip:1443
> administrator/administrator login is not accepted.
>
> Can i reset or modify it from the console, config file or something else?

Check /var/log/ngcp/oss.log for any error messages or something obvious. 
Probably MySQL is not running.

Andreas



From lorenzo.mangani at gmail.com  Sun Mar 17 04:52:00 2013
From: lorenzo.mangani at gmail.com (Lorenzo Mangani)
Date: Sun, 17 Mar 2013 09:52:00 +0100
Subject: [Spce-user] login error Sipwise NGCP Administration Interface
In-Reply-To: <CAFYw3-duvK2yiPn_WaLOGzQoE58fNO65wu04LO8D3hcZd1yEOA@mail.gmail.com>
References: <CAFYw3-duvK2yiPn_WaLOGzQoE58fNO65wu04LO8D3hcZd1yEOA@mail.gmail.com>
Message-ID: <CAKN+pHZ2p4a3upNKOJrckrM=7yUzsisz=-zKbcy2eaKLe66-GA@mail.gmail.com>

Adria,

The below query can be used to quickly reset the administrator account
password:

mysql> use billing;
mysql> UPDATE admins SET md5pass = MD5('test') WHERE login =
'administrator';


Best,

Lorenzo Mangani

HOMER DEV TEAM
QXIP - Network Engineering


On Sun, Mar 17, 2013 at 9:18 AM, Adri? Vidal <adriavidal at gmail.com> wrote:

> I've just installed NGCP Ce on a clean debian server, after install when i
> try to login at https://myip:1443
> administrator/administrator login is not accepted.
>
> Can i reset or modify it from the console, config file or something else?
>
> --
> --
> Adri? Vidal
>
>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
>
>


--
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130317/e64c3f57/attachment.html>

From rea1ity at kaist.ac.kr  Sun Mar 17 05:18:11 2013
From: rea1ity at kaist.ac.kr (=?UTF-8?B?7ZeI7Jyk?=)
Date: Sun, 17 Mar 2013 18:18:11 +0900 (KST)
Subject: [Spce-user] =?utf-8?q?Could_you_let_me_know_how_to_registering_PS?=
	=?utf-8?q?TN_gateway_that_I_have=3F?=
Message-ID: <51458af03fe9_@_imoxion.com>

Hi all,
As I refer SIPWISE document it describe how to registering at a peer server same like following;
================================
6.4.4.2. Registering at a Peering Server
Unfortunately, the credentials configured above are not yet automatically used to register the SPCE at your peer hosts. There is however an easy manual way to do so, until this is addressed.

Configure your peering servers with the corresponding credentials in /etc/ngcp-config/templates/etc/sems/etc/reg_agent.conf.tt2, then execute ngcpcfg apply.
=================================
So that I opened the reg_agent.conf.tt2 file and found some could modify part it could register a peer server same like following;
=================================
1. Realm server name,
2. Realm username,
3. Password
=================================
And there have more information that I could modify same like proxy ... Etc.
But I don't use proxy server so that I didn't modified it.
Anyway, I don't really knows what information could I input to there.
It seems it have input about the PSTN gateway server information to there so I modified it as I think.
Also I don't exactly knows what is username and password information of PSTN gateway required the SIPWISE.
So I'm put on the username and password information as web management account of PSTN gateway.
But PSTN gateway register status still 'unregistered'.
Am I missing information about it?
Please let me know how to register with my PSTN gateway.
Is the file reg_agent.conf.tt2 needed to modify if I want to register with my PSTN gateway, right?
If not, please advise how it works.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130317/eafe9881/attachment.html>

From rea1ity at kaist.ac.kr  Sun Mar 17 05:20:39 2013
From: rea1ity at kaist.ac.kr (=?UTF-8?B?7ZeI7Jyk?=)
Date: Sun, 17 Mar 2013 18:20:39 +0900 (KST)
Subject: [Spce-user] =?utf-8?q?Could_you_let_me_know_how_to_registering_PS?=
	=?utf-8?q?TN_gateway_that_I_have=3F?=
Message-ID: <51458b903fe8_@_imoxion.com>

Hi all,
As I refer SIPWISE document it describe how to registering at a peer server same like following;
================================
6.4.4.2. Registering at a Peering Server
Unfortunately, the credentials configured above are not yet automatically used to register the SPCE at your peer hosts. There is however an easy manual way to do so, until this is addressed.

Configure your peering servers with the corresponding credentials in /etc/ngcp-config/templates/etc/sems/etc/reg_agent.conf.tt2, then execute ngcpcfg apply.
=================================
So that I opened the reg_agent.conf.tt2 file and found some could modify part it could register a peer server same like following;
=================================
1. Realm server name,
2. Realm username,
3. Password
=================================
And there have more information that I could modify same like proxy ... Etc.
But I don't use proxy server so that I didn't modified it.
Anyway, I don't really knows what information could I input to there.
It seems it have input about the PSTN gateway server information to there so I modified it as I think.
Also I don't exactly knows what is username and password information of PSTN gateway required the SIPWISE.
So I'm put on the username and password information as web management account of PSTN gateway.
But PSTN gateway register status still 'unregistered'.
Am I missing information about it?
Please let me know how to register with my PSTN gateway.
Is the file reg_agent.conf.tt2 needed to modify if I want to register with my PSTN gateway, right?
If not, please advise how it works.

Sorry, missing my information.

Regards,

David Ho.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130317/7d551322/attachment.html>

From adriavidal at gmail.com  Sun Mar 17 05:24:20 2013
From: adriavidal at gmail.com (=?UTF-8?B?QWRyacOgIFZpZGFs?=)
Date: Sun, 17 Mar 2013 10:24:20 +0100
Subject: [Spce-user] login error Sipwise NGCP Administration Interface
In-Reply-To: <CAKN+pHZ2p4a3upNKOJrckrM=7yUzsisz=-zKbcy2eaKLe66-GA@mail.gmail.com>
References: <CAFYw3-duvK2yiPn_WaLOGzQoE58fNO65wu04LO8D3hcZd1yEOA@mail.gmail.com>
	<CAKN+pHZ2p4a3upNKOJrckrM=7yUzsisz=-zKbcy2eaKLe66-GA@mail.gmail.com>
Message-ID: <CAFYw3-eazoF7HQP+Cz=H_NyOj55z5kRDq5fE7i=s3M9NS_webw@mail.gmail.com>

2013/3/17 Lorenzo Mangani <lorenzo.mangani at gmail.com>

> Adria,
>
> The below query can be used to quickly reset the administrator account
> password:
>
> mysql> use billing;
> mysql> UPDATE admins SET md5pass = MD5('test') WHERE login =
> 'administrator';
>
>
Thanks seeems InnoDB is not running on my Mysql server, will check that.

-- 
--
Adri? Vidal
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130317/e166a94d/attachment.html>

From jbonilla at sipwise.com  Sun Mar 17 05:30:53 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Sun, 17 Mar 2013 10:30:53 +0100
Subject: [Spce-user] login error Sipwise NGCP Administration Interface
In-Reply-To: <CAFYw3-eazoF7HQP+Cz=H_NyOj55z5kRDq5fE7i=s3M9NS_webw@mail.gmail.com>
References: <CAFYw3-duvK2yiPn_WaLOGzQoE58fNO65wu04LO8D3hcZd1yEOA@mail.gmail.com>
	<CAKN+pHZ2p4a3upNKOJrckrM=7yUzsisz=-zKbcy2eaKLe66-GA@mail.gmail.com>
	<CAFYw3-eazoF7HQP+Cz=H_NyOj55z5kRDq5fE7i=s3M9NS_webw@mail.gmail.com>
Message-ID: <20130317103053.296ad77b@multivac>

El Sun, 17 Mar 2013 10:24:20 +0100
Adri? Vidal <adriavidal at gmail.com> escribi?:

> 2013/3/17 Lorenzo Mangani <lorenzo.mangani at gmail.com>
> 
> > Adria,
> >
> > The below query can be used to quickly reset the administrator account
> > password:
> >
> > mysql> use billing;
> > mysql> UPDATE admins SET md5pass = MD5('test') WHERE login =
> > 'administrator';
> >
> >
> Thanks seeems InnoDB is not running on my Mysql server, will check that.
> 

How much mem do you have in your system? 





From adriavidal at gmail.com  Sun Mar 17 06:26:54 2013
From: adriavidal at gmail.com (=?UTF-8?B?QWRyacOgIFZpZGFs?=)
Date: Sun, 17 Mar 2013 11:26:54 +0100
Subject: [Spce-user] login error Sipwise NGCP Administration Interface
In-Reply-To: <20130317103053.296ad77b@multivac>
References: <CAFYw3-duvK2yiPn_WaLOGzQoE58fNO65wu04LO8D3hcZd1yEOA@mail.gmail.com>
	<CAKN+pHZ2p4a3upNKOJrckrM=7yUzsisz=-zKbcy2eaKLe66-GA@mail.gmail.com>
	<CAFYw3-eazoF7HQP+Cz=H_NyOj55z5kRDq5fE7i=s3M9NS_webw@mail.gmail.com>
	<20130317103053.296ad77b@multivac>
Message-ID: <CAFYw3-cptNWU8Z3fd5-e3F=U2Bm54v+X_R+fLvQBwqxWn377Pw@mail.gmail.com>

>
>
> How much mem do you have in your system?
>
>
Is a openvz instance with 512 ram and 512 swap, uppet to 2024 ram 512 swap
to get everything running (apache + mysql).

Will do some tests now.

Thanks

-- 
--
Adri? Vidal
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130317/91209ea1/attachment.html>

From marcial at tecnimoviles.com  Sun Mar 17 11:12:38 2013
From: marcial at tecnimoviles.com (Marcial Barrios)
Date: Sun, 17 Mar 2013 15:12:38 +0000
Subject: [Spce-user] rate-o-mat
Message-ID: <CAMnKu2FSKYtHaD=oMi4z++rf0WUWy-qju8v5VQQQLfUiM_dgTg@mail.gmail.com>

No calls are rated.  How can check if rate-o-mat is working properly?
-- 
*Marcial Barrios*
*Administrador de** Sistemas*
M?vil: (+34) 603 053 043
  Tel:  (+34) 902 955 290
 Fax:  (+34) 822 201 442
email: marcial at tecnimoviles.com
  web: www.tecnimoviles.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130317/35962318/attachment.html>

From marcial at tecnimoviles.com  Sun Mar 17 11:41:53 2013
From: marcial at tecnimoviles.com (Marcial Barrios)
Date: Sun, 17 Mar 2013 15:41:53 +0000
Subject: [Spce-user] rate-o-mat
In-Reply-To: <CAKN+pHZC9q47uxx=i9A8YSY+1TAbzwuF89wzJ2jpv5G2Y0uVkQ@mail.gmail.com>
References: <CAMnKu2FSKYtHaD=oMi4z++rf0WUWy-qju8v5VQQQLfUiM_dgTg@mail.gmail.com>
	<CAKN+pHYLZ9eMD3+tMJw31eFm_Asu-6uSgL6esfjaVZbsUNn7CQ@mail.gmail.com>
	<CAMnKu2FgPde96M2UHX-B970bnLVT8-qA-CYDMOLNwpLGa4oa1A@mail.gmail.com>
	<CAKN+pHZC9q47uxx=i9A8YSY+1TAbzwuF89wzJ2jpv5G2Y0uVkQ@mail.gmail.com>
Message-ID: <CAMnKu2Gg=i=gfMtU=M3LFZ68VpC5XRhuyn8VpLO_D+ccXCbVHw@mail.gmail.com>

I Lorenzo I did what you suggested

#  service ngcp-rate-o-mat start

but this is the error message

rollback ineffective with AutoCommit enabled at /usr/sbin/rate
                                                                -o-mat line
91.
rollback ineffective with AutoCommit enabled at /usr/sbin/rate-o-mat line
92.
Error rating CDR batch: No fee info for profile 2 and user 'testuser_21' or
doma
      in 'testuser_21 at 176.31.119.223' found


2013/3/17 Lorenzo Mangani <lorenzo.mangani at gmail.com>

> I forgot to copy the list but as we wait for other more valuable
> suggestions, try simply:
>
> #  service ngcp-rate-o-mat start
>
> if you get a message about service being disabled, edit in your config.yml
> to enable rate-o-mat
>
>
>
> Lorenzo
>
>
> On Sun, Mar 17, 2013 at 4:28 PM, Marcial Barrios <marcial at tecnimoviles.com
> > wrote:
>
>> Hi Lorenzo, I apply this:
>>
>> # service ngcp-rate-o-mat status
>>
>> and this is the response:
>>
>> # rate-o-mat is not running ... failed!
>>
>>
>> 2013/3/17 Lorenzo Mangani <lorenzo.mangani at gmail.com>
>>
>>> Hi Marcial,
>>>
>>> You can first of all check and confirm if the service is running on your
>>> system:
>>>
>>> # service ngcp-rate-o-mat status
>>>
>>>
>>>
>>>  Lorenzo Mangani
>>>
>>> HOMER DEV TEAM
>>> QXIP - Network Engineering
>>>
>>>
>>> On Sun, Mar 17, 2013 at 4:12 PM, Marcial Barrios <
>>> marcial at tecnimoviles.com> wrote:
>>>
>>>>
>>>> No calls are rated.  How can check if rate-o-mat is working properly?
>>>> --
>>>> *Marcial Barrios*
>>>> *Administrador de** Sistemas*
>>>>  M?vil: (+34) 603 053 043
>>>>   Tel:  (+34) 902 955 290
>>>>  Fax:  (+34) 822 201 442
>>>> email: marcial at tecnimoviles.com
>>>>   web: www.tecnimoviles.com
>>>>
>>>> _______________________________________________
>>>> Spce-user mailing list
>>>> Spce-user at lists.sipwise.com
>>>> http://lists.sipwise.com/listinfo/spce-user
>>>>
>>>>
>>>
>>>
>>> --
>>>
>>>
>>>
>>>
>>
>>
>> --
>> *Marcial Barrios*
>> *Administrador de** Sistemas*
>> M?vil: (+34) 603 053 043
>>   Tel:  (+34) 902 955 290
>>  Fax:  (+34) 822 201 442
>> email: marcial at tecnimoviles.com
>>   web: www.tecnimoviles.com
>>
>
>
>
> --
>
>
> Lorenzo Mangani
>
> HOMER DEV TEAM
> QXIP - Network Engineering
>
>


-- 
*Marcial Barrios*
*Administrador de** Sistemas*
M?vil: (+34) 603 053 043
  Tel:  (+34) 902 955 290
 Fax:  (+34) 822 201 442
email: marcial at tecnimoviles.com
  web: www.tecnimoviles.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130317/f1bbdff8/attachment.html>

From dgrotti at sipwise.com  Sun Mar 17 11:34:59 2013
From: dgrotti at sipwise.com (Daniel Grotti)
Date: Sun, 17 Mar 2013 16:34:59 +0100
Subject: [Spce-user] rate-o-mat
Message-ID: <j75utrfckctuw1qj68s8qy18.1363534499413@email.android.com>

Hi,
please check what /var/log/ngcp/rate-o-mat.log is saying.

Daniel



Sent from Samsung Mobile

Marcial Barrios <marcial at tecnimoviles.com> wrote:


No calls are rated. ?How can check if rate-o-mat is working properly?
-- 
Marcial Barrios
Administrador de?Sistemas

M?vil: (+34) 603 053 043
? Tel: ?(+34) 902 955 290
?Fax: ?(+34) 822 201 442
email: marcial at tecnimoviles.com
? web:?www.tecnimoviles.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130317/faa71c0a/attachment.html>

From marcial at tecnimoviles.com  Sun Mar 17 12:07:26 2013
From: marcial at tecnimoviles.com (Marcial Barrios)
Date: Sun, 17 Mar 2013 16:07:26 +0000
Subject: [Spce-user] rate-o-mat
In-Reply-To: <CAMnKu2FTfpMAVeCQyXR+0wxOGSqCY5W1uPRi1MJ3KVr8SfEPUQ@mail.gmail.com>
References: <j75utrfckctuw1qj68s8qy18.1363534499413@email.android.com>
	<CAMnKu2FTfpMAVeCQyXR+0wxOGSqCY5W1uPRi1MJ3KVr8SfEPUQ@mail.gmail.com>
Message-ID: <CAMnKu2F03uNr-rr69jHV1G+YVjcm3O7ZE1JwA0UcnykqK422fw@mail.gmail.com>

2013/3/17 Marcial Barrios <marcial at tecnimoviles.com>

> Hi, Daniel this is what rate-o-mat.log is saying:
>
> Trying to connect to billing db...
> Successfully connected to billing db...
> Trying to connect to accounting db...
> Successfully connected to accounting db...
> Up and running.
> rate cdr #13
> No reseller info for user id 10d81223-c19c-4030-b2dd-429af4dd8376 found
> Error rating CDR batch: No reseller info for user id
> 10d81223-c19c-4030-b2dd-429af4dd8376 found
>
>
>
> 2013/3/17 Daniel Grotti <dgrotti at sipwise.com>
>
>> Hi,
>> please check what /var/log/ngcp/rate-o-mat.log is saying.
>>
>> Daniel
>>
>>
>>
>> Sent from Samsung Mobile
>>
>>
>>
>> Marcial Barrios <marcial at tecnimoviles.com> wrote:
>>
>>
>>
>> No calls are rated.  How can check if rate-o-mat is working properly?
>> --
>> *Marcial Barrios*
>> *Administrador de** Sistemas*
>> M?vil: (+34) 603 053 043
>>   Tel:  (+34) 902 955 290
>>  Fax:  (+34) 822 201 442
>> email: marcial at tecnimoviles.com
>>   web: www.tecnimoviles.com
>>
>
>
>
> --
> *Marcial Barrios*
> *Administrador de** Sistemas*
> M?vil: (+34) 603 053 043
>   Tel:  (+34) 902 955 290
>  Fax:  (+34) 822 201 442
> email: marcial at tecnimoviles.com
>   web: www.tecnimoviles.com
>



-- 
*Marcial Barrios*
*Administrador de** Sistemas*
M?vil: (+34) 603 053 043
  Tel:  (+34) 902 955 290
 Fax:  (+34) 822 201 442
email: marcial at tecnimoviles.com
  web: www.tecnimoviles.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130317/bfbdeeef/attachment.html>

From jbonilla at sipwise.com  Sun Mar 17 13:09:35 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Sun, 17 Mar 2013 18:09:35 +0100
Subject: [Spce-user] login error Sipwise NGCP Administration Interface
In-Reply-To: <CAFYw3-cptNWU8Z3fd5-e3F=U2Bm54v+X_R+fLvQBwqxWn377Pw@mail.gmail.com>
References: <CAFYw3-duvK2yiPn_WaLOGzQoE58fNO65wu04LO8D3hcZd1yEOA@mail.gmail.com>
	<CAKN+pHZ2p4a3upNKOJrckrM=7yUzsisz=-zKbcy2eaKLe66-GA@mail.gmail.com>
	<CAFYw3-eazoF7HQP+Cz=H_NyOj55z5kRDq5fE7i=s3M9NS_webw@mail.gmail.com>
	<20130317103053.296ad77b@multivac>
	<CAFYw3-cptNWU8Z3fd5-e3F=U2Bm54v+X_R+fLvQBwqxWn377Pw@mail.gmail.com>
Message-ID: <20130317180935.060ff1ad@multivac>

El Sun, 17 Mar 2013 11:26:54 +0100
Adri? Vidal <adriavidal at gmail.com> escribi?:

> >
> >
> > How much mem do you have in your system?
> >
> >
> Is a openvz instance with 512 ram and 512 swap, uppet to 2024 ram 512 swap
> to get everything running (apache + mysql).
> 
> Will do some tests now.
> 
> Thanks
> 

512 ram is too low to run a spce with the default configuration. To lower the
requisites try to play with the following in your config.yml:

- Disable tls if you're not using it
- Disable presence if you're not using it
- Set lb and kamailio number of children to 2 instead of 8
- Lower the buffer_pool_size asigned to mysql to 128MB
- Lower the pkg_mem of the lb to 8
- set the shm mem of proxy and lb to 64


apply the changes and see if everything is running fine.

cheers,

Jon




From kmasse at questblue.com  Sun Mar 17 13:22:57 2013
From: kmasse at questblue.com (Kevin Masse)
Date: Sun, 17 Mar 2013 13:22:57 -0400
Subject: [Spce-user] rate-o-mat
References: <j75utrfckctuw1qj68s8qy18.1363534499413@email.android.com><CAMnKu2FTfpMAVeCQyXR+0wxOGSqCY5W1uPRi1MJ3KVr8SfEPUQ@mail.gmail.com>
	<CAMnKu2F03uNr-rr69jHV1G+YVjcm3O7ZE1JwA0UcnykqK422fw@mail.gmail.com>
Message-ID: <CEEE718799582B47A55D7897DCB264FC28024C@bvusaex.bestvoipusa.local>

Are you using XML or SOAP to update subscribers?  If so it is possible that you have deleted the subscriber before the rating process went over the CDRs.  


The only way to get your rating back running again is to update the accounting.cdr table manually by setting the rating_status to "ok" for source_subscriber_id=10d81223-c19c-4030-b2dd-429af4dd8376

 

 

If you are not using the XML or SOAP to update it still could be possible you deleted a subscriber before calls were rated.

 

This may have to be repeated a few times to get the status "ok" in all fields.

 

Kevin

 

 

 

 

From: spce-user-bounces at lists.sipwise.com [mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of Marcial Barrios
Sent: Sunday, March 17, 2013 12:07 PM
To: Daniel Grotti; spce-user at lists.sipwise.com
Subject: Re: [Spce-user] rate-o-mat

 

 

 

2013/3/17 Marcial Barrios <marcial at tecnimoviles.com>

Hi, Daniel this is what rate-o-mat.log is saying:

 

Trying to connect to billing db...

Successfully connected to billing db...

Trying to connect to accounting db...

Successfully connected to accounting db...

Up and running.

rate cdr #13

No reseller info for user id 10d81223-c19c-4030-b2dd-429af4dd8376 found

Error rating CDR batch: No reseller info for user id 10d81223-c19c-4030-b2dd-429af4dd8376 found

 

 

2013/3/17 Daniel Grotti <dgrotti at sipwise.com>

Hi,

please check what /var/log/ngcp/rate-o-mat.log is saying.

 

Daniel




Sent from Samsung Mobile 




Marcial Barrios <marcial at tecnimoviles.com> wrote:



 

No calls are rated.  How can check if rate-o-mat is working properly?

-- 

Marcial Barrios

Administrador de Sistemas

M?vil: (+34) 603 053 043 <tel:%28%2B34%29%20603%20053%20043> 

  Tel:  (+34) 902 955 290 <tel:%28%2B34%29%20902%20955%20290> 

 Fax:  (+34) 822 201 442 <tel:%28%2B34%29%20822%20201%20442> 

email: marcial at tecnimoviles.com

  web: www.tecnimoviles.com <http://www.tecnimoviles.com/> 





 

-- 

Marcial Barrios

Administrador de Sistemas

 <http://tecnimoviles.com/wp-content/uploads/2012/10/cropped-cropped-headertecnimoviles1-e1351371092469.png> 

M?vil: (+34) 603 053 043 <tel:%28%2B34%29%20603%20053%20043> 

  Tel:  (+34) 902 955 290 <tel:%28%2B34%29%20902%20955%20290> 

 Fax:  (+34) 822 201 442 <tel:%28%2B34%29%20822%20201%20442> 

email: marcial at tecnimoviles.com

  web: www.tecnimoviles.com





 

-- 

Marcial Barrios

Administrador de Sistemas

 <http://tecnimoviles.com/wp-content/uploads/2012/10/cropped-cropped-headertecnimoviles1-e1351371092469.png> 

M?vil: (+34) 603 053 043 <tel:%28%2B34%29%20603%20053%20043> 

  Tel:  (+34) 902 955 290 <tel:%28%2B34%29%20902%20955%20290> 

 Fax:  (+34) 822 201 442 <tel:%28%2B34%29%20822%20201%20442> 

email: marcial at tecnimoviles.com

  web: www.tecnimoviles.com

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130317/d553e818/attachment.html>

From marcial at tecnimoviles.com  Sun Mar 17 16:09:20 2013
From: marcial at tecnimoviles.com (Marcial Barrios)
Date: Sun, 17 Mar 2013 20:09:20 +0000
Subject: [Spce-user] rate-o-mat
In-Reply-To: <CEEE718799582B47A55D7897DCB264FC28024C@bvusaex.bestvoipusa.local>
References: <j75utrfckctuw1qj68s8qy18.1363534499413@email.android.com>
	<CAMnKu2FTfpMAVeCQyXR+0wxOGSqCY5W1uPRi1MJ3KVr8SfEPUQ@mail.gmail.com>
	<CAMnKu2F03uNr-rr69jHV1G+YVjcm3O7ZE1JwA0UcnykqK422fw@mail.gmail.com>
	<CEEE718799582B47A55D7897DCB264FC28024C@bvusaex.bestvoipusa.local>
Message-ID: <CAMnKu2Gp40A3KXkoVzUbow29tS1f0e=OW3aV0tfxdenUFMxqVg@mail.gmail.com>

Hi Kevin, I do the following and works

update cdr set rating_status ="ok" where
 source_user_id="10d81223-c19c-4030-b2dd-429af4dd8376"

but when I tried to

#service ngcp-rate-o-mat start

I have this other message:

rollback ineffective with AutoCommit enabled at /usr/sbin/rate-o-mat line
91.
rollback ineffective with AutoCommit enabled at /usr/sbin/rate-o-mat line
92.
Error rating CDR batch: No fee info for profile 2 and user '0034922388318'
or domain '0034922388318 at 91.192.110.234' found





2013/3/17 Kevin Masse <kmasse at questblue.com>

> Are you using XML or SOAP to update subscribers?  If so it is possible
> that you have deleted the subscriber before the rating process went over
> the CDRs.  ****
>
>
> The only way to get your rating back running again is to update the
> accounting.cdr table manually by setting the rating_status to ?ok? for
> source_subscriber_id=10d81223-c19c-4030-b2dd-429af4dd8376****
>
> ** **
>
> ** **
>
> If you are not using the XML or SOAP to update it still could be possible
> you deleted a subscriber before calls were rated.****
>
> ** **
>
> This may have to be repeated a few times to get the status ?ok? in all
> fields.****
>
> ** **
>
> Kevin****
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>
> *From:* spce-user-bounces at lists.sipwise.com [mailto:
> spce-user-bounces at lists.sipwise.com] *On Behalf Of *Marcial Barrios
> *Sent:* Sunday, March 17, 2013 12:07 PM
> *To:* Daniel Grotti; spce-user at lists.sipwise.com
> *Subject:* Re: [Spce-user] rate-o-mat****
>
> ** **
>
> ** **
>
> ** **
>
> 2013/3/17 Marcial Barrios <marcial at tecnimoviles.com>****
>
> Hi, Daniel this is what rate-o-mat.log is saying:****
>
> ** **
>
> Trying to connect to billing db...****
>
> Successfully connected to billing db...****
>
> Trying to connect to accounting db...****
>
> Successfully connected to accounting db...****
>
> Up and running.****
>
> rate cdr #13****
>
> No reseller info for user id 10d81223-c19c-4030-b2dd-429af4dd8376 found***
> *
>
> Error rating CDR batch: No reseller info for user id
> 10d81223-c19c-4030-b2dd-429af4dd8376 found****
>
> ** **
>
> ** **
>
> 2013/3/17 Daniel Grotti <dgrotti at sipwise.com>****
>
> Hi,****
>
> please check what /var/log/ngcp/rate-o-mat.log is saying.****
>
> ** **
>
> Daniel****
>
>
>
>
> Sent from Samsung Mobile ****
>
>
>
>
> Marcial Barrios <marcial at tecnimoviles.com> wrote:
>
> ****
>
> ** **
>
> No calls are rated.  How can check if rate-o-mat is working properly?****
>
> -- ****
>
> *Marcial Barrios*****
>
> *Administrador de Sistemas*****
>
> ****
>
> M?vil: (+34) 603 053 043****
>
>   Tel:  (+34) 902 955 290****
>
>  Fax:  (+34) 822 201 442****
>
> email: marcial at tecnimoviles.com****
>
>   web: www.tecnimoviles.com****
>
>
>
> ****
>
> ** **
>
> -- ****
>
> *Marcial Barrios*****
>
> *Administrador de Sistemas*****
>
> ****
>
> M?vil: (+34) 603 053 043****
>
>   Tel:  (+34) 902 955 290****
>
>  Fax:  (+34) 822 201 442****
>
> email: marcial at tecnimoviles.com****
>
>   web: www.tecnimoviles.com****
>
>
>
> ****
>
> ** **
>
> -- ****
>
> *Marcial Barrios*****
>
> *Administrador de Sistemas*****
>
> ****
>
> M?vil: (+34) 603 053 043****
>
>   Tel:  (+34) 902 955 290****
>
>  Fax:  (+34) 822 201 442****
>
> email: marcial at tecnimoviles.com****
>
>   web: www.tecnimoviles.com****
>



-- 
*Marcial Barrios*
*Administrador de** Sistemas*
M?vil: (+34) 603 053 043
  Tel:  (+34) 902 955 290
 Fax:  (+34) 822 201 442
email: marcial at tecnimoviles.com
  web: www.tecnimoviles.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130317/052fa8a1/attachment.html>

From dgrotti at sipwise.com  Sun Mar 17 16:18:50 2013
From: dgrotti at sipwise.com (Daniel Grotti)
Date: Sun, 17 Mar 2013 21:18:50 +0100
Subject: [Spce-user] rate-o-mat
Message-ID: <1xtygsph2mi02o3q5h6d2hy5.1363551530821@email.android.com>

Hi,
this means that you don't have any billing fees for calls to 0034*.
Please add a proper billing fees and restart rate-o-mat.

Daniel



Sent from Samsung Mobile

Marcial Barrios <marcial at tecnimoviles.com> wrote:

Hi Kevin, I do the following and works?

update cdr set rating_status ="ok" where ?source_user_id="10d81223-c19c-4030-b2dd-429af4dd8376"

but when I tried to?

#service ngcp-rate-o-mat start

I have this other message:

rollback ineffective with AutoCommit enabled at /usr/sbin/rate-o-mat line 91.
rollback ineffective with AutoCommit enabled at /usr/sbin/rate-o-mat line 92.
Error rating CDR batch: No fee info for profile 2 and user '0034922388318' or domain '0034922388318 at 91.192.110.234' found





2013/3/17 Kevin Masse <kmasse at questblue.com>
Are you using XML or SOAP to update subscribers?? If so it is possible that you have deleted the subscriber before the rating process went over the CDRs.?


The only way to get your rating back running again is to update the accounting.cdr table manually by setting the rating_status to ?ok? for source_subscriber_id=10d81223-c19c-4030-b2dd-429af4dd8376

?

?

If you are not using the XML or SOAP to update it still could be possible you deleted a subscriber before calls were rated.

?

This may have to be repeated a few times to get the status ?ok? in all fields.

?

Kevin

?

?

?

?

From: spce-user-bounces at lists.sipwise.com [mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of Marcial Barrios
Sent: Sunday, March 17, 2013 12:07 PM
To: Daniel Grotti; spce-user at lists.sipwise.com
Subject: Re: [Spce-user] rate-o-mat

?

?

?

2013/3/17 Marcial Barrios <marcial at tecnimoviles.com>

Hi, Daniel this is what rate-o-mat.log is saying:

?

Trying to connect to billing db...

Successfully connected to billing db...

Trying to connect to accounting db...

Successfully connected to accounting db...

Up and running.

rate cdr #13

No reseller info for user id 10d81223-c19c-4030-b2dd-429af4dd8376 found

Error rating CDR batch: No reseller info for user id 10d81223-c19c-4030-b2dd-429af4dd8376 found

?

?

2013/3/17 Daniel Grotti <dgrotti at sipwise.com>

Hi,

please check what /var/log/ngcp/rate-o-mat.log is saying.

?

Daniel




Sent from Samsung Mobile




Marcial Barrios <marcial at tecnimoviles.com> wrote:


?

No calls are rated. ?How can check if rate-o-mat is working properly?

--

Marcial Barrios

Administrador de?Sistemas


M?vil: (+34) 603 053 043

? Tel: ?(+34) 902 955 290

?Fax: ?(+34) 822 201 442

email: marcial at tecnimoviles.com

? web:?www.tecnimoviles.com




?

--

Marcial Barrios

Administrador de?Sistemas



M?vil: (+34) 603 053 043

? Tel: ?(+34) 902 955 290

?Fax: ?(+34) 822 201 442

email: marcial at tecnimoviles.com

? web:?www.tecnimoviles.com




?

--

Marcial Barrios

Administrador de?Sistemas



M?vil: (+34) 603 053 043

? Tel: ?(+34) 902 955 290

?Fax: ?(+34) 822 201 442

email: marcial at tecnimoviles.com

? web:?www.tecnimoviles.com




-- 
Marcial Barrios
Administrador de?Sistemas

M?vil: (+34) 603 053 043
? Tel: ?(+34) 902 955 290
?Fax: ?(+34) 822 201 442
email: marcial at tecnimoviles.com
? web:?www.tecnimoviles.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130317/1a056e0b/attachment.html>

From vinceoscar at gmail.com  Sun Mar 17 17:05:38 2013
From: vinceoscar at gmail.com (Vince Oscar)
Date: Sun, 17 Mar 2013 17:05:38 -0400
Subject: [Spce-user] setup pstn gateway in and out
Message-ID: <CABAd6xvQi5hsdtQYgRdbN8EyQ68m=hGMPfxPubABy+r3rkwNkQ@mail.gmail.com>

I would like to setup gateways to pstn also to freeswitch and asterisk.
How would I do this? Where would I find the documentation.

I have tried to setup peering but with failure.  I am surely doing things
wrong.  Also what log files should I be checking to observe
success/failures.

Does a default install include all modules to do this?

Please advise

Thank you
Vince
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130317/54ef25cd/attachment.html>

From marcial at tecnimoviles.com  Sun Mar 17 17:07:10 2013
From: marcial at tecnimoviles.com (Marcial Barrios)
Date: Sun, 17 Mar 2013 21:07:10 +0000
Subject: [Spce-user] rate-o-mat
In-Reply-To: <1xtygsph2mi02o3q5h6d2hy5.1363551530821@email.android.com>
References: <1xtygsph2mi02o3q5h6d2hy5.1363551530821@email.android.com>
Message-ID: <CAMnKu2FAYhoKUvvLiLLEbo+HEYYctM4HoPOpy+1b0VEisd73XA@mail.gmail.com>

Yes you were right I add the billing fees and work but this is like the
story that never end. This is the new error message:

 #rollback ineffective with AutoCommit enabled at /usr/sbin/rate-o-mat line
91.
rollback ineffective with AutoCommit enabled at /usr/sbin/rate-o-mat line
92.
Error rating CDR batch: No reseller info for user id
909d45f8-1b67-4b44-9655-27a8ccd755d5 found


2013/3/17 Daniel Grotti <dgrotti at sipwise.com>

> Hi,
> this means that you don't have any billing fees for calls to 0034*.
> Please add a proper billing fees and restart rate-o-mat.
>
> Daniel
>
>
>
> Sent from Samsung Mobile
>
>
>
> Marcial Barrios <marcial at tecnimoviles.com> wrote:
>
>
> Hi Kevin, I do the following and works
>
> update cdr set rating_status ="ok" where
>  source_user_id="10d81223-c19c-4030-b2dd-429af4dd8376"
>
> but when I tried to
>
> #service ngcp-rate-o-mat start
>
> I have this other message:
>
> rollback ineffective with AutoCommit enabled at /usr/sbin/rate-o-mat line
> 91.
> rollback ineffective with AutoCommit enabled at /usr/sbin/rate-o-mat line
> 92.
> Error rating CDR batch: No fee info for profile 2 and user '0034922388318'
> or domain '0034922388318 at 91.192.110.234' found
>
>
>
>
>
> 2013/3/17 Kevin Masse <kmasse at questblue.com>
>
>> Are you using XML or SOAP to update subscribers?  If so it is possible
>> that you have deleted the subscriber before the rating process went over
>> the CDRs.  ****
>>
>>
>> The only way to get your rating back running again is to update the
>> accounting.cdr table manually by setting the rating_status to ?ok? for
>> source_subscriber_id=10d81223-c19c-4030-b2dd-429af4dd8376****
>>
>> ** **
>>
>> ** **
>>
>> If you are not using the XML or SOAP to update it still could be possible
>> you deleted a subscriber before calls were rated.****
>>
>> ** **
>>
>> This may have to be repeated a few times to get the status ?ok? in all
>> fields.****
>>
>> ** **
>>
>> Kevin****
>>
>> ** **
>>
>> ** **
>>
>> ** **
>>
>> ** **
>>
>> *From:* spce-user-bounces at lists.sipwise.com [mailto:
>> spce-user-bounces at lists.sipwise.com] *On Behalf Of *Marcial Barrios
>> *Sent:* Sunday, March 17, 2013 12:07 PM
>> *To:* Daniel Grotti; spce-user at lists.sipwise.com
>> *Subject:* Re: [Spce-user] rate-o-mat****
>>
>> ** **
>>
>> ** **
>>
>> ** **
>>
>> 2013/3/17 Marcial Barrios <marcial at tecnimoviles.com>****
>>
>> Hi, Daniel this is what rate-o-mat.log is saying:****
>>
>> ** **
>>
>> Trying to connect to billing db...****
>>
>> Successfully connected to billing db...****
>>
>> Trying to connect to accounting db...****
>>
>> Successfully connected to accounting db...****
>>
>> Up and running.****
>>
>> rate cdr #13****
>>
>> No reseller info for user id 10d81223-c19c-4030-b2dd-429af4dd8376 found**
>> **
>>
>> Error rating CDR batch: No reseller info for user id
>> 10d81223-c19c-4030-b2dd-429af4dd8376 found****
>>
>> ** **
>>
>> ** **
>>
>> 2013/3/17 Daniel Grotti <dgrotti at sipwise.com>****
>>
>> Hi,****
>>
>> please check what /var/log/ngcp/rate-o-mat.log is saying.****
>>
>> ** **
>>
>> Daniel****
>>
>>
>>
>>
>> Sent from Samsung Mobile ****
>>
>>
>>
>>
>> Marcial Barrios <marcial at tecnimoviles.com> wrote:
>>
>> ****
>>
>> ** **
>>
>> No calls are rated.  How can check if rate-o-mat is working properly?****
>>
>> -- ****
>>
>> *Marcial Barrios*****
>>
>> *Administrador de Sistemas*****
>>
>> ****
>>
>> M?vil: (+34) 603 053 043****
>>
>>   Tel:  (+34) 902 955 290****
>>
>>  Fax:  (+34) 822 201 442****
>>
>> email: marcial at tecnimoviles.com****
>>
>>   web: www.tecnimoviles.com****
>>
>>
>>
>> ****
>>
>> ** **
>>
>> -- ****
>>
>> *Marcial Barrios*****
>>
>> *Administrador de Sistemas*****
>>
>> ****
>>
>> M?vil: (+34) 603 053 043****
>>
>>   Tel:  (+34) 902 955 290****
>>
>>  Fax:  (+34) 822 201 442****
>>
>> email: marcial at tecnimoviles.com****
>>
>>   web: www.tecnimoviles.com****
>>
>>
>>
>> ****
>>
>> ** **
>>
>> -- ****
>>
>> *Marcial Barrios*****
>>
>> *Administrador de Sistemas*****
>>
>> ****
>>
>> M?vil: (+34) 603 053 043****
>>
>>   Tel:  (+34) 902 955 290****
>>
>>  Fax:  (+34) 822 201 442****
>>
>> email: marcial at tecnimoviles.com****
>>
>>   web: www.tecnimoviles.com****
>>
>
>
>
> --
> *Marcial Barrios*
> *Administrador de** Sistemas*
> M?vil: (+34) 603 053 043
>   Tel:  (+34) 902 955 290
>  Fax:  (+34) 822 201 442
> email: marcial at tecnimoviles.com
>   web: www.tecnimoviles.com
>



-- 
*Marcial Barrios*
*Administrador de** Sistemas*
M?vil: (+34) 603 053 043
  Tel:  (+34) 902 955 290
 Fax:  (+34) 822 201 442
email: marcial at tecnimoviles.com
  web: www.tecnimoviles.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130317/aae1e1d4/attachment.html>

From jbonilla at sipwise.com  Sun Mar 17 17:51:19 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Sun, 17 Mar 2013 22:51:19 +0100
Subject: [Spce-user] rate-o-mat
In-Reply-To: <CAMnKu2FAYhoKUvvLiLLEbo+HEYYctM4HoPOpy+1b0VEisd73XA@mail.gmail.com>
References: <1xtygsph2mi02o3q5h6d2hy5.1363551530821@email.android.com>
	<CAMnKu2FAYhoKUvvLiLLEbo+HEYYctM4HoPOpy+1b0VEisd73XA@mail.gmail.com>
Message-ID: <20130317225119.70b3b605@quenya>

El Sun, 17 Mar 2013 21:07:10 +0000
Marcial Barrios <marcial at tecnimoviles.com> escribi?:

> Yes you were right I add the billing fees and work but this is like the
> story that never end. This is the new error message:
> 
>  #rollback ineffective with AutoCommit enabled at /usr/sbin/rate-o-mat line
> 91.
> rollback ineffective with AutoCommit enabled at /usr/sbin/rate-o-mat line
> 92.
> Error rating CDR batch: No reseller info for user id
> 909d45f8-1b67-4b44-9655-27a8ccd755d5 found
> 
> 

You can alway use a "match everything" approach. It's normal that you hav all
the destination your customers are using in your billing profile. 

Don't delete the subscribers before the accounting has been finished. You can
terminate them without deleting them.

You can always disable rating and get plain cdrs if you're not interested in
rating.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130317/c1a540a9/attachment.asc>

From jbonilla at sipwise.com  Sun Mar 17 17:53:05 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Sun, 17 Mar 2013 22:53:05 +0100
Subject: [Spce-user] setup pstn gateway in and out
In-Reply-To: <CABAd6xvQi5hsdtQYgRdbN8EyQ68m=hGMPfxPubABy+r3rkwNkQ@mail.gmail.com>
References: <CABAd6xvQi5hsdtQYgRdbN8EyQ68m=hGMPfxPubABy+r3rkwNkQ@mail.gmail.com>
Message-ID: <20130317225305.60ecf372@quenya>

El Sun, 17 Mar 2013 17:05:38 -0400
Vince Oscar <vinceoscar at gmail.com> escribi?:

> I would like to setup gateways to pstn also to freeswitch and asterisk.
> How would I do this? Where would I find the documentation.
> 
> I have tried to setup peering but with failure.  I am surely doing things
> wrong.  Also what log files should I be checking to observe
> success/failures.
> 
> Does a default install include all modules to do this?
> 
> Please advise
> 
> Thank you
> Vince


Yes. The spce is a full platform and you indeed add as many peering groups with
as many billing profiles as you want.

Please detail what you did and what you expected. The peering part is quite
easy and it's present in the handbook.


cheers,

Jon

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130317/87835360/attachment.asc>

From marcial at tecnimoviles.com  Sun Mar 17 18:22:43 2013
From: marcial at tecnimoviles.com (Marcial Barrios)
Date: Sun, 17 Mar 2013 22:22:43 +0000
Subject: [Spce-user] rate-o-mat
In-Reply-To: <CAMnKu2FAYhoKUvvLiLLEbo+HEYYctM4HoPOpy+1b0VEisd73XA@mail.gmail.com>
References: <1xtygsph2mi02o3q5h6d2hy5.1363551530821@email.android.com>
	<CAMnKu2FAYhoKUvvLiLLEbo+HEYYctM4HoPOpy+1b0VEisd73XA@mail.gmail.com>
Message-ID: <CAMnKu2Hvh9wheoWc8sRA6nB22fwsvO6SrKCxuN0_dKifXTaLdw@mail.gmail.com>

Thanks to all, now is working rate-o-mat and the system is billing.

Resume:

Service rate-o-mat stopped because some users were eliminated with calls
not billed (they had not been associated with any fee).

The solution was to manually delete records from MySql database "accounting"
that present problems.

example:

#service ngcp-rate-o-mat start
#rollback ineffective with AutoCommit enabled at /usr/sbin/rate-o-mat line
91. rollback ineffective with AutoCommit enabled at /usr/sbin/rate-o-mat
line 92. Error rating CDR batch: No reseller info for user id
909d45f8-1b67-4b44-9655-27a8ccd755d5
found

MySql query:
delete from cdr where source_user_id="909d45f8-1b67-4b44-9655-27a8ccd755d5"





2013/3/17 Marcial Barrios <marcial at tecnimoviles.com>

> Yes you were right I add the billing fees and work but this is like the
> story that never end. This is the new error message:
>
>  #rollback ineffective with AutoCommit enabled at /usr/sbin/rate-o-mat
> line 91.
>  rollback ineffective with AutoCommit enabled at /usr/sbin/rate-o-mat
> line 92.
> Error rating CDR batch: No reseller info for user id
> 909d45f8-1b67-4b44-9655-27a8ccd755d5 found
>
>
> 2013/3/17 Daniel Grotti <dgrotti at sipwise.com>
>
>> Hi,
>> this means that you don't have any billing fees for calls to 0034*.
>> Please add a proper billing fees and restart rate-o-mat.
>>
>> Daniel
>>
>>
>>
>> Sent from Samsung Mobile
>>
>>
>>
>> Marcial Barrios <marcial at tecnimoviles.com> wrote:
>>
>>
>> Hi Kevin, I do the following and works
>>
>> update cdr set rating_status ="ok" where
>>  source_user_id="10d81223-c19c-4030-b2dd-429af4dd8376"
>>
>> but when I tried to
>>
>> #service ngcp-rate-o-mat start
>>
>> I have this other message:
>>
>> rollback ineffective with AutoCommit enabled at /usr/sbin/rate-o-mat line
>> 91.
>> rollback ineffective with AutoCommit enabled at /usr/sbin/rate-o-mat line
>> 92.
>> Error rating CDR batch: No fee info for profile 2 and user '0034922388318'
>> or domain '0034922388318 at 91.192.110.234' found
>>
>>
>>
>>
>>
>> 2013/3/17 Kevin Masse <kmasse at questblue.com>
>>
>>> Are you using XML or SOAP to update subscribers?  If so it is possible
>>> that you have deleted the subscriber before the rating process went over
>>> the CDRs.  ****
>>>
>>>
>>> The only way to get your rating back running again is to update the
>>> accounting.cdr table manually by setting the rating_status to ?ok? for
>>> source_subscriber_id=10d81223-c19c-4030-b2dd-429af4dd8376****
>>>
>>> ** **
>>>
>>> ** **
>>>
>>> If you are not using the XML or SOAP to update it still could be
>>> possible you deleted a subscriber before calls were rated.****
>>>
>>> ** **
>>>
>>> This may have to be repeated a few times to get the status ?ok? in all
>>> fields.****
>>>
>>> ** **
>>>
>>> Kevin****
>>>
>>> ** **
>>>
>>> ** **
>>>
>>> ** **
>>>
>>> ** **
>>>
>>> *From:* spce-user-bounces at lists.sipwise.com [mailto:
>>> spce-user-bounces at lists.sipwise.com] *On Behalf Of *Marcial Barrios
>>> *Sent:* Sunday, March 17, 2013 12:07 PM
>>> *To:* Daniel Grotti; spce-user at lists.sipwise.com
>>> *Subject:* Re: [Spce-user] rate-o-mat****
>>>
>>> ** **
>>>
>>> ** **
>>>
>>> ** **
>>>
>>> 2013/3/17 Marcial Barrios <marcial at tecnimoviles.com>****
>>>
>>> Hi, Daniel this is what rate-o-mat.log is saying:****
>>>
>>> ** **
>>>
>>> Trying to connect to billing db...****
>>>
>>> Successfully connected to billing db...****
>>>
>>> Trying to connect to accounting db...****
>>>
>>> Successfully connected to accounting db...****
>>>
>>> Up and running.****
>>>
>>> rate cdr #13****
>>>
>>> No reseller info for user id 10d81223-c19c-4030-b2dd-429af4dd8376 found*
>>> ***
>>>
>>> Error rating CDR batch: No reseller info for user id
>>> 10d81223-c19c-4030-b2dd-429af4dd8376 found****
>>>
>>> ** **
>>>
>>> ** **
>>>
>>> 2013/3/17 Daniel Grotti <dgrotti at sipwise.com>****
>>>
>>> Hi,****
>>>
>>> please check what /var/log/ngcp/rate-o-mat.log is saying.****
>>>
>>> ** **
>>>
>>> Daniel****
>>>
>>>
>>>
>>>
>>> Sent from Samsung Mobile ****
>>>
>>>
>>>
>>>
>>> Marcial Barrios <marcial at tecnimoviles.com> wrote:
>>>
>>> ****
>>>
>>> ** **
>>>
>>> No calls are rated.  How can check if rate-o-mat is working properly?***
>>> *
>>>
>>> -- ****
>>>
>>> *Marcial Barrios*****
>>>
>>> *Administrador de Sistemas*****
>>>
>>> ****
>>>
>>> M?vil: (+34) 603 053 043****
>>>
>>>   Tel:  (+34) 902 955 290****
>>>
>>>  Fax:  (+34) 822 201 442****
>>>
>>> email: marcial at tecnimoviles.com****
>>>
>>>   web: www.tecnimoviles.com****
>>>
>>>
>>>
>>> ****
>>>
>>> ** **
>>>
>>> -- ****
>>>
>>> *Marcial Barrios*****
>>>
>>> *Administrador de Sistemas*****
>>>
>>> ****
>>>
>>> M?vil: (+34) 603 053 043****
>>>
>>>   Tel:  (+34) 902 955 290****
>>>
>>>  Fax:  (+34) 822 201 442****
>>>
>>> email: marcial at tecnimoviles.com****
>>>
>>>   web: www.tecnimoviles.com****
>>>
>>>
>>>
>>> ****
>>>
>>> ** **
>>>
>>> -- ****
>>>
>>> *Marcial Barrios*****
>>>
>>> *Administrador de Sistemas*****
>>>
>>> ****
>>>
>>> M?vil: (+34) 603 053 043****
>>>
>>>   Tel:  (+34) 902 955 290****
>>>
>>>  Fax:  (+34) 822 201 442****
>>>
>>> email: marcial at tecnimoviles.com****
>>>
>>>   web: www.tecnimoviles.com****
>>>
>>
>>
>>
>>  --
>> *Marcial Barrios*
>> *Administrador de** Sistemas*
>> M?vil: (+34) 603 053 043
>>   Tel:  (+34) 902 955 290
>>  Fax:  (+34) 822 201 442
>> email: marcial at tecnimoviles.com
>>   web: www.tecnimoviles.com
>>
>
>
>
> --
> *Marcial Barrios*
> *Administrador de** Sistemas*
> M?vil: (+34) 603 053 043
>   Tel:  (+34) 902 955 290
>  Fax:  (+34) 822 201 442
> email: marcial at tecnimoviles.com
>   web: www.tecnimoviles.com
>



-- 
*Marcial Barrios*
*Administrador de** Sistemas*
M?vil: (+34) 603 053 043
  Tel:  (+34) 902 955 290
 Fax:  (+34) 822 201 442
email: marcial at tecnimoviles.com
  web: www.tecnimoviles.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130317/11e2bc0f/attachment.html>

From john.murray at skyracktelecom.com  Mon Mar 18 05:33:35 2013
From: john.murray at skyracktelecom.com (John Murray)
Date: Mon, 18 Mar 2013 09:33:35 -0000
Subject: [Spce-user] Send peer inbound call straight to voicemail
In-Reply-To: <20130315102630.32c7a024@quenya>
References: <021301ce1b80$79b93db0$6d2bb910$@murray@skyracktelecom.com>
	<20130315102630.32c7a024@quenya>
Message-ID: <5146df74.c95fb40a.075c.16e1@mx.google.com>

Jon,

The problem initially to be authentication.

The PBX is actually a gateway and does not register with SPCE. I need to give each gateway subscriber an SPCE subscriber so I use the trusted source model.
IP address of gateway + userpart of From: field matches the subscriber for them to make calls.
For calls to the subscriber I have a static contact/location pointing at the gateway.

However when a call is put out to the subscriber on the gateway and they are not available a call comes in from the gateway to SPCE with 'From:' the original caller 'To:' the voicemail number and 'Divert:' the subscriber details.

This call is rejected with '407 Proxy Authentication Required' as there is no PSTN peer and no trusted source matching the 'From:' because the From: is the originating caller in the PSTN.

I cannot add a peer with the gateway's address as in the logic this is checked before trusted sources and non of my subscribers are then matched.

Would it not be better to check trusted sources first?

Many thanks for your help.

Regards

John

-----Original Message-----
From: spce-user-bounces at lists.sipwise.com [mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of Jon Bonilla (Manwe)
Sent: 15 March 2013 09:27
To: spce-user at lists.sipwise.com
Subject: Re: [Spce-user] Send peer inbound call straight to voicemail

El Thu, 7 Mar 2013 22:09:44 -0000
"John Murray" <john.murray at skyracktelecom.com> escribi?:

> Hi,
> 
>  
> 
> I want to send a specific call type to voicemail.
> 
>  
> 
> I send a call out to a PBX peer who has no voicemail. If the handset 
> is not answered I get a call from the peer with RURI 500 which is the 
> voicemail number, From is the original caller and a Divert: header 
> with the subscriber ddi.
> 
>  
> 
> I know how to take off the Divert header and get the subscriber ddi 
> but what would I put in the RURI to send straight to the subscribers voicemail?
> 
> I've tried prefixing 'vmu' to the ddi but that didn't work.
> 
>  

what was the ngrep-sip (including port 5062), the kamailio-proxy log and the asterisk console output of that call? 





From vinceoscar at gmail.com  Mon Mar 18 10:10:15 2013
From: vinceoscar at gmail.com (Vince Oscar)
Date: Mon, 18 Mar 2013 10:10:15 -0400
Subject: [Spce-user] peer registration fails
Message-ID: <CABAd6xtS0edr_mti=+AKpWHjCarHWA39o7SJ6_R9rOyRdkhmtA@mail.gmail.com>

I cannot get a peer server to PSTN to register.
Server settings are correct with x.x.x.x:5080
5080 is the correct required port

I think the peering rules are incorrect.  First who is outbound, caller or
callee.  I am confused who is who.

second I am confused how to form the peering rules.  for example how would
I allow all incoming, but only allow 10 digit (ac = 3 and sn=7) and 411 and
911

Thank you
Vince
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130318/48928ba7/attachment.html>

From gavin.sweet at skyracktelecom.com  Mon Mar 18 17:24:45 2013
From: gavin.sweet at skyracktelecom.com (Gavin Sweet)
Date: Mon, 18 Mar 2013 21:24:45 -0000
Subject: [Spce-user] CSC www UI layout/customisation
Message-ID: <0ae101ce241f$0371d160$0a557420$@sweet@skyracktelecom.com>

Hi everyone - 

 

I can see from config.yml that some appearance/branding options are
available and that tabs in the view can be turned on/off (e.g.
calllist=1/0).

These options seem to be: account,   addressbook,  callblock, callforward,
calllist, desktop, device, reminder, voicebox

I was wondering, is there a deeper level of modification that can be done as
configuration, or does that require modification to code?

For instance, sub-components on pages, say, to show call list but not
charges.

If there is more, is there a quick "how-to" that someone is able to post to
extend the documentation regarding CSC admin config?

 

 

Many thanks,

Gavin

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130318/fae8aa0f/attachment.html>

From joan at megatro.com  Tue Mar 19 06:13:30 2013
From: joan at megatro.com (Joan Jou)
Date: Tue, 19 Mar 2013 11:13:30 +0100
Subject: [Spce-user] voicemail
Message-ID: <51483A4A.3020809@megatro.com>

Hi,
How can , activate/ desactivate voicemail from user interface and/ admin 
interface /or from user phone?

Thank you
-- 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130319/949181af/attachment.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signatura1.gif
Type: image/gif
Size: 13564 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130319/949181af/attachment.gif>

From rea1ity at kaist.ac.kr  Tue Mar 19 06:17:36 2013
From: rea1ity at kaist.ac.kr (=?UTF-8?B?RGF2aWQgSG8=?=)
Date: Tue, 19 Mar 2013 19:17:36 +0900 (KST)
Subject: [Spce-user] =?utf-8?q?Please_let_me_know_why_I_faced_401/403=28Un?=
	=?utf-8?q?authorized/Invalid_To-URI_Detected=29=2E?=
Message-ID: <51483ef63fea_@_imoxion.com>


p{margin:0;padding:0;}


Hi all,

I tried to register my PSTN Gateway with SIPWISE.
 
the following is communicate log between PSTN Gateway and SIPWISE.
 
Please see the following log;
============================================================================
REGISTER sip:sip.baekdoo.local SIP/2.0
Via:SIP/2.0/UDP 192.168.102.254:5060;branch=z9hG4bK1129594316
From:<sip:5799128 at sip.baekdoo.local>;tag=1561012923
To:<sip:5799128 at sip.baekdoo.local>
Call-ID:952467015 at 192.168.102.254
CSeq:20 REGISTER
Contact:*
Max-Forwards: 70
Expires: 0
user-agent: HUMAN TECHNOLOGY - HMG-1016
Allow:INVITE, CANCEL, PRACK, REFER, ACK, INFO, BYE, NOTIFY
Content-Length:0
 
SIP/2.0 100 Trying
Via:SIP/2.0/UDP 192.168.102.254:5060;rport=5060;branch=z9hG4bK1129594316
From:<sip:5799128 at sip.baekdoo.local>;tag=1561012923
To:<sip:5799128 at sip.baekdoo.local>
Call-ID:952467015 at 192.168.102.254
CSeq:20 REGISTER
Server: Sipwise NGCP Proxy 2.X
Content-Length: 0
 
SIP/2.0 401 Unauthorized
Via:SIP/2.0/UDP 192.168.102.254:5060;rport=5060;branch=z9hG4bK1129594316
From:<sip:5799128 at sip.baekdoo.local>;tag=1561012923
To:<sip:5799128 at sip.baekdoo.local>;tag=1d24a28a0bded6c40d31e6db8aab9ac6.1edb
Call-ID:952467015 at 192.168.102.254
CSeq:20 REGISTER
WWW-Authenticate: Digest realm="sip.baekdoo.local", nonce="UUgkOlFIIw4AagrF0Gf5ISH7uv3SM61l"
Server: Sipwise NGCP Proxy 2.X
Content-Length: 0
 
REGISTER sip:sip.baekdoo.local SIP/2.0
Via:SIP/2.0/UDP 192.168.102.254:5060;branch=z9hG4bK1374480614
From:<sip:5799128 at sip.baekdoo.local>;tag=1784607549
To:<sip:5799128 at sip.baekdoo.local>
Call-ID:952467015 at 192.168.102.254
CSeq:21 REGISTER
Contact:*
Authorization:Digest username="testuser4", realm="sip.baekdoo.local", nonce="UUgkOlFIIw4AagrF0Gf5ISH7uv3SM61l", uri="sip:sip.baekdoo.local", response="ff2c8b1a5c065d441453fed04e3904f6", algorithm=MD5
Max-Forwards: 70
Expires: 0
user-agent: HUMAN TECHNOLOGY - HMG-1016
Allow:INVITE, CANCEL, PRACK, REFER, ACK, INFO, BYE, NOTIFY
Content-Length:0
 
SIP/2.0 100 Trying
Via:SIP/2.0/UDP 192.168.102.254:5060;rport=5060;branch=z9hG4bK1374480614
From:<sip:5799128 at sip.baekdoo.local>;tag=1784607549
To:<sip:5799128 at sip.baekdoo.local>
Call-ID:952467015 at 192.168.102.254
CSeq:21 REGISTER
Server: Sipwise NGCP Proxy 2.X
Content-Length: 0
 
SIP/2.0 403 Invalid To-URI Detected
Via:SIP/2.0/UDP 192.168.102.254:5060;rport=5060;branch=z9hG4bK1374480614
From:<sip:5799128 at sip.baekdoo.local>;tag=1784607549
To:<sip:5799128 at sip.baekdoo.local>;tag=1d24a28a0bded6c40d31e6db8aab9ac6.4d11
Call-ID:952467015 at 192.168.102.254
CSeq:21 REGISTER
Server: Sipwise NGCP Proxy 2.X
Content-Length: 0
 
REGISTER sip:sip.baekdoo.local SIP/2.0
Via:SIP/2.0/UDP 192.168.102.254:5060;branch=z9hG4bK3917095761
From:<sip:5799128 at sip.baekdoo.local>;tag=1232014036
To:<sip:5799128 at sip.baekdoo.local>
Call-ID:94144624 at 192.168.102.254
CSeq:20 REGISTER
Contact:<sip:5799128 at 192.168.102.254:5060>
Max-Forwards: 70
Expires: 3600
user-agent: HUMAN TECHNOLOGY - HMG-1016
Allow:INVITE, CANCEL, PRACK, REFER, ACK, INFO, BYE, NOTIFY
Content-Length:0
 
SIP/2.0 100 Trying
Via:SIP/2.0/UDP 192.168.102.254:5060;rport=5060;branch=z9hG4bK3917095761
From:<sip:5799128 at sip.baekdoo.local>;tag=1232014036
To:<sip:5799128 at sip.baekdoo.local>
Call-ID:94144624 at 192.168.102.254
CSeq:20 REGISTER
Server: Sipwise NGCP Proxy 2.X
Content-Length: 0
 
SIP/2.0 401 Unauthorized
Via:SIP/2.0/UDP 192.168.102.254:5060;rport=5060;branch=z9hG4bK3917095761
From:<sip:5799128 at sip.baekdoo.local>;tag=1232014036
To:<sip:5799128 at sip.baekdoo.local>;tag=1d24a28a0bded6c40d31e6db8aab9ac6.98b4
Call-ID:94144624 at 192.168.102.254
CSeq:20 REGISTER
WWW-Authenticate: Digest realm="sip.baekdoo.local", nonce="UUgkP1FIIxO9M9CJvgTf7zCTDIambZIq"
Server: Sipwise NGCP Proxy 2.X
Content-Length: 0
 
REGISTER sip:sip.baekdoo.local SIP/2.0
Via:SIP/2.0/UDP 192.168.102.254:5060;branch=z9hG4bK517439459
From:<sip:5799128 at sip.baekdoo.local>;tag=192689838
To:<sip:5799128 at sip.baekdoo.local>
Call-ID:94144624 at 192.168.102.254
CSeq:21 REGISTER
Contact:<sip:5799128 at 192.168.102.254:5060>
Authorization:Digest username="testuser4", realm="sip.baekdoo.local", nonce="UUgkP1FIIxO9M9CJvgTf7zCTDIambZIq", uri="sip:sip.baekdoo.local", response="ef1a60da2d0e72ad5bad87ebaa551626", algorithm=MD5
Max-Forwards: 70
Expires: 3600
user-agent: HUMAN TECHNOLOGY - HMG-1016
Allow:INVITE, CANCEL, PRACK, REFER, ACK, INFO, BYE, NOTIFY
Content-Length:0
 
SIP/2.0 100 Trying
Via:SIP/2.0/UDP 192.168.102.254:5060;rport=5060;branch=z9hG4bK517439459
From:<sip:5799128 at sip.baekdoo.local>;tag=192689838
To:<sip:5799128 at sip.baekdoo.local>
Call-ID:94144624 at 192.168.102.254
CSeq:21 REGISTER
Server: Sipwise NGCP Proxy 2.X
Content-Length: 0
 
SIP/2.0 403 Invalid To-URI Detected
Via:SIP/2.0/UDP 192.168.102.254:5060;rport=5060;branch=z9hG4bK517439459
From:<sip:5799128 at sip.baekdoo.local>;tag=192689838
To:<sip:5799128 at sip.baekdoo.local>;tag=1d24a28a0bded6c40d31e6db8aab9ac6.a338
Call-ID:94144624 at 192.168.102.254
CSeq:21 REGISTER
Server: Sipwise NGCP Proxy 2.X
Content-Length: 0
=============================================================================
 
As refer the upper log, PSTN gateway tried three times registerating to SIPWISE but The SIPWISE server didn't register PSTN gateway caused by 
 
unathorized(401) and last time received message as Invalid To-URI Detected(403).
 
So I tried several times that I modified values it related authorized with WISESIP server at PSTN Gateway.
 
You could see the values what I could modified it attached as image file name is PSTN_Gateway.png
 
There have three values(Represented number, ID, Password) it needed to modify if PSTN gateway could register to SIPWISE.
 
But results are still same like upper log.
 
Also I attached the image file(test-gw-1.png) it could you know how to configured at SIP Peerings menu in SIPWISE server.
 
Sorry for long message so it could take your time if you could read it.
 
But I really want to know what is problemed.
 
It'll be appreciate give me advise to solve this problem.

Regards,

David Ho.

 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130319/e29aee08/attachment.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: test-gw-1.png
Type: application/octet-stream
Size: 51372 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130319/e29aee08/attachment.png>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: PSTN_Gateway.png
Type: application/octet-stream
Size: 11219 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130319/e29aee08/attachment-0001.png>

From agranig at sipwise.com  Tue Mar 19 07:02:45 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Tue, 19 Mar 2013 12:02:45 +0100
Subject: [Spce-user] Delete subscriber does not delete 'trusted' entry
In-Reply-To: <035601ce2027$8e13aab0$aa3b0010$@murray@skyracktelecom.com>
References: <035601ce2027$8e13aab0$aa3b0010$@murray@skyracktelecom.com>
Message-ID: <514845D5.5040901@sipwise.com>

Hi John,

On 03/13/2013 09:15 PM, John Murray wrote:
> In v2.7 when I subscriber is deleted the entry in the trusted table is
> not deleted therefore when adding the same pattern to a new subscriber
> the original (deleted) subscriber is matched.

Another stupid mysql "feature" where triggers are not executed on 
foreign key delete cascades.

We've pushed a fix to 2.7, please do:

	apt-get update && apt-get upgrade && ngcp-update-db-schema

You'll still need to remove zombie entries from the kamailio.trusted 
table manually.

Let me know if it works for you.

Andreas



From rea1ity at kaist.ac.kr  Tue Mar 19 10:31:19 2013
From: rea1ity at kaist.ac.kr (=?UTF-8?B?RGF2aWQgSG8=?=)
Date: Tue, 19 Mar 2013 23:31:19 +0900 (KST)
Subject: [Spce-user] =?utf-8?q?peer_registration_fails?=
Message-ID: <51487a803fdf_@_imoxion.com>


p{margin:0;padding:0;}


Vince,
How could you tried to use WireShark?
So you could capture the more detailed packet between PSTN and Peer server it would be help to figure out what was problemed. 
David


-----------------------????-----------------------
???? : Vince Oscar <vinceoscar at gmail.com>
????: spce-user at lists.sipwise.com
????: 2013-03-18 23:10:15 GMT +0900 (Asia/Seoul)
??: [Spce-user] peer registration fails

I cannot get a peer server to PSTN to register.Server settings are correct with x.x.x.x:5080 5080 is the correct required port I think the peering rules are incorrect.  First who is outbound, caller or callee.  I am confused who is who.
 second I am confused how to form the peering rules.  for example how would I allow all incoming, but only allow 10 digit (ac = 3 and sn=7) and 411 and 911 Thank youVince
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130319/563e362e/attachment.html>

From thilo.bangert at gmail.com  Tue Mar 19 14:17:53 2013
From: thilo.bangert at gmail.com (Thilo Bangert)
Date: Tue, 19 Mar 2013 19:17:53 +0100
Subject: [Spce-user] CSC www UI layout/customisation
In-Reply-To: <51478649.41ae0e0a.09f2.55e2SMTPIN_ADDED_BROKEN@mx.google.com>
References: <51478649.41ae0e0a.09f2.55e2SMTPIN_ADDED_BROKEN@mx.google.com>
Message-ID: <2642087.DcGUBIufWZ@marsupilami>

On Monday 18 March 2013 21:24:45 Gavin Sweet wrote:
> Hi everyone -
> 
> 
> 
> I can see from config.yml that some appearance/branding options are
> available and that tabs in the view can be turned on/off (e.g.
> calllist=1/0).
> 
> These options seem to be: account,   addressbook,  callblock, callforward,
> calllist, desktop, device, reminder, voicebox
> 
> I was wondering, is there a deeper level of modification that can be done as
> configuration, or does that require modification to code?
> 
> For instance, sub-components on pages, say, to show call list but not
> charges.
> 
> If there is more, is there a quick "how-to" that someone is able to post to
> extend the documentation regarding CSC admin config?


this is not a how-to per se, but:

check out the files at /usr/share/ngcp-www-csc

it is the path to the perl catalyst files implementing the csc. for example by 
editing the .tt files in lib/csc/root/tt/ and the layout files in 
lib/csc/root/layout will allow you to customize the appearance to your own 
whishes.

good luck!

> 
> 
> 
> 
> 
> Many thanks,
> 
> Gavin



From gavin.sweet at skyracktelecom.com  Tue Mar 19 17:06:55 2013
From: gavin.sweet at skyracktelecom.com (Gavin Sweet)
Date: Tue, 19 Mar 2013 21:06:55 -0000
Subject: [Spce-user] CSC www UI layout/customisation
In-Reply-To: <2642087.DcGUBIufWZ@marsupilami>
References: <51478649.41ae0e0a.09f2.55e2SMTPIN_ADDED_BROKEN@mx.google.com>
	<2642087.DcGUBIufWZ@marsupilami>
Message-ID: <0c6e01ce24e5$aff9af90$0fed0eb0$@sweet@skyracktelecom.com>

Thanks Thilo - very useful.

I wanted to let users login, but not display charges, only calls (as charges
not yet reliable for us).
I just edited calllist.tt, and removed the elements containing the charge
items (fairly obvious).

This:
            [% IF subscriber.account.billing_profile %]
            <td class="anrufkosten">[% Catalyst.loc('charges') %]</td>
            [% END %]

And this:
            [% IF subscriber.account.billing_profile %]
            <td class="anrufkosten">[%
subscriber.account.billing_profile.data.currency %] [% callentry.call_fee
%]</td>
            [% END %]


Works well!



> -----Original Message-----
> From: Thilo Bangert [mailto:thilo.bangert at gmail.com]
> Sent: 19 March 2013 18:18
> To: spce-user at lists.sipwise.com; gavin.sweet at skyracktelecom.com
> Subject: Re: [Spce-user] CSC www UI layout/customisation
> 
> On Monday 18 March 2013 21:24:45 Gavin Sweet wrote:
> > Hi everyone -
> >
> >
> >
> > I can see from config.yml that some appearance/branding options are
> > available and that tabs in the view can be turned on/off (e.g.
> > calllist=1/0).
> >
> > These options seem to be: account,   addressbook,  callblock,
> callforward,
> > calllist, desktop, device, reminder, voicebox
> >
> > I was wondering, is there a deeper level of modification that can be
> > done as configuration, or does that require modification to code?
> >
> > For instance, sub-components on pages, say, to show call list but not
> > charges.
> >
> > If there is more, is there a quick "how-to" that someone is able to
> > post to extend the documentation regarding CSC admin config?
> 
> 
> this is not a how-to per se, but:
> 
> check out the files at /usr/share/ngcp-www-csc
> 
> it is the path to the perl catalyst files implementing the csc. for
> example by editing the .tt files in lib/csc/root/tt/ and the layout
> files in lib/csc/root/layout will allow you to customize the appearance
> to your own whishes.
> 
> good luck!
> 
> >
> >
> >
> >
> >
> > Many thanks,
> >
> > Gavin
> -----
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2013.0.2904 / Virus Database: 2641/6166 - Release Date:
> 03/12/13 Internal Virus Database is out of date.




From jbonilla at sipwise.com  Tue Mar 19 18:05:24 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Tue, 19 Mar 2013 23:05:24 +0100
Subject: [Spce-user] voicemail
In-Reply-To: <51483A4A.3020809@megatro.com>
References: <51483A4A.3020809@megatro.com>
Message-ID: <20130319230524.65a5a596@quenya>

El Tue, 19 Mar 2013 11:13:30 +0100
Joan Jou <joan at megatro.com> escribi?:

> Hi,
> How can , activate/ desactivate voicemail from user interface and/ admin 
> interface /or from user phone?
> 
> Thank you


You can disable it globally in config.yml


For each subscriber, if you don't forward the calls to voicemail it will never
show.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130319/47ef0cc4/attachment.asc>

From jbonilla at sipwise.com  Tue Mar 19 18:06:28 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Tue, 19 Mar 2013 23:06:28 +0100
Subject: [Spce-user] Please let me know why I faced
 401/403(Unauthorized/Invalid To-URI Detected).
In-Reply-To: <51483ef63fea_@_imoxion.com>
References: <51483ef63fea_@_imoxion.com>
Message-ID: <20130319230628.317e5009@quenya>

El Tue, 19 Mar 2013 19:17:36 +0900 (KST)
David Ho<rea1ity at kaist.ac.kr> escribi?:

> 
> p{margin:0;padding:0;}
> 
> 
> Hi all,
> 
> I tried to register my PSTN Gateway with SIPWISE.
>  
>

why? pstn gateways are peers and peer do not register against the spce. They
are authenticated by ip address.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130319/33361afd/attachment.asc>

From jbonilla at sipwise.com  Tue Mar 19 18:08:10 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Tue, 19 Mar 2013 23:08:10 +0100
Subject: [Spce-user] peer registration fails
In-Reply-To: <CABAd6xtS0edr_mti=+AKpWHjCarHWA39o7SJ6_R9rOyRdkhmtA@mail.gmail.com>
References: <CABAd6xtS0edr_mti=+AKpWHjCarHWA39o7SJ6_R9rOyRdkhmtA@mail.gmail.com>
Message-ID: <20130319230810.24ddb12b@quenya>

El Mon, 18 Mar 2013 10:10:15 -0400
Vince Oscar <vinceoscar at gmail.com> escribi?:

> I cannot get a peer server to PSTN to register.
> Server settings are correct with x.x.x.x:5080
> 5080 is the correct required port
> 
> I think the peering rules are incorrect.  First who is outbound, caller or
> callee.  I am confused who is who.
> 
> second I am confused how to form the peering rules.  for example how would
> I allow all incoming, but only allow 10 digit (ac = 3 and sn=7) and 411 and
> 911
> 
> Thank you
> Vince


Do you want to register the spce to a peer? Or a peer to the spce? If it's the
second option don't. Peers don't register.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130319/0ac56956/attachment.asc>

From jbonilla at sipwise.com  Tue Mar 19 18:11:41 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Tue, 19 Mar 2013 23:11:41 +0100
Subject: [Spce-user] BYE not correct when sent by called party
In-Reply-To: <7BA44D46-0ECA-4012-8B14-406B5295A4E9@mac.com>
References: <mailman.1176.1363363037.2547.spce-user@lists.sipwise.com>
	<7BA44D46-0ECA-4012-8B14-406B5295A4E9@mac.com>
Message-ID: <20130319231141.2d5aa4e8@quenya>

El Fri, 15 Mar 2013 17:30:51 +0100
Jirka Jirout <jirka.jirout at mac.com> escribi?:

> Hello,
> 
> > If you have it enabled, please send us the invite sent by the proxy to sems.
> > sems is controlled by custom headers sent by the proxy (be careful as they
> > contain sensitive info). 
> > 
> > If the headers are right, enable sems debug and let's see what it says.
> 
> here is the INVITE. As far as I can tell, the sst headers are there in
> P-App-Param:
> 

Yes. They look good to me.


> I do not see anything in the log. As a matter of fact, the logging seems to
> have stopped since i set the log level to 3 like you advised me :-)
> 

Please put that in sems.conf and restart sems. Sems debug level 3 is really
verbose.



-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130319/489bb053/attachment.asc>

From m.rys at tri-lakes.net  Tue Mar 19 18:29:41 2013
From: m.rys at tri-lakes.net (Marc Rys)
Date: Tue, 19 Mar 2013 17:29:41 -0500 (CDT)
Subject: [Spce-user] Please let me know why I
	faced401/403(Unauthorized/Invalid To-URI Detected).
In-Reply-To: <20130319230628.317e5009@quenya>
Message-ID: <78ca2f18-8c92-4b31-b7d5-4de651d94f1e@usbsnpzimbra01v.servers.rystec.com>

I'm having the same problem.  error 403 on my test phone when trying to dial a 10 digit north America number.

I just setup SPCE from scratch yesterday, my installation went really well, i got my test phone registered successfully, but when I got to the Peer setup my progress came to a stop.  My peer is another soft switch I have on my network and I have it setup using the same method I've used for trunking to other SIP devices...  (Cisco IADs, Asterisk, Broadworks, SIP LD term providers)

I'm so green, can somebody please point me in the direction of where to look at the SIP messaging logs?  I figure this would explain why Im getting the 403 forbidden error.

I'm going to also wireshark my single NIC interface on SPCE right now to see if that gives me any clues

Thanks in advance..

Marc

http://www.tri-lakes.net
http://www.rystec.com




----- Original Message -----
From: "Jon Bonilla" <jbonilla at sipwise.com>
To: spce-user at lists.sipwise.com
Sent: Tuesday, March 19, 2013 5:06:28 PM
Subject: Re: [Spce-user] Please let me know why I faced401/403(Unauthorized/Invalid To-URI Detected).

El Tue, 19 Mar 2013 19:17:36 +0900 (KST)
David Ho<rea1ity at kaist.ac.kr> escribi?:

> 
> p{margin:0;padding:0;}
> 
> 
> Hi all,
> 
> I tried to register my PSTN Gateway with SIPWISE.
>  
>

why? pstn gateways are peers and peer do not register against the spce. They
are authenticated by ip address.


_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com
http://lists.sipwise.com/listinfo/spce-user



From jbonilla at sipwise.com  Tue Mar 19 18:33:02 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Tue, 19 Mar 2013 23:33:02 +0100
Subject: [Spce-user] Send peer inbound call straight to voicemail
In-Reply-To: <5146df74.c95fb40a.075c.16e1@mx.google.com>
References: <021301ce1b80$79b93db0$6d2bb910$@murray@skyracktelecom.com>
	<20130315102630.32c7a024@quenya>
	<5146df74.c95fb40a.075c.16e1@mx.google.com>
Message-ID: <20130319233302.3f768ba9@quenya>

El Mon, 18 Mar 2013 09:33:35 -0000
"John Murray" <john.murray at skyracktelecom.com> escribi?:

> Jon,
> 
> The problem initially to be authentication.
> 
> The PBX is actually a gateway and does not register with SPCE. I need to give
> each gateway subscriber an SPCE subscriber so I use the trusted source model.
> IP address of gateway + userpart of From: field matches the subscriber for
> them to make calls. For calls to the subscriber I have a static
> contact/location pointing at the gateway.
> 
> However when a call is put out to the subscriber on the gateway and they are
> not available a call comes in from the gateway to SPCE with 'From:' the
> original caller 'To:' the voicemail number and 'Divert:' the subscriber
> details.
> 
> This call is rejected with '407 Proxy Authentication Required' as there is no
> PSTN peer and no trusted source matching the 'From:' because the From: is the
> originating caller in the PSTN.
> 
> I cannot add a peer with the gateway's address as in the logic this is
> checked before trusted sources and non of my subscribers are then matched.
> 
> Would it not be better to check trusted sources first?
> 
>

You could modify the proxy configuration and modify ROUTE_AUTH. There you have
the "allow_trusted" function call.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130319/f6ae6a2d/attachment.asc>

From jbonilla at sipwise.com  Tue Mar 19 18:33:54 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Tue, 19 Mar 2013 23:33:54 +0100
Subject: [Spce-user] SOAP array problem
In-Reply-To: <51308D35.4070804@hfcnetwork.hu>
References: <5119FF27.8020107@hfcnetwork.hu> <20130301103828.3b3b31ce@quenya>
	<51308D35.4070804@hfcnetwork.hu>
Message-ID: <20130319233354.7fee53dd@quenya>

El Fri, 01 Mar 2013 12:12:53 +0100
HFC-Network Kft - Szab? J?nos <szabojanos at hfcnetwork.hu> escribi?:

> Hi,
> 
> No, it's still under investigation.
> 
>

Is this still open?

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130319/3a34acdb/attachment.asc>

From jbonilla at sipwise.com  Tue Mar 19 18:37:50 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Tue, 19 Mar 2013 23:37:50 +0100
Subject: [Spce-user] Please let me know why I
 faced401/403(Unauthorized/Invalid To-URI Detected).
In-Reply-To: <78ca2f18-8c92-4b31-b7d5-4de651d94f1e@usbsnpzimbra01v.servers.rystec.com>
References: <20130319230628.317e5009@quenya>
	<78ca2f18-8c92-4b31-b7d5-4de651d94f1e@usbsnpzimbra01v.servers.rystec.com>
Message-ID: <20130319233750.043dd6c7@quenya>

El Tue, 19 Mar 2013 17:29:41 -0500 (CDT)
Marc Rys <m.rys at tri-lakes.net> escribi?:

> I'm having the same problem.  error 403 on my test phone when trying to dial
> a 10 digit north America number.
> 
> I just setup SPCE from scratch yesterday, my installation went really well, i
> got my test phone registered successfully, but when I got to the Peer setup
> my progress came to a stop.  My peer is another soft switch I have on my
> network and I have it setup using the same method I've used for trunking to
> other SIP devices...  (Cisco IADs, Asterisk, Broadworks, SIP LD term
> providers)
> 
> I'm so green, can somebody please point me in the direction of where to look
> at the SIP messaging logs?  I figure this would explain why Im getting the
> 403 forbidden error.
> 
> I'm going to also wireshark my single NIC interface on SPCE right now to see
> if that gives me any clues
> 
> Thanks in advance..
> 
>

ngrep-sip is the quickest way to check what's going on in the network.

checking the proxy logs is usually the best method to see what happened with a
particular call. If the problem is related to permisions or routing there you
have the answer.  You can find the logs under /var/log/ngcp/*

cheers,

Jon

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130319/7df369ab/attachment.asc>

From john.murray at skyracktelecom.com  Tue Mar 19 19:41:00 2013
From: john.murray at skyracktelecom.com (John Murray)
Date: Tue, 19 Mar 2013 23:41:00 -0000
Subject: [Spce-user] Delete subscriber does not delete 'trusted' entry
In-Reply-To: <514845D5.5040901@sipwise.com>
References: <035601ce2027$8e13aab0$aa3b0010$@murray@skyracktelecom.com>
	<514845D5.5040901@sipwise.com>
Message-ID: <5148f78d.8a0db40a.2cc6.180c@mx.google.com>

Andreas, looking good!

Many thanks again.

John

-----Original Message-----
From: spce-user-bounces at lists.sipwise.com
[mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of Andreas Granig
Sent: 19 March 2013 11:03
To: spce-user at lists.sipwise.com
Subject: Re: [Spce-user] Delete subscriber does not delete 'trusted' entry

Hi John,

On 03/13/2013 09:15 PM, John Murray wrote:
> In v2.7 when I subscriber is deleted the entry in the trusted table is 
> not deleted therefore when adding the same pattern to a new subscriber 
> the original (deleted) subscriber is matched.

Another stupid mysql "feature" where triggers are not executed on foreign
key delete cascades.

We've pushed a fix to 2.7, please do:

	apt-get update && apt-get upgrade && ngcp-update-db-schema

You'll still need to remove zombie entries from the kamailio.trusted table
manually.

Let me know if it works for you.

Andreas

_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com
http://lists.sipwise.com/listinfo/spce-user




From john.murray at skyracktelecom.com  Tue Mar 19 19:58:01 2013
From: john.murray at skyracktelecom.com (John Murray)
Date: Tue, 19 Mar 2013 23:58:01 -0000
Subject: [Spce-user] Send peer inbound call straight to voicemail
In-Reply-To: <20130319233302.3f768ba9@quenya>
References: <021301ce1b80$79b93db0$6d2bb910$@murray@skyracktelecom.com>
	<20130315102630.32c7a024@quenya>
	<5146df74.c95fb40a.075c.16e1@mx.google.com>
	<20130319233302.3f768ba9@quenya>
Message-ID: <5148fb8a.a567b40a.0ed1.1a23@mx.google.com>

Jon hi,

So I got this fixed.
I modified ROUTE_AUTH as:

route[ROUTE_AUTH]
{
	if(allow_trusted("$avp(s:ip)", "$avp(s:proto)"))
	{
                if(is_avp_set("$avp(s:peer_uuid)/s"))
                {
                        # use tag-column from trusted-table as uuid for this caller
                        avp_copy("$avp(s:peer_uuid)", "$avp(s:caller_uuid)/d");
  NEW --------->        avp_db_query("select username, domain from subscriber where uuid='$avp(S:caller_uuid)'", "$avp(s:caller_username), $avp(s:caller_domain)");
  NEW --------->         xlog("L_INFO", "Trusted peer with username '$avp(s:caller_username)' - [% logreq -%]\n");
                }
                else
                {
                        # if no uuid is set, use "0" as default uuid
                        $avp(s:caller_uuid) = "0";
                }

Then in route[ROUTE_LOAD_CALLER_PREF]:

if($var(forward) == 1 || $var(cf_loop) == 1 || $avp(s:from_faxserver) == 1)
        {
                $var(pref_domain) = "sip:" + $fU + "@" + $avp(s:acc_caller_domain);
        }
        else
        {
                if ($var(from_trusted) != 1)
                {
                        $var(pref_domain) = "sip:" + $fU + "@" + $avp(s:caller_domain);
                }
                else
                {
NEW ------------->      $var(pref_domain) = "sip:" + $avp(s:caller_username) + "@" + $avp(s:caller_domain);
                        #$var(pref_domain) = $fu;
                }



Many thanks

John


-----Original Message-----
From: Jon Bonilla (Manwe) [mailto:jbonilla at sipwise.com] 
Sent: 19 March 2013 22:33
To: spce-user at lists.sipwise.com
Cc: John Murray
Subject: Re: [Spce-user] Send peer inbound call straight to voicemail

El Mon, 18 Mar 2013 09:33:35 -0000
"John Murray" <john.murray at skyracktelecom.com> escribi?:

> Jon,
> 
> The problem initially to be authentication.
> 
> The PBX is actually a gateway and does not register with SPCE. I need 
> to give each gateway subscriber an SPCE subscriber so I use the trusted source model.
> IP address of gateway + userpart of From: field matches the subscriber 
> for them to make calls. For calls to the subscriber I have a static 
> contact/location pointing at the gateway.
> 
> However when a call is put out to the subscriber on the gateway and 
> they are not available a call comes in from the gateway to SPCE with 
> 'From:' the original caller 'To:' the voicemail number and 'Divert:' 
> the subscriber details.
> 
> This call is rejected with '407 Proxy Authentication Required' as 
> there is no PSTN peer and no trusted source matching the 'From:' 
> because the From: is the originating caller in the PSTN.
> 
> I cannot add a peer with the gateway's address as in the logic this is 
> checked before trusted sources and non of my subscribers are then matched.
> 
> Would it not be better to check trusted sources first?
> 
>

You could modify the proxy configuration and modify ROUTE_AUTH. There you have the "allow_trusted" function call.





From rea1ity at kaist.ac.kr  Wed Mar 20 05:00:45 2013
From: rea1ity at kaist.ac.kr (=?UTF-8?B?RGF2aWQgSG8=?=)
Date: Wed, 20 Mar 2013 18:00:45 +0900 (KST)
Subject: [Spce-user]
	=?utf-8?q?Please_let_me_know_why_Ifaced401/403=28Unau?=
	=?utf-8?q?thorized/Invalid_To-URI_Detected=29=2E?=
Message-ID: <51497c2b3fe4_@_imoxion.com>

Hi Jon,
 
When I open at "/var/log/ngcp/sem/sems.log" file, there are messages same like following;
-----------------------------------------------------------------------------------------------------------------
Mar 20 08:44:14 spce sems[2164]: [#7f601e77d700] [doRegistration, AmSipRegistration.cpp:139] ERROR: failed to send registration.
-----------------------------------------------------------------------------------------------------------------
It seems to occurred "doRegistration" function error in AmSipRegistration.cpp when I tried to register the PSTN gateway.
How do you think it?
Please let me know if you have to solve this problem.
 
Regards,
David Ho
 


-----------------------????-----------------------
???? : "Jon Bonilla (Manwe)" <jbonilla at sipwise.com>
????: spce-user at lists.sipwise.com
????: 2013-03-20 07:37:50 GMT +0900 (Asia/Seoul)
??: Re: [Spce-user] Please let me know why Ifaced401/403(Unauthorized/Invalid To-URI Detected).

El Tue, 19 Mar 2013 17:29:41 -0500 (CDT)

Marc Rys <m.rys at tri-lakes.net> escribi?:



> I'm having the same problem.  error 403 on my test phone when trying to dial

> a 10 digit north America number.

> 

> I just setup SPCE from scratch yesterday, my installation went really well, i

> got my test phone registered successfully, but when I got to the Peer setup

> my progress came to a stop.  My peer is another soft switch I have on my

> network and I have it setup using the same method I've used for trunking to

> other SIP devices...  (Cisco IADs, Asterisk, Broadworks, SIP LD term

> providers)

> 

> I'm so green, can somebody please point me in the direction of where to look

> at the SIP messaging logs?  I figure this would explain why Im getting the

> 403 forbidden error.

> 

> I'm going to also wireshark my single NIC interface on SPCE right now to see

> if that gives me any clues

> 

> Thanks in advance..

> 

>



ngrep-sip is the quickest way to check what's going on in the network.



checking the proxy logs is usually the best method to see what happened with a

particular call. If the problem is related to permisions or routing there you

have the answer.  You can find the logs under /var/log/ngcp/*



cheers,



Jon



_______________________________________________

Spce-user mailing list

Spce-user at lists.sipwise.com

http://lists.sipwise.com/listinfo/spce-user

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130320/ddc242ad/attachment.html>

From jbonilla at sipwise.com  Wed Mar 20 05:08:06 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Wed, 20 Mar 2013 10:08:06 +0100
Subject: [Spce-user] Please let me know why
 Ifaced401/403(Unauthorized/Invalid To-URI Detected).
In-Reply-To: <51497c273fec_@_imoxion.com>
References: <51497c273fec_@_imoxion.com>
Message-ID: <20130320100806.03427c63@quenya>

El Wed, 20 Mar 2013 18:00:45 +0900 (KST)
David Ho<rea1ity at kaist.ac.kr> escribi?:

> Hi Jon,
>  
> When I open at "/var/log/ngcp/sem/sems.log" file, there are messages same
> like following;
> -----------------------------------------------------------------------------------------------------------------
> Mar 20 08:44:14 spce sems[2164]: [#7f601e77d700] [doRegistration,
> AmSipRegistration.cpp:139] ERROR: failed to send registration.
> -----------------------------------------------------------------------------------------------------------------
> It seems to occurred "doRegistration" function error in AmSipRegistration.cpp
> when I tried to register the PSTN gateway. How do you think it? Please let me
> know if you have to solve this problem. Regards,
> David Ho
>  
> 

IMHO you're mixing things. 403 to your subscribers, registration against a peer
gw... please let's go one by one. 

Let's start first with you peer registration. Do you need to register there?
What did you do for that? How does you configuration files look like?

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130320/8414d01a/attachment.asc>

From rea1ity at kaist.ac.kr  Wed Mar 20 06:18:21 2013
From: rea1ity at kaist.ac.kr (=?UTF-8?B?RGF2aWQgSG8=?=)
Date: Wed, 20 Mar 2013 19:18:21 +0900 (KST)
Subject: [Spce-user]
	=?utf-8?q?Please_let_me_know_whyIfaced401/403=28Unaut?=
	=?utf-8?q?horized/Invalid_To-URI_Detected=29=2E?=
Message-ID: <51498ece3fd8_@_imoxion.com>


p{margin:0;padding:0;}


Oh my GOD!
You are right.
I think that I faced seriousley critical problemed.
When I opened the other files as "kamailio-options-proxy_log" and "kamailio-proxy_log" and "kamailio-lb_log" at "/var/log/ngcp/" path,
There have so many logs made it appear New request and request reply per a second.
Moreover, there have the user that I didn't set it but it could working internally.
I.E."R=sip:nagios at voip.sipwise.local F=sip:sipsak at 127.0.0.1:8485 T=sip:nagios at voip.sipwise.local"
My SIPWISE server operate it downloaded from vmware image.
So may I have some missing to configurate of sipwise. :(
I'm just tried to configured it as following SIPWISE document version 2.7 because I don't know well about VoIP.
Now I understand why the document recommended manually install SIPWISE.
Maybe, this problem must be solve first before that I configuration of peer registration modified.
Anyway, I set a peer same like the image files that I attached it name is "test-gw-1.png".
 
 


-----------------------????-----------------------
???? : "Jon Bonilla (Manwe)" <jbonilla at sipwise.com>
????: "spce-user at lists.sipwise.com" <spce-user at lists.sipwise.com>
? ?: David Ho <rea1ity at kaist.ac.kr>
????: 2013-03-20 18:08:06 GMT +0900 (Asia/Seoul)
??: Re: [Spce-user] Please let me know whyIfaced401/403(Unauthorized/Invalid To-URI Detected).

El Wed, 20 Mar 2013 18:00:45 +0900 (KST)

David Ho<rea1ity at kaist.ac.kr> escribi?:



> Hi Jon,

>  

> When I open at "/var/log/ngcp/sem/sems.log" file, there are messages same

> like following;

> -----------------------------------------------------------------------------------------------------------------

> Mar 20 08:44:14 spce sems[2164]: [#7f601e77d700] [doRegistration,

> AmSipRegistration.cpp:139] ERROR: failed to send registration.

> -----------------------------------------------------------------------------------------------------------------

> It seems to occurred "doRegistration" function error in AmSipRegistration.cpp

> when I tried to register the PSTN gateway. How do you think it? Please let me

> know if you have to solve this problem. Regards,

> David Ho

>  

> 



IMHO you're mixing things. 403 to your subscribers, registration against a peer

gw... please let's go one by one. 



Let's start first with you peer registration. Do you need to register there?

What did you do for that? How does you configuration files look like?




-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130320/5e822de2/attachment.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: kamailio-options-proxy_log.txt
Type: application/octet-stream
Size: 4981 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130320/5e822de2/attachment.txt>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: kamailio-proxy_log.txt
Type: application/octet-stream
Size: 6458 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130320/5e822de2/attachment-0001.txt>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: kamailio-lb_log.txt
Type: application/octet-stream
Size: 3943 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130320/5e822de2/attachment-0002.txt>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: test-gw-1.png
Type: application/octet-stream
Size: 51372 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130320/5e822de2/attachment.png>

From jbonilla at sipwise.com  Wed Mar 20 06:35:53 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Wed, 20 Mar 2013 11:35:53 +0100
Subject: [Spce-user] Please let me know
 whyIfaced401/403(Unauthorized/Invalid To-URI Detected).
In-Reply-To: <51498ecb3fe0_@_imoxion.com>
References: <51498ecb3fe0_@_imoxion.com>
Message-ID: <20130320113553.0345a804@quenya>

El Wed, 20 Mar 2013 19:18:21 +0900 (KST)
David Ho<rea1ity at kaist.ac.kr> escribi?:

> 
> p{margin:0;padding:0;}
> 
> 
> Oh my GOD!
> You are right.
> I think that I faced seriousley critical problemed.
> When I opened the other files as "kamailio-options-proxy_log" and
> "kamailio-proxy_log" and "kamailio-lb_log" at "/var/log/ngcp/" path, There
> have so many logs made it appear New request and request reply per a second.
> Moreover, there have the user that I didn't set it but it could working
> internally. I.E."R=sip:nagios at voip.sipwise.local F=sip:sipsak at 127.0.0.1:8485
> T=sip:nagios at voip.sipwise.local" My SIPWISE server operate it downloaded from
> vmware image. So may I have some missing to configurate of sipwise. :( I'm
> just tried to configured it as following SIPWISE document version 2.7 because
> I don't know well about VoIP. Now I understand why the document recommended
> manually install SIPWISE.


No, what you achieve manually installing the provider is more or less the same
as you get with the vmware image. The internal sipsak user you have found is an
internal user for monitoring the system. 

> Maybe, this problem must be solve first before that
> I configuration of peer registration modified. Anyway, I set a peer same like
> the image files that I attached it name is "test-gw-1.png". 


you need to go step by step. Follow the handbook, create you doain, create and
account, create two subscribers and call from one subscriber to the other. Once
you're there we'll talk about peering servers.

I think you are mixing too many things so either you follow the handbook
quick-start or you'll need to learn about Voip (or both).


cheers,

Jon

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130320/5863e37a/attachment.asc>

From vinceoscar at gmail.com  Wed Mar 20 07:42:43 2013
From: vinceoscar at gmail.com (Vince Oscar)
Date: Wed, 20 Mar 2013 07:42:43 -0400
Subject: [Spce-user] confused how to form the peering rules
Message-ID: <CABAd6xsY44VhkXSTefa=aCiR2dGJk8tQvfiP4Qeanj4y0-c5KQ@mail.gmail.com>

I cannot get an outgoing call to complete to PSTN where vendor requires
port 5080
Server settings are correct with x.x.x.x:5080
5080 is the correct required port
Is there a config file the I need to modify?

I think maybe the peering rules are incorrect.

First, who is outbound, the caller or the callee?  I am confused who is who.

Second
I am .

For example,
how would I allow all incoming calls, but only allow 10 digit (ac = 3 and
sn=7) and 411 and
911 for outbound?
Thank you
Vince
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130320/52b2d81f/attachment.html>

From jbonilla at sipwise.com  Wed Mar 20 10:22:43 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Wed, 20 Mar 2013 15:22:43 +0100
Subject: [Spce-user] confused how to form the peering rules
In-Reply-To: <CABAd6xsY44VhkXSTefa=aCiR2dGJk8tQvfiP4Qeanj4y0-c5KQ@mail.gmail.com>
References: <CABAd6xsY44VhkXSTefa=aCiR2dGJk8tQvfiP4Qeanj4y0-c5KQ@mail.gmail.com>
Message-ID: <20130320152243.354755ce@quenya>

El Wed, 20 Mar 2013 07:42:43 -0400
Vince Oscar <vinceoscar at gmail.com> escribi?:

> I cannot get an outgoing call to complete to PSTN where vendor requires
> port 5080

So your peer listens port 5080. No problem as when you define a gateway you can
put there ip and port. You have fields for that.

> Server settings are correct with x.x.x.x:5080
> 5080 is the correct required port
> Is there a config file the I need to modify?

Not for configuring a peer server inside a peer host.


> 
> I think maybe the peering rules are incorrect.
> 
> First, who is outbound, the caller or the callee?  I am confused who is who.
> 

Caller makes the call. Callee is the destination of the call.


> Second
> I am .
> 
> For example,
> how would I allow all incoming calls, but only allow 10 digit (ac = 3 and
> sn=7) and 411 and
> 911 for outbound?
> Thank you
> Vince

Incoming calls are always accepted rom pstn peers by default. Peering rules
tell the system which outgoing calls can be sent to the peering servers in that
group.

If you use callee_preffix you can use just prefixes like 43
 which would be Vienna.

If you want to allow only 10 digit destinations you need to use callee_patter
instead. Something like ^[0.9]{10} 

But peering rules are not the way to allow/disallow your customers to dial to
certain destinations. That's NCOS and call barring. Peering rules is for
sending calls to certain peers for peering contract reasons.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130320/f9888ffe/attachment.asc>

From csaroli at isysp.com  Wed Mar 20 11:11:56 2013
From: csaroli at isysp.com (Christopher Saroli)
Date: Wed, 20 Mar 2013 15:11:56 +0000
Subject: [Spce-user] Billing Profiles/Fees
In-Reply-To: <20130315102112.37d2c298@quenya>
References: <EE7149FDD60B3045A4377F05AAFFF3F118342A46@ISS-Server.ISS.local>
	<20130315102112.37d2c298@quenya>
Message-ID: <EE7149FDD60B3045A4377F05AAFFF3F118358CB8@ISS-Server.ISS.local>

Thanks for responding Jon.  So I started the rate-o-mat service.  Guess it was off. Is this supposed to run on startup? Either way half of it is working now.

So outgoing calls are all being rated and showing up in the exported cdr but incoming calls are not. 

Interesting enough, when I check the customer portal the outgoing calls are listed in the following format +11xxxxxxxxxx while incoming are +xxxxxxxxxx.  The duration of the call is listed in the customer portal for both. Rewrite rule?  

Thanks

---------------------------------
Christopher Saroli
csaroli at isysp.com

???????????????????????????????? 586-226-7800 (Main Office)
                                 586-884-0800 (Direct)


-----Original Message-----
From: spce-user-bounces at lists.sipwise.com [mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of Jon Bonilla
Sent: Friday, March 15, 2013 5:21 AM
To: spce-user at lists.sipwise.com
Subject: Re: [Spce-user] Billing Profiles/Fees

El Wed, 13 Mar 2013 15:52:10 +0000
Christopher Saroli <csaroli at isysp.com> escribi?:

> Little confused on this, every call I make/receive does not get a fee 
> applied to it.  Just trying right now to apply a fee to all calls.  I 
> might be totally wrong here, any info would be appreciated.
> 
> This is my config:
> 
> Billing Profile:
> Interval Charge:  0.02
> Freetime: 	  0
> Free Cash: 	  0
> Monthly Lim	  0
> Lock		  none
> Currency: 	 $
> 
> 
> Bililng Fees
> Destination:  	.*
> Zone:      	All
> Init Rate:	1
> Init interval:	1
> Follow Rate: 	1
> Follow interval: 1
> 
> 
> Also I have no outgoing calls recorded in the CDR under the admin 
> interface but in the customer portal they are there. Is this by design 
> or is something wrong.
>

Well, he cdrs should be shown in both interfaces. We can go there later on. But anyways, to check if your calls are rated fine you should check the exported cdrs (if any).

Is rate-o-mat running in your system? If not, your calls won't be rated and you won't get exported cdrs.


From jbonilla at sipwise.com  Wed Mar 20 11:20:06 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Wed, 20 Mar 2013 16:20:06 +0100
Subject: [Spce-user] Billing Profiles/Fees
In-Reply-To: <EE7149FDD60B3045A4377F05AAFFF3F118358CB8@ISS-Server.ISS.local>
References: <EE7149FDD60B3045A4377F05AAFFF3F118342A46@ISS-Server.ISS.local>
	<20130315102112.37d2c298@quenya>
	<EE7149FDD60B3045A4377F05AAFFF3F118358CB8@ISS-Server.ISS.local>
Message-ID: <20130320162006.1d0f2522@quenya>

El Wed, 20 Mar 2013 15:11:56 +0000
Christopher Saroli <csaroli at isysp.com> escribi?:

> Thanks for responding Jon.  So I started the rate-o-mat service.  Guess it
> was off. Is this supposed to run on startup? Either way half of it is working
> now.
> 

It should start on startup unless you disable it in config.yml. But in that
case the cdrs would be exported without rating information.

> So outgoing calls are all being rated and showing up in the exported cdr but
> incoming calls are not. 

If you want incoming calls to be exported you need to enable that option in
config.yml under the cdrexport section. By default only outgoing sucessfull
calls are exported.

> 
> Interesting enough, when I check the customer portal the outgoing calls are
> listed in the following format +11xxxxxxxxxx while incoming are +xxxxxxxxxx.
> The duration of the call is listed in the customer portal for both. Rewrite
> rule?  
> 

I *strongly* suggest you and every spce user to *always* normalize the incoming
numbers to E164 format and send them in any way you want denormalizing them via
rewrite rules.

The system and all it's components should always work in E164 format. And you
should always work in that way when creating peeing rules, billing profiles...
etc


PD: Did I mention *strongly suggest E164 format*? I meant it.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130320/445df4ec/attachment.asc>

From csaroli at isysp.com  Wed Mar 20 11:58:02 2013
From: csaroli at isysp.com (Christopher Saroli)
Date: Wed, 20 Mar 2013 15:58:02 +0000
Subject: [Spce-user] Billing Profiles/Fees
In-Reply-To: <20130320162006.1d0f2522@quenya>
References: <EE7149FDD60B3045A4377F05AAFFF3F118342A46@ISS-Server.ISS.local>
	<20130315102112.37d2c298@quenya>
	<EE7149FDD60B3045A4377F05AAFFF3F118358CB8@ISS-Server.ISS.local>
	<20130320162006.1d0f2522@quenya>
Message-ID: <EE7149FDD60B3045A4377F05AAFFF3F118358D13@ISS-Server.ISS.local>

Thanks Jon, I will check the config and enable the incoming.

As for rewrite rules, when my peering sends north American calls to me it is not in E.164.  It is without the CC. So I have a rewrite rule in place to add the 1. All the phone systems here do not want the CC so I have to strip the 1 when I send it back out. Is there a better way for me to go about this. I have the following rewrite rules in place

(Applied to domain)
Outbound Rewite Rules for Callee 
^(1)([1-9][0-9]+)$                   \2                          
Inbound Rewrite Rules for Callee
^([1-9][0-9]{9})                        1\1   (needed this in order for subscribers within sipwise to call each other)


(Applied to my peering)
Outbound rewrite rules for Calle
^([0-9]{7})$                             ${caller_cc}${caller_ac}\1
Inbound Rewrite rules for Callee
^([1-9][0-9]{9})                         1\1

Thanks for all your help








---------------------------------
Christopher Saroli
csaroli at isysp.com

?????????????????


-----Original Message-----
From: Jon Bonilla [mailto:jbonilla at sipwise.com] 
Sent: Wednesday, March 20, 2013 11:20 AM
To: spce-user@
Cc: Christopher Saroli
Subject: Re: [Spce-user] Billing Profiles/Fees

El Wed, 20 Mar 2013 15:11:56 +0000
Christopher Saroli <csaroli at isysp.com> escribi?:

> Thanks for responding Jon.  So I started the rate-o-mat service.  
> Guess it was off. Is this supposed to run on startup? Either way half 
> of it is working now.
> 

It should start on startup unless you disable it in config.yml. But in that case the cdrs would be exported without rating information.

> So outgoing calls are all being rated and showing up in the exported 
> cdr but incoming calls are not.

If you want incoming calls to be exported you need to enable that option in config.yml under the cdrexport section. By default only outgoing sucessfull calls are exported.

> 
> Interesting enough, when I check the customer portal the outgoing 
> calls are listed in the following format +11xxxxxxxxxx while incoming are +xxxxxxxxxx.
> The duration of the call is listed in the customer portal for both. 
> Rewrite rule?
> 

I *strongly* suggest you and every spce user to *always* normalize the incoming numbers to E164 format and send them in any way you want denormalizing them via rewrite rules.

The system and all it's components should always work in E164 format. And you should always work in that way when creating peeing rules, billing profiles...
etc


PD: Did I mention *strongly suggest E164 format*? I meant it.


From jbonilla at sipwise.com  Wed Mar 20 12:31:27 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Wed, 20 Mar 2013 17:31:27 +0100
Subject: [Spce-user] Billing Profiles/Fees
In-Reply-To: <EE7149FDD60B3045A4377F05AAFFF3F118358D13@ISS-Server.ISS.local>
References: <EE7149FDD60B3045A4377F05AAFFF3F118342A46@ISS-Server.ISS.local>
	<20130315102112.37d2c298@quenya>
	<EE7149FDD60B3045A4377F05AAFFF3F118358CB8@ISS-Server.ISS.local>
	<20130320162006.1d0f2522@quenya>
	<EE7149FDD60B3045A4377F05AAFFF3F118358D13@ISS-Server.ISS.local>
Message-ID: <20130320173127.606ca7e7@quenya>

El Wed, 20 Mar 2013 15:58:02 +0000
Christopher Saroli <csaroli at isysp.com> escribi?:

> Thanks Jon, I will check the config and enable the incoming.
> 
> As for rewrite rules, when my peering sends north American calls to me it is
> not in E.164.  It is without the CC. So I have a rewrite rule in place to add
> the 1. All the phone systems here do not want the CC so I have to strip the 1
> when I send it back out.

Yes. that's exactly how it should be and that's why we have rewrite rules
there. In every install I need to normalize what peers/subscribers send and
also denormalize what I send them to the format they expect. 

This might be confusing at the beginning but once you're done rewrite rules are
your best friends.


> Is there a better way for me to go about this. I
> have the following rewrite rules in place
> 
> (Applied to domain)
> Outbound Rewite Rules for Callee 
> ^(1)([1-9][0-9]+)$                   \2                          
> Inbound Rewrite Rules for Callee
> ^([1-9][0-9]{9})                        1\1   (needed this in order for
> subscribers within sipwise to call each other)
> 

Looks good as starting point. remember that you have to normalize everything
you'd expect a user to dial. International, emergency, national, short
numbers... etc


> 
> (Applied to my peering)
> Outbound rewrite rules for Calle
> ^([0-9]{7})$                             ${caller_cc}${caller_ac}\1
> Inbound Rewrite rules for Callee
> ^([1-9][0-9]{9})                         1\1
> 
> Thanks for all your help
> 
> 

again looks good as starting point.


You'll see it in kamailio-proxy logs. How the numbers are normalized and once
you create peering rules and ncos levels in E164 format you'll see that calls
cannot be routed or rated in case they were not normalized.


You're in the right path :)

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130320/e369f3ba/attachment.asc>

From rea1ity at kaist.ac.kr  Wed Mar 20 23:24:34 2013
From: rea1ity at kaist.ac.kr (=?UTF-8?B?RGF2aWQgSG8=?=)
Date: Thu, 21 Mar 2013 12:24:34 +0900 (KST)
Subject: [Spce-user]
	=?utf-8?q?Please_let_me_knowwhyIfaced401/403=28Unauth?=
	=?utf-8?q?orized/Invalid_To-URI_Detected=29=2E?=
Message-ID: <514a7ee23fe6_@_imoxion.com>

 
 
Jon,
 
I've already followed up the document that you provide "The sip:provider CE Handbook 2.7" so I'm followed setup 1 to 6 chapter and almost setting same like the document.
And now I'm tried to modify values of Peering Servers several times and you could know where I'm modified the values at "6.4.4. Authenticating and Registering against Peering Servers".
the URL following;
http://www.sipwise.com/doc/2.7/spce/ar01s06.html#_authenticating_and_registering_against_peering_servers
 
So manytimes, SIPWISE server denied my PSTN gateway caused by Unauthorized(401).
And I found what is problemed via sems log file("/var/log/ngcp/sem/sems.log") it appeared error message before I mentioned same like following;
-----------------------------------------------------------------------------------------------------------------
Mar 20 08:44:14 spce sems[2164]: [#7f601e77d700] [doRegistration, AmSipRegistration.cpp:139] ERROR: failed to send registration.
-----------------------------------------------------------------------------------------------------------------
So 'doRegistration function in AmSipRegistration.cpp file occurred ERROR message.
If you were me, how could solve the error occurred?
 
And it is another one that I afraid thing.
the log files in SIPWISE as kamailio-lb.log and kamailio-options-proxy.log increased capacity too much.
my kamailio-lb.log file capacity is over than 28MB and kamailio-options-proxy.log file capacity is more than 7MB.
The reason of the log too much increased capacity caused by stacked approax. four to five log messages per a second!
Is it alright to leave it even though it will be huge capacity log file after than this log file could occurr system error for leak of hard disk capacity.
 
David


-----------------------????-----------------------
???? : "Jon Bonilla (Manwe)" <jbonilla at sipwise.com>
????: "spce-user at lists.sipwise.com" <spce-user at lists.sipwise.com>
? ?: David Ho <rea1ity at kaist.ac.kr>
????: 2013-03-20 19:35:53 GMT +0900 (Asia/Seoul)
??: Re: [Spce-user] Please let me knowwhyIfaced401/403(Unauthorized/Invalid To-URI Detected).

El Wed, 20 Mar 2013 19:18:21 +0900 (KST)

David Ho<rea1ity at kaist.ac.kr> escribi?:



> 

> p{margin:0;padding:0;}

> 

> 

> Oh my GOD!

> You are right.

> I think that I faced seriousley critical problemed.

> When I opened the other files as "kamailio-options-proxy_log" and

> "kamailio-proxy_log" and "kamailio-lb_log" at "/var/log/ngcp/" path, There

> have so many logs made it appear New request and request reply per a second.

> Moreover, there have the user that I didn't set it but it could working

> internally. I.E."R=sip:nagios at voip.sipwise.local F=sip:sipsak at 127.0.0.1:8485

> T=sip:nagios at voip.sipwise.local" My SIPWISE server operate it downloaded from

> vmware image. So may I have some missing to configurate of sipwise. :( I'm

> just tried to configured it as following SIPWISE document version 2.7 because

> I don't know well about VoIP. Now I understand why the document recommended

> manually install SIPWISE.





No, what you achieve manually installing the provider is more or less the same

as you get with the vmware image. The internal sipsak user you have found is an

internal user for monitoring the system. 



> Maybe, this problem must be solve first before that

> I configuration of peer registration modified. Anyway, I set a peer same like

> the image files that I attached it name is "test-gw-1.png". 





you need to go step by step. Follow the handbook, create you doain, create and

account, create two subscribers and call from one subscriber to the other. Once

you're there we'll talk about peering servers.



I think you are mixing too many things so either you follow the handbook

quick-start or you'll need to learn about Voip (or both).





cheers,



Jon



-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130321/c752c8f0/attachment.html>

From jbonilla at sipwise.com  Thu Mar 21 06:24:36 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Thu, 21 Mar 2013 11:24:36 +0100
Subject: [Spce-user] Please let me
 knowwhyIfaced401/403(Unauthorized/Invalid To-URI Detected).
In-Reply-To: <514a7edf3fe1_@_imoxion.com>
References: <514a7edf3fe1_@_imoxion.com>
Message-ID: <20130321112436.5a1fcc54@quenya>

El Thu, 21 Mar 2013 12:24:34 +0900 (KST)
David Ho<rea1ity at kaist.ac.kr> escribi?:

>  
>  
> Jon,
>  
> I've already followed up the document that you provide "The sip:provider CE
> Handbook 2.7" so I'm followed setup 1 to 6 chapter and almost setting same
> like the document. And now I'm tried to modify values of Peering Servers
> several times and you could know where I'm modified the values at "6.4.4.
> Authenticating and Registering against Peering Servers". the URL following;
> http://www.sipwise.com/doc/2.7/spce/ar01s06.html#_authenticating_and_registering_against_peering_servers
> So manytimes, SIPWISE server denied my PSTN gateway caused by
> Unauthorized(401). And I found what is problemed via sems log
> file("/var/log/ngcp/sem/sems.log") it appeared error message before I
> mentioned same like following;


So you already created two subscribers and you ca call from one to the other?

If so, send us what you did to register against your peer server and all the
related information, network capture and configuration files. We cannot guess
what's going on without information.


> -----------------------------------------------------------------------------------------------------------------
> Mar 20 08:44:14 spce sems[2164]: [#7f601e77d700] [doRegistration,
> AmSipRegistration.cpp:139] ERROR: failed to send registration.
> -----------------------------------------------------------------------------------------------------------------
> So 'doRegistration function in AmSipRegistration.cpp file occurred ERROR
> message. If you were me, how could solve the error occurred? And it is
> another one that I afraid thing. the log files in SIPWISE as kamailio-lb.log
> and kamailio-options-proxy.log increased capacity too much. my
> kamailio-lb.log file capacity is over than 28MB and
> kamailio-options-proxy.log file capacity is more than 7MB. The reason of the
> log too much increased capacity caused by stacked approax. four to five log
> messages per a second! Is it alright to leave it even though it will be huge
> capacity log file after than this log file could occurr system error for leak
> of hard disk capacity. David


You can expect several GB of logs per day in a production system. 28MB is a
very small log. THe spce logs a lot of information for each request.




-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130321/a0e215ad/attachment.asc>

From nh.developer at gmail.com  Thu Mar 21 07:29:09 2013
From: nh.developer at gmail.com (Developing Tomorrow)
Date: Thu, 21 Mar 2013 12:29:09 +0100
Subject: [Spce-user] Running Sipwise on Centos
Message-ID: <001c01ce2627$4eb52f00$ec1f8d00$@gmail.com>

Hi.,

 

I have tested sipwise on debian, as per the recommendation.

But the truth is that I prefer centos 6

 

Could someone direct me to the source code to compile or to some
instructions on 

 How to install it on centos?

 

Help would be much appreciated.

cheers

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130321/9f4c9e9f/attachment.html>

From jbonilla at sipwise.com  Thu Mar 21 08:39:58 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Thu, 21 Mar 2013 13:39:58 +0100
Subject: [Spce-user] Running Sipwise on Centos
In-Reply-To: <001c01ce2627$4eb52f00$ec1f8d00$@gmail.com>
References: <001c01ce2627$4eb52f00$ec1f8d00$@gmail.com>
Message-ID: <20130321133958.7967731a@quenya>

El Thu, 21 Mar 2013 12:29:09 +0100
"Developing Tomorrow" <nh.developer at gmail.com> escribi?:

> Hi.,
> 
>  
> 
> I have tested sipwise on debian, as per the recommendation.
> 
> But the truth is that I prefer centos 6
> 
>  
> 
> Could someone direct me to the source code to compile or to some
> instructions on 
> 
>  How to install it on centos?
> 
>  
> 
> Help would be much appreciated.
> 


Sorry but I won't help you here. I don't know if it's even possible to compile
all the packages on centos and change the paths of the files that should be in
different places. This is more about religion, holy war, vim vs emacs or
whatever you'd like to call the rpm vs deb or debian vs any_other_distro.

Debian is the only distro we support and amd64 the only arch.  

This goes far away from what the spce mailing list can offer as support.

Maybe we could change the thread and talk about something more important: How
can anyone prefer Centos over Debian? 

:P


cheers,

Jon

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130321/ebacb4a7/attachment.asc>

From alesg at coit.es  Thu Mar 21 11:43:58 2013
From: alesg at coit.es (Alejandro Sanchez Gimenez)
Date: Thu, 21 Mar 2013 16:43:58 +0100 (CET)
Subject: [Spce-user] Fwd: ngcp-fraud-auto-lock
In-Reply-To: <22675ad6-47d1-4690-ace6-dc6f663687db@mail.coit.es>
Message-ID: <84392596-5fc6-46c3-b4e2-c30c75f3b65b@mail.coit.es>

Hi again!! 


I?m still blocked at this point. I can?t get fraud auto-lock working. In my test configuration I have in my billing profile: 


Fraud Monthly Limit: 1 (this is "one cent", as I understand from documentation) 
Fraund Monthly Lock: incoming and outgoing 


I have only one catch-all fee for this billing profile, where init rate is 15 (cents), so in the first call, the account reaches its limit. I can see cdrs for the subscriber working correctly, with correct data in duration and charges. For example, for the subscriber?s first call: 


duration: 00:05 
charges: 0.15? 


At this point I execute "ngcp-fraud-auto-lock" but the subscriber can still make and receive calls. Where could be the problem? Rating is also working properly, because I have also tested with interval free time/cash and I get correct data. 


Thanks a lot. 


----- Mensaje original -----

De: "Alejandro Sanchez Gimenez" <alesg at coit.es> 
Para: spce-user at lists.sipwise.com 
Enviados: Martes, 26 de Febrero 2013 12:59:41 
Asunto: ngcp-fraud-auto-lock 


Hi all!! 


I have read it is possible to change the cronjob ngcp-fraud-auto-lock to be executed more than once a day. Would it be any problem if it is executed for exmple each five minutes? Could the system play any announcement if the subscriber?s balance is about to expire? 


Thanks a lot!! 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130321/63456217/attachment.html>

From rfuchs at sipwise.com  Thu Mar 21 13:33:09 2013
From: rfuchs at sipwise.com (Richard Fuchs)
Date: Thu, 21 Mar 2013 13:33:09 -0400
Subject: [Spce-user] Fwd: ngcp-fraud-auto-lock
In-Reply-To: <84392596-5fc6-46c3-b4e2-c30c75f3b65b@mail.coit.es>
References: <84392596-5fc6-46c3-b4e2-c30c75f3b65b@mail.coit.es>
Message-ID: <514B4455.4090007@sipwise.com>

Hi,

Can you enter your mysql console and post the output of the following
commands, substituting $ID for the account ID of your subscriber?

> select * from billing.contracts where id = $ID;

> select * from billing.contract_balances where contract_id = $ID;

> select * from billing.billing_mappings m, billing.billing_profiles p
where p.id = m.billing_profile_id and m.contract_id = $ID;

Thanks


On 03/21/13 11:43, Alejandro Sanchez Gimenez wrote:
> Hi again!!
> 
> I?m still blocked at this point. I can?t get fraud auto-lock working. In
> my test configuration I have in my billing profile:
> 
> Fraud Monthly Limit: 1 (this is "one cent", as I understand from
> documentation)
> Fraund Monthly Lock: incoming and outgoing
> 
> I have only one catch-all fee for this billing profile, where init rate
> is 15 (cents), so in the first call, the account reaches its limit. I
> can see cdrs for the subscriber working correctly, with correct data in
> duration and charges. For example, for the subscriber?s first call:
> 
> duration: 00:05
> charges: 0.15?
> 
> At this point I execute "ngcp-fraud-auto-lock" but the subscriber can
> still make and receive calls. Where could be the problem? Rating is also
> working properly, because I have also tested with interval free
> time/cash and I get correct data.
> 
> Thanks a lot.
> 
> 
> ------------------------------------------------------------------------
> *De: *"Alejandro Sanchez Gimenez" <alesg at coit.es>
> *Para: *spce-user at lists.sipwise.com
> *Enviados: *Martes, 26 de Febrero 2013 12:59:41
> *Asunto: *ngcp-fraud-auto-lock
> 
> Hi all!!
> 
> I have read it is possible to change the cronjob ngcp-fraud-auto-lock to
> be executed more than once a day. Would it be any problem if it is
> executed for exmple each five minutes? Could the system play any
> announcement if the subscriber?s balance is about to expire?
> 
> Thanks a lot!!
> 
> 
> 
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 900 bytes
Desc: OpenPGP digital signature
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130321/e36cefe0/attachment.asc>

From alesg at coit.es  Thu Mar 21 15:00:45 2013
From: alesg at coit.es (Alejandro Sanchez Gimenez)
Date: Thu, 21 Mar 2013 20:00:45 +0100 (CET)
Subject: [Spce-user]  Fwd: ngcp-fraud-auto-lock
Message-ID: <52a220ff-2caa-4f08-a955-667f985fb213@mail.coit.es>

Hello Richard, 


this is what I get: 



mysql> select * from contracts where id = 22; 
+----+-------------+-------------+------------+----------+--------+-------------+---------------------+---------------------+---------------------+---------------------+ 
| id | customer_id | reseller_id | contact_id | order_id | status | external_id | modify_timestamp | create_timestamp | activate_timestamp | terminate_timestamp | 
+----+-------------+-------------+------------+----------+--------+-------------+---------------------+---------------------+---------------------+---------------------+ 
| 22 | NULL | 1 | NULL | NULL | active | asdfasdfsdf | 2013-03-21 15:02:14 | 2013-03-21 15:02:14 | 2013-03-21 15:02:14 | NULL | 
+----+-------------+-------------+------------+----------+--------+-------------+---------------------+---------------------+---------------------+---------------------+ 
1 row in set (0.00 sec) 




mysql> select * from contract_balances where contract_id = 22; 
+----+-------------+--------------+-----------------------+-------------------+----------------------------+---------------------+---------------------+------------+ 
| id | contract_id | cash_balance | cash_balance_interval | free_time_balance | free_time_balance_interval | start | end | invoice_id | 
+----+-------------+--------------+-----------------------+-------------------+----------------------------+---------------------+---------------------+------------+ 
| 21 | 22 | 0 | 0 | 0 | 0 | 2013-03-01 00:00:00 | 2013-03-31 23:59:59 | NULL | 
+----+-------------+--------------+-----------------------+-------------------+----------------------------+---------------------+---------------------+------------+ 
1 row in set (0.00 sec) 




mysql> select * from billing.billing_mappings m, billing.billing_profiles p 
-> where p.id = m.billing_profile_id and m.contract_id = 22; 
+----+------------+----------+--------------------+-------------+------------+----+-------------+--------------+--------------+---------+-----------------+--------------------+--------------------+---------------+----------------+----------------------+---------------------+-------------------------------+-------------------+------------------+--------------------+----------+----------+--------------+ 
| id | start_date | end_date | billing_profile_id | contract_id | product_id | id | reseller_id | handle | name | prepaid | interval_charge | interval_free_time | interval_free_cash | interval_unit | interval_count | fraud_interval_limit | fraud_interval_lock | fraud_interval_notify | fraud_daily_limit | fraud_daily_lock | fraud_daily_notify | currency | vat_rate | vat_included | 
+----+------------+----------+--------------------+-------------+------------+----+-------------+--------------+--------------+---------+-----------------+--------------------+--------------------+---------------+----------------+----------------------+---------------------+-------------------------------+-------------------+------------------+--------------------+----------+----------+--------------+ 
| 22 | NULL | NULL | 4 | 22 | NULL | 4 | 1 | Tarifa_plana | Tarifa plana | 0 | 0 | 0 | 0 | month | 1 | 100 | 3 | | 100 | 3 | NULL | ?? | 0 | 0 | 
+----+------------+----------+--------------------+-------------+------------+----+-------------+--------------+--------------+---------+-----------------+--------------------+--------------------+---------------+----------------+----------------------+---------------------+-------------------------------+-------------------+------------------+--------------------+----------+----------+--------------+ 
1 row in set (0.00 sec) 




Thank you very much!!! 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130321/c59db2c1/attachment.html>

From jbonilla at sipwise.com  Thu Mar 21 14:48:35 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Thu, 21 Mar 2013 19:48:35 +0100
Subject: [Spce-user] Fwd: ngcp-fraud-auto-lock
In-Reply-To: <52a220ff-2caa-4f08-a955-667f985fb213@mail.coit.es>
References: <52a220ff-2caa-4f08-a955-667f985fb213@mail.coit.es>
Message-ID: <20130321194835.0807d079@quenya>

El Thu, 21 Mar 2013 20:00:45 +0100 (CET)
Alejandro Sanchez Gimenez <alesg at coit.es> escribi?:

> Hello Richard, 
> 
> 
> this is what I get: 
> 

Please, can you repeat the same queries with \G at the end?

Like

mysql> select * from contracts where id = 22\G;



btw I don't know why your mail is not an answer to the one Richard sent. It's
in a new thread.

 

> 
> 
> 


> +----+-------------+-------------+------------+----------+--------+-------------+---------------------+---------------------+---------------------+---------------------+ 
> | id | customer_id | reseller_id | contact_id | order_id | status |
> external_id | modify_timestamp | create_timestamp | activate_timestamp |
> terminate_timestamp |
> +----+-------------+-------------+------------+----------+--------+-------------+---------------------+---------------------+---------------------+---------------------+
> | 22 | NULL | 1 | NULL | NULL | active | asdfasdfsdf | 2013-03-21 15:02:14 |
> 2013-03-21 15:02:14 | 2013-03-21 15:02:14 | NULL |
> +----+-------------+-------------+------------+----------+--------+-------------+---------------------+---------------------+---------------------+---------------------+
> 1 row in set (0.00 sec) 
> 
> 
> 
> 
> mysql> select * from contract_balances where contract_id = 22; 
> +----+-------------+--------------+-----------------------+-------------------+----------------------------+---------------------+---------------------+------------+ 
> | id | contract_id | cash_balance | cash_balance_interval | free_time_balance
> | free_time_balance_interval | start | end | invoice_id |
> +----+-------------+--------------+-----------------------+-------------------+----------------------------+---------------------+---------------------+------------+
> | 21 | 22 | 0 | 0 | 0 | 0 | 2013-03-01 00:00:00 | 2013-03-31 23:59:59 | NULL
> |
> +----+-------------+--------------+-----------------------+-------------------+----------------------------+---------------------+---------------------+------------+
> 1 row in set (0.00 sec) 
> 
> 
> 
> 
> mysql> select * from billing.billing_mappings m, billing.billing_profiles p 
> -> where p.id = m.billing_profile_id and m.contract_id = 22; 
> +----+------------+----------+--------------------+-------------+------------+----+-------------+--------------+--------------+---------+-----------------+--------------------+--------------------+---------------+----------------+----------------------+---------------------+-------------------------------+-------------------+------------------+--------------------+----------+----------+--------------+ 
> | id | start_date | end_date | billing_profile_id | contract_id | product_id
> | id | reseller_id | handle | name | prepaid | interval_charge |
> interval_free_time | interval_free_cash | interval_unit | interval_count |
> fraud_interval_limit | fraud_interval_lock | fraud_interval_notify |
> fraud_daily_limit | fraud_daily_lock | fraud_daily_notify | currency |
> vat_rate | vat_included |
> +----+------------+----------+--------------------+-------------+------------+----+-------------+--------------+--------------+---------+-----------------+--------------------+--------------------+---------------+----------------+----------------------+---------------------+-------------------------------+-------------------+------------------+--------------------+----------+----------+--------------+
> | 22 | NULL | NULL | 4 | 22 | NULL | 4 | 1 | Tarifa_plana | Tarifa plana | 0
> | 0 | 0 | 0 | month | 1 | 100 | 3 | | 100 | 3 | NULL | ?? | 0 | 0 |
> +----+------------+----------+--------------------+-------------+------------+----+-------------+--------------+--------------+---------+-----------------+--------------------+--------------------+---------------+----------------+----------------------+---------------------+-------------------------------+-------------------+------------------+--------------------+----------+----------+--------------+
> 1 row in set (0.00 sec) 
> 
> 
> 
> 
> Thank you very much!!! 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130321/c6bdda5b/attachment.asc>

From rfuchs at sipwise.com  Thu Mar 21 15:19:07 2013
From: rfuchs at sipwise.com (Richard Fuchs)
Date: Thu, 21 Mar 2013 15:19:07 -0400
Subject: [Spce-user] Fwd: ngcp-fraud-auto-lock
In-Reply-To: <52a220ff-2caa-4f08-a955-667f985fb213@mail.coit.es>
References: <52a220ff-2caa-4f08-a955-667f985fb213@mail.coit.es>
Message-ID: <514B5D2B.30008@sipwise.com>

There's a problem with the ratings, the contract_balances don't show any
charges being made to the account, everything is zero. If the CDRs are
rated correctly, then the association between calling subscriber and
account must be incorrect. Check the source_account_id field of the
CDRs, it should contain 22, and the customer_cost field should contain
the cost.

BR


On 03/21/13 15:00, Alejandro Sanchez Gimenez wrote:
> Hello Richard,
> 
> this is what I get:
> 
> mysql> select * from contracts where id = 22;
> +----+-------------+-------------+------------+----------+--------+-------------+---------------------+---------------------+---------------------+---------------------+
> | id | customer_id | reseller_id | contact_id | order_id | status |
> external_id | modify_timestamp    | create_timestamp    |
> activate_timestamp  | terminate_timestamp |
> +----+-------------+-------------+------------+----------+--------+-------------+---------------------+---------------------+---------------------+---------------------+
> | 22 |        NULL |           1 |       NULL |     NULL | active |
> asdfasdfsdf | 2013-03-21 15:02:14 | 2013-03-21 15:02:14 | 2013-03-21
> 15:02:14 |                NULL |
> +----+-------------+-------------+------------+----------+--------+-------------+---------------------+---------------------+---------------------+---------------------+
> 1 row in set (0.00 sec)
> 
> 
> mysql> select * from contract_balances where contract_id = 22;
> +----+-------------+--------------+-----------------------+-------------------+----------------------------+---------------------+---------------------+------------+
> | id | contract_id | cash_balance | cash_balance_interval |
> free_time_balance | free_time_balance_interval | start               |
> end                 | invoice_id |
> +----+-------------+--------------+-----------------------+-------------------+----------------------------+---------------------+---------------------+------------+
> | 21 |          22 |            0 |                     0 |            
>     0 |                          0 | 2013-03-01 00:00:00 | 2013-03-31
> 23:59:59 |       NULL |
> +----+-------------+--------------+-----------------------+-------------------+----------------------------+---------------------+---------------------+------------+
> 1 row in set (0.00 sec)
> 
> 
> mysql> select * from billing.billing_mappings m, billing.billing_profiles p
>     -> where p.id = m.billing_profile_id and m.contract_id = 22;
> +----+------------+----------+--------------------+-------------+------------+----+-------------+--------------+--------------+---------+-----------------+--------------------+--------------------+---------------+----------------+----------------------+---------------------+-------------------------------+-------------------+------------------+--------------------+----------+----------+--------------+
> | id | start_date | end_date | billing_profile_id | contract_id |
> product_id | id | reseller_id | handle       | name         | prepaid |
> interval_charge | interval_free_time | interval_free_cash |
> interval_unit | interval_count | fraud_interval_limit |
> fraud_interval_lock | fraud_interval_notify         | fraud_daily_limit
> | fraud_daily_lock | fraud_daily_notify | currency | vat_rate |
> vat_included |
> +----+------------+----------+--------------------+-------------+------------+----+-------------+--------------+--------------+---------+-----------------+--------------------+--------------------+---------------+----------------+----------------------+---------------------+-------------------------------+-------------------+------------------+--------------------+----------+----------+--------------+
> | 22 | NULL       | NULL     |                  4 |          22 |      
> NULL |  4 |           1 | Tarifa_plana | Tarifa plana |       0 |      
>         0 |                  0 |                  0 | month         |  
>            1 |                  100 |                   3 |            
>                   |               100 |                3 | NULL        
>       | ??        |        0 |            0 |
> +----+------------+----------+--------------------+-------------+------------+----+-------------+--------------+--------------+---------+-----------------+--------------------+--------------------+---------------+----------------+----------------------+---------------------+-------------------------------+-------------------+------------------+--------------------+----------+----------+--------------+
> 1 row in set (0.00 sec)
> 
> 
> Thank you very much!!!
> 
> 
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 900 bytes
Desc: OpenPGP digital signature
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130321/d2ec5288/attachment.asc>

From nh.developer at gmail.com  Thu Mar 21 17:41:48 2013
From: nh.developer at gmail.com (Developing Tomorrow)
Date: Thu, 21 Mar 2013 22:41:48 +0100
Subject: [Spce-user] Running Sipwise on Centos
Message-ID: <004801ce267c$e4ecd6d0$aec68470$@gmail.com>

HI Jon.
[] 
Thank you for that.
It was really deep.

I guess I will have to set for Debian or .......

Thank for the answer




From alesg at coit.es  Thu Mar 21 18:14:44 2013
From: alesg at coit.es (Alejandro Sanchez Gimenez)
Date: Thu, 21 Mar 2013 23:14:44 +0100 (CET)
Subject: [Spce-user]  Fwd: ngcp-fraud-auto-lock
In-Reply-To: <40c228be-4d1d-4f0a-b91f-677f44f52e55@mail.coit.es>
Message-ID: <4d3c7dda-2db9-450f-bd1f-3fb71ff4647e@mail.coit.es>

Hi again Richard!! 


I?ve checked accounting.cdr table and source_account_id field of the contains 22, and the customer_cost field contains the cost. Here you can see the query with relevant data: 



mysql> select * from cdr where source_account_id like "22"\G; 
*************************** 1. row *************************** 
id: 52 
update_time: 2013-03-21 15:03:32 
source_user_id: 635a08f3-3df3-4559-a6fd-d5e280d9761e 
source_provider_id: 1 
source_external_subscriber_id: XXXXXXXX 
source_external_contract_id: 
source_account_id: 22 
source_user: XXXXXXXXX 
source_domain: xxx.xxx.es 
source_cli: 34XXXXXXXXX 
source_clir: 0 
source_ip: 192.168.0.24 
destination_user_id: 43317f99-0859-4e89-a63b-ddf3d9bffb20 
destination_provider_id: 1 
destination_external_subscriber_id: XXXXXXXXX 
destination_external_contract_id: XXXXX 
destination_account_id: 8 
destination_user: XXXXXXXXX 
destination_domain: xxx.xxx.es 
destination_user_dialed: XXXXXXXXX 
destination_user_in: XXXXXXXXX 
destination_domain_in: xxx.xxx.es 
peer_auth_user: 
peer_auth_realm: 
call_type: call 
call_status: ok 
call_code: 200 
init_time: 1363874581.577 
start_time: 1363874584.840 
duration: 15.680 
call_id: OWIzN2RmY2M5M2RlMmFmMWU1MzVhNzkyNTVkNzI2YmM. 
carrier_cost: 0.00 
reseller_cost: 0.00 
customer_cost: 16.50 
carrier_free_time: 0 
reseller_free_time: 0 
customer_free_time: 0 
carrier_billing_fee_id: NULL 
reseller_billing_fee_id: NULL 
customer_billing_fee_id: 7 
carrier_billing_zone_id: NULL 
reseller_billing_zone_id: NULL 
customer_billing_zone_id: 7 
frag_carrier_onpeak: NULL 
frag_reseller_onpeak: NULL 
frag_customer_onpeak: NULL 
is_fragmented: NULL 
split: 0 
rated_at: 2013-03-21 15:03:32 
rating_status: ok 
exported_at: NULL 
export_status: unexported 


Jon, I paste the query with \G: 



mysql> select * from contracts where id = 22\G; 
*************************** 1. row *************************** 
id: 22 
customer_id: NULL 
reseller_id: 1 
contact_id: NULL 
order_id: NULL 
status: active 
external_id: 
modify_timestamp: 2013-03-21 15:02:14 
create_timestamp: 2013-03-21 15:02:14 
activate_timestamp: 2013-03-21 15:02:14 
terminate_timestamp: NULL 
1 row in set (0.00 sec) 


mysql> select * from contract_balances where contract_id = 22\G; 
*************************** 1. row *************************** 
id: 21 
contract_id: 22 
cash_balance: 0 
cash_balance_interval: 0 
free_time_balance: 0 
free_time_balance_interval: 0 
start: 2013-03-01 00:00:00 
end: 2013-03-31 23:59:59 
invoice_id: NULL 
1 row in set (0.00 sec) 


mysql> select * from billing.billing_mappings m, billing.billing_profiles p 
-> where p.id = m.billing_profile_id and m.contract_id = 22\G; 
*************************** 1. row *************************** 
id: 22 
start_date: NULL 
end_date: NULL 
billing_profile_id: 4 
contract_id: 22 
product_id: NULL 
id: 4 
reseller_id: 1 
handle: Tarifa_plana 
name: Tarifa plana 
prepaid: 0 
interval_charge: 0 
interval_free_time: 0 
interval_free_cash: 0 
interval_unit: month 
interval_count: 1 
fraud_interval_limit: 100 
fraud_interval_lock: 3 
fraud_interval_notify: 
fraud_daily_limit: 100 
fraud_daily_lock: 3 
fraud_daily_notify: NULL 
currency: ?? 
vat_rate: 0 
vat_included: 0 
1 row in set (0.00 sec) 


Thank you very much!! 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130321/df7f1dc7/attachment.html>

From jbonilla at sipwise.com  Thu Mar 21 17:58:07 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Thu, 21 Mar 2013 22:58:07 +0100
Subject: [Spce-user] Running Sipwise on Centos
In-Reply-To: <004801ce267c$e4ecd6d0$aec68470$@gmail.com>
References: <004801ce267c$e4ecd6d0$aec68470$@gmail.com>
Message-ID: <20130321225807.15a56dcb@quenya>

El Thu, 21 Mar 2013 22:41:48 +0100
"Developing Tomorrow" <nh.developer at gmail.com> escribi?:

> HI Jon.
> [] 
> Thank you for that.
> It was really deep.
> 
> I guess I will have to set for Debian or .......
> 
> Thank for the answer
> 
>

did you use the "answer" button to reply my message? It appears as a new thread
to me.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130321/571067a1/attachment.asc>

From jbonilla at sipwise.com  Thu Mar 21 17:58:33 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Thu, 21 Mar 2013 22:58:33 +0100
Subject: [Spce-user] Fwd: ngcp-fraud-auto-lock
In-Reply-To: <4d3c7dda-2db9-450f-bd1f-3fb71ff4647e@mail.coit.es>
References: <40c228be-4d1d-4f0a-b91f-677f44f52e55@mail.coit.es>
	<4d3c7dda-2db9-450f-bd1f-3fb71ff4647e@mail.coit.es>
Message-ID: <20130321225833.342c05c1@quenya>

El Thu, 21 Mar 2013 23:14:44 +0100 (CET)
Alejandro Sanchez Gimenez <alesg at coit.es> escribi?:

> Hi again Richard!! 
> 
> 
> I?ve checked accounting.cdr table and source_account_id field of the contains
> 22, and the customer_cost field contains the cost. Here you can see the query
> with relevant data: 
> 
> 

did you use the "answer" button to reply the last message? It appears as a new
thread to me.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130321/746d2f7f/attachment.asc>

From alesg at coit.es  Thu Mar 21 18:24:04 2013
From: alesg at coit.es (Alejandro Sanchez Gimenez)
Date: Thu, 21 Mar 2013 23:24:04 +0100 (CET)
Subject: [Spce-user] Fwd: ngcp-fraud-auto-lock
In-Reply-To: <20130321225833.342c05c1@quenya>
Message-ID: <b812af0c-3ced-41cf-9e33-a8f4a293b96f@mail.coit.es>

Sorry Jon, I didn?t.

----- Mensaje original -----
De: "Jon Bonilla" <jbonilla at sipwise.com>
Para: spce-user at lists.sipwise.com
Enviados: Jueves, 21 de Marzo 2013 22:58:33
Asunto: Re: [Spce-user] Fwd: ngcp-fraud-auto-lock

El Thu, 21 Mar 2013 23:14:44 +0100 (CET)
Alejandro Sanchez Gimenez <alesg at coit.es> escribi?:

> Hi again Richard!! 
> 
> 
> I?ve checked accounting.cdr table and source_account_id field of the contains
> 22, and the customer_cost field contains the cost. Here you can see the query
> with relevant data: 
> 
> 

did you use the "answer" button to reply the last message? It appears as a new
thread to me.


_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com
http://lists.sipwise.com/listinfo/spce-user



From rea1ity at kaist.ac.kr  Fri Mar 22 00:55:39 2013
From: rea1ity at kaist.ac.kr (=?UTF-8?B?RGF2aWQgSG8=?=)
Date: Fri, 22 Mar 2013 13:55:39 +0900 (KST)
Subject: [Spce-user]
	=?utf-8?q?Please_let_meknowwhyIfaced401/403=28Unautho?=
	=?utf-8?q?rized/Invalid_To-URI_Detected=29=2E?=
Message-ID: <514be7813fe6_@_imoxion.com>

Jon,
 
I was succeed to register with my PSTN gateway last night.
The reason I couldn't register to my PSTN gateway was have no peer about incorrectly ID/PW because my PSTN gateway ID only gets number(i.e. 0 to 9).
And my PSTN GW required three kinds of values it is 'Represent telephone number', 'ID','PW'.
So I'm put in the values to there that I set on SIPWISE peer same as ID and PW.
But the one of value as 'Represent telephone number' used at ID on SIPWISE.(It caused confused me.)
So when I tried input a same value as 'Represent TEL No' and 'ID', The PSTN GW as well registered on SIPWISE server. (hooray!!)
It so hard time for me because its a first time to set VoIP system it configured SIP server and registering with third party PSTN GW.
But It is alright. thank you for your support.
Registered problem was cleared.
 
And it is another one of the log capacities.
As you mentioned the log files capacity can increased over the several GB.
If I want to modified the log files capacity has limited as several MB, how can I do that?
I didn't want to it would be problemed that SIP server hard drive capacity was full of log file so it occurred system error.
 
David
 


-----------------------????-----------------------
???? : "Jon Bonilla (Manwe)" <jbonilla at sipwise.com>
????: "spce-user at lists.sipwise.com" <spce-user at lists.sipwise.com>
? ?: David Ho <rea1ity at kaist.ac.kr>
????: 2013-03-21 19:24:36 GMT +0900 (Asia/Seoul)
??: Re: [Spce-user] Please let meknowwhyIfaced401/403(Unauthorized/Invalid To-URI Detected).

El Thu, 21 Mar 2013 12:24:34 +0900 (KST)

David Ho<rea1ity at kaist.ac.kr> escribi?:



>  

>  

> Jon,

>  

> I've already followed up the document that you provide "The sip:provider CE

> Handbook 2.7" so I'm followed setup 1 to 6 chapter and almost setting same

> like the document. And now I'm tried to modify values of Peering Servers

> several times and you could know where I'm modified the values at "6.4.4.

> Authenticating and Registering against Peering Servers". the URL following;

> http://www.sipwise.com/doc/2.7/spce/ar01s06.html#_authenticating_and_registering_against_peering_servers

> So manytimes, SIPWISE server denied my PSTN gateway caused by

> Unauthorized(401). And I found what is problemed via sems log

> file("/var/log/ngcp/sem/sems.log") it appeared error message before I

> mentioned same like following;





So you already created two subscribers and you ca call from one to the other?



If so, send us what you did to register against your peer server and all the

related information, network capture and configuration files. We cannot guess

what's going on without information.





> -----------------------------------------------------------------------------------------------------------------

> Mar 20 08:44:14 spce sems[2164]: [#7f601e77d700] [doRegistration,

> AmSipRegistration.cpp:139] ERROR: failed to send registration.

> -----------------------------------------------------------------------------------------------------------------

> So 'doRegistration function in AmSipRegistration.cpp file occurred ERROR

> message. If you were me, how could solve the error occurred? And it is

> another one that I afraid thing. the log files in SIPWISE as kamailio-lb.log

> and kamailio-options-proxy.log increased capacity too much. my

> kamailio-lb.log file capacity is over than 28MB and

> kamailio-options-proxy.log file capacity is more than 7MB. The reason of the

> log too much increased capacity caused by stacked approax. four to five log

> messages per a second! Is it alright to leave it even though it will be huge

> capacity log file after than this log file could occurr system error for leak

> of hard disk capacity. David





You can expect several GB of logs per day in a production system. 28MB is a

very small log. THe spce logs a lot of information for each request.







-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130322/06fa8722/attachment.html>

From agranig at sipwise.com  Fri Mar 22 03:26:05 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Fri, 22 Mar 2013 08:26:05 +0100
Subject: [Spce-user] Please let
 meknowwhyIfaced401/403(Unauthorized/Invalid To-URI Detected).
In-Reply-To: <514be7813fe6_@_imoxion.com>
References: <514be7813fe6_@_imoxion.com>
Message-ID: <514C078D.80102@sipwise.com>

Hi,

On 03/22/2013 05:55 AM, David Ho wrote:
> And it is another one of the log capacities.
>
> As you mentioned the log files capacity can increased over the several GB.
>
> If I want to modified the log files capacity has limited as several MB,
> how can I do that?
>
> I didn't want to it would be problemed that SIP server hard drive
> capacity was full of log file so it occurred system error.

It's true that the log files (especially kamailio-proxy.log and 
kamailio-lb.log) are pretty extensive, and can increase over several 
gigs per day. However, you'll appreciate that when you run into a 
customer issue for the first time, where you'll be able to clearly 
comprehend what was going on. And it's not like the SPCE is a tiny 
system, so be prepared for quite some data (e.g. you get CDRs for each 
call, which will be kept for 3 months by default, then backed up to 
/var/backups/cdr for 12 months by default etc.

Also note that log files get rotated and compressed nightly, and are 
kept for however long you choose to (3 months by default). You can 
fine-tune that by copying 
/etc/ngcp-config/templates/etc/logrotate.d/ngcp.tt2 to 
/etc/ngcp-config/templates/etc/logrotate.d/ngcp.customtt.tt2, adapt it 
to your needs and execute "ngcpcfg apply" afterwards.

That's the preferred way to keep your used disk space low, instead of 
starting to reduce log messages.

Andreas



From alesg at coit.es  Fri Mar 22 09:14:48 2013
From: alesg at coit.es (Alejandro Sanchez Gimenez)
Date: Fri, 22 Mar 2013 14:14:48 +0100 (CET)
Subject: [Spce-user] Fwd: ngcp-fraud-auto-lock
In-Reply-To: <b812af0c-3ced-41cf-9e33-a8f4a293b96f@mail.coit.es>
Message-ID: <c12b2fa4-a0a6-4c42-bdff-24d2bce3a7fb@mail.coit.es>

I think that the problem reported by Mario Contreras On 07/16/12 is still happening: "I have tested, and I can asume it works like I supposed. The fraud only compares de free cash you have assigned to the account...". As he said, if I assign free cash to the account, fraud lock works properly. If I don?t assign free cash account balance doesn?t increase, so the comparison in /usr/sbin/ngcp-fraud-auto-lock: "b.cash_balance_interval >= p.fraud_interval_limit" never happens. It was supposed to be fixed in 2.6. Is this correct?

Thank you very much!!

----- Mensaje original -----
De: "Alejandro Sanchez Gimenez" <alesg at coit.es>
Para: "Jon Bonilla" <jbonilla at sipwise.com>
CC: spce-user at lists.sipwise.com
Enviados: Jueves, 21 de Marzo 2013 23:24:04
Asunto: Re: [Spce-user] Fwd: ngcp-fraud-auto-lock

Sorry Jon, I didn?t.

----- Mensaje original -----
De: "Jon Bonilla" <jbonilla at sipwise.com>
Para: spce-user at lists.sipwise.com
Enviados: Jueves, 21 de Marzo 2013 22:58:33
Asunto: Re: [Spce-user] Fwd: ngcp-fraud-auto-lock

El Thu, 21 Mar 2013 23:14:44 +0100 (CET)
Alejandro Sanchez Gimenez <alesg at coit.es> escribi?:

> Hi again Richard!! 
> 
> 
> I?ve checked accounting.cdr table and source_account_id field of the contains
> 22, and the customer_cost field contains the cost. Here you can see the query
> with relevant data: 
> 
> 

did you use the "answer" button to reply the last message? It appears as a new
thread to me.


_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com
http://lists.sipwise.com/listinfo/spce-user

_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com
http://lists.sipwise.com/listinfo/spce-user



From vermeulen.deon at gmail.com  Fri Mar 22 09:16:19 2013
From: vermeulen.deon at gmail.com (Deon Vermeulen)
Date: Fri, 22 Mar 2013 15:16:19 +0200
Subject: [Spce-user] Invoicing and Billing Peers/Trunks
Message-ID: <514C59A3.7090101@gmail.com>

Good Day

Is there any possibility that one could generate an Invoice to customers 
with a Trunk to SPCE 2.7?

Is there any possibility to Generate a Bill to a Peering Partner for 
calls sent to SPCE 2.7. i.e Reverse billing?


Thank you.

KR
Deon
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130322/d84e9e77/attachment.html>

From rfuchs at sipwise.com  Fri Mar 22 09:27:54 2013
From: rfuchs at sipwise.com (Richard Fuchs)
Date: Fri, 22 Mar 2013 09:27:54 -0400
Subject: [Spce-user] Fwd: ngcp-fraud-auto-lock
In-Reply-To: <c12b2fa4-a0a6-4c42-bdff-24d2bce3a7fb@mail.coit.es>
References: <c12b2fa4-a0a6-4c42-bdff-24d2bce3a7fb@mail.coit.es>
Message-ID: <514C5C5A.4090800@sipwise.com>

Can you check which version of rate-o-mat is installed?

dpkg -l ngcp-rate-o-mat


On 03/22/13 09:14, Alejandro Sanchez Gimenez wrote:
> I think that the problem reported by Mario Contreras On 07/16/12 is still happening: "I have tested, and I can asume it works like I supposed. The fraud only compares de free cash you have assigned to the account...". As he said, if I assign free cash to the account, fraud lock works properly. If I don?t assign free cash account balance doesn?t increase, so the comparison in /usr/sbin/ngcp-fraud-auto-lock: "b.cash_balance_interval >= p.fraud_interval_limit" never happens. It was supposed to be fixed in 2.6. Is this correct?
> 
> Thank you very much!!
> 
> ----- Mensaje original -----
> De: "Alejandro Sanchez Gimenez" <alesg at coit.es>
> Para: "Jon Bonilla" <jbonilla at sipwise.com>
> CC: spce-user at lists.sipwise.com
> Enviados: Jueves, 21 de Marzo 2013 23:24:04
> Asunto: Re: [Spce-user] Fwd: ngcp-fraud-auto-lock
> 
> Sorry Jon, I didn?t.
> 
> ----- Mensaje original -----
> De: "Jon Bonilla" <jbonilla at sipwise.com>
> Para: spce-user at lists.sipwise.com
> Enviados: Jueves, 21 de Marzo 2013 22:58:33
> Asunto: Re: [Spce-user] Fwd: ngcp-fraud-auto-lock
> 
> El Thu, 21 Mar 2013 23:14:44 +0100 (CET)
> Alejandro Sanchez Gimenez <alesg at coit.es> escribi?:
> 
>> Hi again Richard!! 
>>
>>
>> I?ve checked accounting.cdr table and source_account_id field of the contains
>> 22, and the customer_cost field contains the cost. Here you can see the query
>> with relevant data: 
>>
>>
> 
> did you use the "answer" button to reply the last message? It appears as a new
> thread to me.
> 
> 
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
> 
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
> 
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 900 bytes
Desc: OpenPGP digital signature
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130322/fa313295/attachment.asc>

From alesg at coit.es  Fri Mar 22 09:56:53 2013
From: alesg at coit.es (Alejandro Sanchez Gimenez)
Date: Fri, 22 Mar 2013 14:56:53 +0100 (CET)
Subject: [Spce-user] Fwd: ngcp-fraud-auto-lock
In-Reply-To: <514C5C5A.4090800@sipwise.com>
Message-ID: <1c03da20-53f6-4e8b-8bc0-b5927529c899@mail.coit.es>

Hello Richard,

The version installed is 1.4.0

Thanks

----- Mensaje original -----
De: "Richard Fuchs" <rfuchs at sipwise.com>
Para: spce-user at lists.sipwise.com
Enviados: Viernes, 22 de Marzo 2013 14:27:54
Asunto: Re: [Spce-user] Fwd: ngcp-fraud-auto-lock

Can you check which version of rate-o-mat is installed?

dpkg -l ngcp-rate-o-mat


On 03/22/13 09:14, Alejandro Sanchez Gimenez wrote:
> I think that the problem reported by Mario Contreras On 07/16/12 is still happening: "I have tested, and I can asume it works like I supposed. The fraud only compares de free cash you have assigned to the account...". As he said, if I assign free cash to the account, fraud lock works properly. If I don?t assign free cash account balance doesn?t increase, so the comparison in /usr/sbin/ngcp-fraud-auto-lock: "b.cash_balance_interval >= p.fraud_interval_limit" never happens. It was supposed to be fixed in 2.6. Is this correct?
> 
> Thank you very much!!
> 
> ----- Mensaje original -----
> De: "Alejandro Sanchez Gimenez" <alesg at coit.es>
> Para: "Jon Bonilla" <jbonilla at sipwise.com>
> CC: spce-user at lists.sipwise.com
> Enviados: Jueves, 21 de Marzo 2013 23:24:04
> Asunto: Re: [Spce-user] Fwd: ngcp-fraud-auto-lock
> 
> Sorry Jon, I didn?t.
> 
> ----- Mensaje original -----
> De: "Jon Bonilla" <jbonilla at sipwise.com>
> Para: spce-user at lists.sipwise.com
> Enviados: Jueves, 21 de Marzo 2013 22:58:33
> Asunto: Re: [Spce-user] Fwd: ngcp-fraud-auto-lock
> 
> El Thu, 21 Mar 2013 23:14:44 +0100 (CET)
> Alejandro Sanchez Gimenez <alesg at coit.es> escribi?:
> 
>> Hi again Richard!! 
>>
>>
>> I?ve checked accounting.cdr table and source_account_id field of the contains
>> 22, and the customer_cost field contains the cost. Here you can see the query
>> with relevant data: 
>>
>>
> 
> did you use the "answer" button to reply the last message? It appears as a new
> thread to me.
> 
> 
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
> 
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
> 
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user


_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com
http://lists.sipwise.com/listinfo/spce-user



From rfuchs at sipwise.com  Fri Mar 22 10:30:06 2013
From: rfuchs at sipwise.com (Richard Fuchs)
Date: Fri, 22 Mar 2013 10:30:06 -0400
Subject: [Spce-user] Fwd: ngcp-fraud-auto-lock
In-Reply-To: <1c03da20-53f6-4e8b-8bc0-b5927529c899@mail.coit.es>
References: <1c03da20-53f6-4e8b-8bc0-b5927529c899@mail.coit.es>
Message-ID: <514C6AEE.7050705@sipwise.com>

Hey,

Turns out that this was fixed last July, but then broken again in
October. We'll have a (new) fix for it shortly.

cheers


On 03/22/13 09:56, Alejandro Sanchez Gimenez wrote:
> Hello Richard,
> 
> The version installed is 1.4.0
> 
> Thanks
> 
> ----- Mensaje original -----
> De: "Richard Fuchs" <rfuchs at sipwise.com>
> Para: spce-user at lists.sipwise.com
> Enviados: Viernes, 22 de Marzo 2013 14:27:54
> Asunto: Re: [Spce-user] Fwd: ngcp-fraud-auto-lock
> 
> Can you check which version of rate-o-mat is installed?
> 
> dpkg -l ngcp-rate-o-mat
> 
> 
> On 03/22/13 09:14, Alejandro Sanchez Gimenez wrote:
>> I think that the problem reported by Mario Contreras On 07/16/12 is still happening: "I have tested, and I can asume it works like I supposed. The fraud only compares de free cash you have assigned to the account...". As he said, if I assign free cash to the account, fraud lock works properly. If I don?t assign free cash account balance doesn?t increase, so the comparison in /usr/sbin/ngcp-fraud-auto-lock: "b.cash_balance_interval >= p.fraud_interval_limit" never happens. It was supposed to be fixed in 2.6. Is this correct?
>>
>> Thank you very much!!
>>
>> ----- Mensaje original -----
>> De: "Alejandro Sanchez Gimenez" <alesg at coit.es>
>> Para: "Jon Bonilla" <jbonilla at sipwise.com>
>> CC: spce-user at lists.sipwise.com
>> Enviados: Jueves, 21 de Marzo 2013 23:24:04
>> Asunto: Re: [Spce-user] Fwd: ngcp-fraud-auto-lock
>>
>> Sorry Jon, I didn?t.
>>
>> ----- Mensaje original -----
>> De: "Jon Bonilla" <jbonilla at sipwise.com>
>> Para: spce-user at lists.sipwise.com
>> Enviados: Jueves, 21 de Marzo 2013 22:58:33
>> Asunto: Re: [Spce-user] Fwd: ngcp-fraud-auto-lock
>>
>> El Thu, 21 Mar 2013 23:14:44 +0100 (CET)
>> Alejandro Sanchez Gimenez <alesg at coit.es> escribi?:
>>
>>> Hi again Richard!! 
>>>
>>>
>>> I?ve checked accounting.cdr table and source_account_id field of the contains
>>> 22, and the customer_cost field contains the cost. Here you can see the query
>>> with relevant data: 
>>>
>>>
>>
>> did you use the "answer" button to reply the last message? It appears as a new
>> thread to me.
>>
>>
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/listinfo/spce-user
>>
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/listinfo/spce-user
>>
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/listinfo/spce-user
> 
> 
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 900 bytes
Desc: OpenPGP digital signature
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130322/97815cab/attachment.asc>

From gavin.sweet at skyracktelecom.com  Fri Mar 22 12:00:26 2013
From: gavin.sweet at skyracktelecom.com (Gavin Sweet)
Date: Fri, 22 Mar 2013 16:00:26 -0000
Subject: [Spce-user] Inbound CDR tariffs?
Message-ID: <113701ce2716$61b99880$252cc980$@sweet@skyracktelecom.com>

Hi - 

We have inbound CDR exporting switched on, and I had a couple of questions,
as we'd ideally like to be able to actually rate (apply tariff with cost) to
certain inbound calls, because they carry a cost to us (we have a mobile
network integrated for which there are transit charges in both directions).

1. It actually appears that all inbound calls are classified with
destinations and rates that are not in the billing profile fees? Can this be
modified?

E.g. where a call outbound to a specific mobile picks up the appropriate
destinations from billing fees:
carrier_zone	customer_zone	carrier_destination	customer_destination
UKN			UKN			UKNFM1
UKNFM1

However, calls *inbound* from that same mobile to the same user show:
carrier_zone	customer_zone	carrier_destination	customer_destination
onnet						platform internal	


2. Is source_account_id = 0 fully reliable as an indicator in a CDR file
that the call was from an external user? does anything break that
assumption?


Thanks as ever.

Gavin




From alesg at coit.es  Fri Mar 22 12:36:17 2013
From: alesg at coit.es (Alejandro Sanchez Gimenez)
Date: Fri, 22 Mar 2013 17:36:17 +0100 (CET)
Subject: [Spce-user] Fwd: ngcp-fraud-auto-lock
In-Reply-To: <514C6AEE.7050705@sipwise.com>
Message-ID: <b48c9ac9-ded7-4130-af92-b0106adb36a1@mail.coit.es>

Ok Richard, thank you very much.

Will the fix be available in a new version of sipwise:ce or via patch?

Regards



----- Mensaje original -----
De: Richard Fuchs <rfuchs at sipwise.com>
Para: spce-user at lists.sipwise.com
Enviado: Fri, 22 Mar 2013 15:30:06 +0100 (CET)
Asunto: Re: [Spce-user] Fwd: ngcp-fraud-auto-lock

Hey,

Turns out that this was fixed last July, but then broken again in
October. We'll have a (new) fix for it shortly.

cheers


On 03/22/13 09:56, Alejandro Sanchez Gimenez wrote:
> Hello Richard,
> 
> The version installed is 1.4.0
> 
> Thanks
> 
> ----- Mensaje original -----
> De: "Richard Fuchs" <rfuchs at sipwise.com>
> Para: spce-user at lists.sipwise.com
> Enviados: Viernes, 22 de Marzo 2013 14:27:54
> Asunto: Re: [Spce-user] Fwd: ngcp-fraud-auto-lock
> 
> Can you check which version of rate-o-mat is installed?
> 
> dpkg -l ngcp-rate-o-mat
> 
> 
> On 03/22/13 09:14, Alejandro Sanchez Gimenez wrote:
>> I think that the problem reported by Mario Contreras On 07/16/12 is still happening: "I have tested, and I can asume it works like I supposed. The fraud only compares de free cash you have assigned to the account...". As he said, if I assign free cash to the account, fraud lock works properly. If I don?t assign free cash account balance doesn?t increase, so the comparison in /usr/sbin/ngcp-fraud-auto-lock: "b.cash_balance_interval >= p.fraud_interval_limit" never happens. It was supposed to be fixed in 2.6. Is this correct?
>>
>> Thank you very much!!
>>
>> ----- Mensaje original -----
>> De: "Alejandro Sanchez Gimenez" <alesg at coit.es>
>> Para: "Jon Bonilla" <jbonilla at sipwise.com>
>> CC: spce-user at lists.sipwise.com
>> Enviados: Jueves, 21 de Marzo 2013 23:24:04
>> Asunto: Re: [Spce-user] Fwd: ngcp-fraud-auto-lock
>>
>> Sorry Jon, I didn?t.
>>
>> ----- Mensaje original -----
>> De: "Jon Bonilla" <jbonilla at sipwise.com>
>> Para: spce-user at lists.sipwise.com
>> Enviados: Jueves, 21 de Marzo 2013 22:58:33
>> Asunto: Re: [Spce-user] Fwd: ngcp-fraud-auto-lock
>>
>> El Thu, 21 Mar 2013 23:14:44 +0100 (CET)
>> Alejandro Sanchez Gimenez <alesg at coit.es> escribi?:
>>
>>> Hi again Richard!! 
>>>
>>>
>>> I?ve checked accounting.cdr table and source_account_id field of the contains
>>> 22, and the customer_cost field contains the cost. Here you can see the query
>>> with relevant data: 
>>>
>>>
>>
>> did you use the "answer" button to reply the last message? It appears as a new
>> thread to me.
>>
>>
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/listinfo/spce-user
>>
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/listinfo/spce-user
>>
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/listinfo/spce-user
> 
> 
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user





From agranig at sipwise.com  Fri Mar 22 12:44:45 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Fri, 22 Mar 2013 17:44:45 +0100
Subject: [Spce-user] Inbound CDR tariffs?
In-Reply-To: <113701ce2716$61b99880$252cc980$@sweet@skyracktelecom.com>
References: <113701ce2716$61b99880$252cc980$@sweet@skyracktelecom.com>
Message-ID: <514C8A7D.2080009@sipwise.com>

Hi,

On 03/22/2013 05:00 PM, Gavin Sweet wrote:
> 1. It actually appears that all inbound calls are classified with
> destinations and rates that are not in the billing profile fees? Can this be
> modified?

Inbound calls are not rated at all in 2.7. You've to wait for another 
week or so for 2.8 to come out :)


> 2. Is source_account_id = 0 fully reliable as an indicator in a CDR file
> that the call was from an external user? does anything break that
> assumption?

Should do the trick, but I'd go for source_user_id instead, which is 
more obvious.

Andreas



From gavin.sweet at skyracktelecom.com  Fri Mar 22 13:13:57 2013
From: gavin.sweet at skyracktelecom.com (Gavin Sweet)
Date: Fri, 22 Mar 2013 17:13:57 -0000
Subject: [Spce-user] Inbound CDR tariffs?
In-Reply-To: <514C8A7D.2080009@sipwise.com>
References: <113701ce2716$61b99880$252cc980$@sweet@skyracktelecom.com>
	<514C8A7D.2080009@sipwise.com>
Message-ID: <114a01ce2720$a3ba5ee0$eb2f1ca0$@sweet@skyracktelecom.com>

Thanks Andreas, look forward to that.
As soon as you have the new feature/fix list, would be great to see it.

Cheers,
Gavin

> -----Original Message-----
> From: spce-user-bounces at lists.sipwise.com [mailto:spce-user-
> bounces at lists.sipwise.com] On Behalf Of Andreas Granig
> Sent: 22 March 2013 16:45
> To: spce-user at lists.sipwise.com
> Subject: Re: [Spce-user] Inbound CDR tariffs?
> 
> Hi,
> 
> On 03/22/2013 05:00 PM, Gavin Sweet wrote:
> > 1. It actually appears that all inbound calls are classified with
> > destinations and rates that are not in the billing profile fees? Can
> > this be modified?
> 
> Inbound calls are not rated at all in 2.7. You've to wait for another
> week or so for 2.8 to come out :)
> 
> 
> > 2. Is source_account_id = 0 fully reliable as an indicator in a CDR
> file
> > that the call was from an external user? does anything break that
> > assumption?
> 
> Should do the trick, but I'd go for source_user_id instead, which is
> more obvious.
> 
> Andreas
> 
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
> -----
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2013.0.2904 / Virus Database: 2641/6166 - Release Date:
> 03/12/13
> Internal Virus Database is out of date.




From wfulton at thirdhatch.com  Fri Mar 22 19:27:39 2013
From: wfulton at thirdhatch.com (William Fulton)
Date: Fri, 22 Mar 2013 16:27:39 -0700
Subject: [Spce-user] Sipwise CE 2.7 SBC config revisited
Message-ID: <0F449623D8AAB145A3278882C750BA5831C713@sea2k3ex06.thirdhatch.corp>

Guys,

 

I have spent a couple of weeks trying to make this work with absolutely
no luck.  I have read the manual cover to cover and there is clearly a
section missing.  Section 6 references a section in Section 7 about
advanced network configuration that does not exist.

 

I have two interfaces:

 

eth0 and eth1

 

*         eth0 is on the external subnet and this configuration was easy
as I simply assigned the sip_ext, web_ext, ssh_ext, etc to this adaptor.

*         eth1 is on the internal subnet and I have added the role of
sip_int to this adaptor.  

*         I created a new subscriber and set the preferences to
authenticate to the peer and set the peer username, password and realm

*         Despite my best efforts, I cannot get the subscriber external
to the Sipwise CE system to authenticate to the Asterisk system internal
to the Sipwise CE system.  

 

What am I missing?  I'm sure it's pretty simple but I can't see it.

 

Thanks,

William

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130322/743fedd6/attachment.html>

From jbonilla at sipwise.com  Sat Mar 23 06:13:16 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Sat, 23 Mar 2013 11:13:16 +0100
Subject: [Spce-user] Sipwise CE 2.7 SBC config revisited
In-Reply-To: <0F449623D8AAB145A3278882C750BA5831C713@sea2k3ex06.thirdhatch.corp>
References: <0F449623D8AAB145A3278882C750BA5831C713@sea2k3ex06.thirdhatch.corp>
Message-ID: <20130323111316.773fab5b@quenya>

El Fri, 22 Mar 2013 16:27:39 -0700
"William Fulton" <wfulton at thirdhatch.com> escribi?:

> Guys,
> 
>  
> 
> I have spent a couple of weeks trying to make this work with absolutely
> no luck.  I have read the manual cover to cover and there is clearly a
> section missing.  Section 6 references a section in Section 7 about
> advanced network configuration that does not exist.
> 
>  
> 
> I have two interfaces:
> 
>  
> 
> eth0 and eth1
> 
>  
> 
> *         eth0 is on the external subnet and this configuration was easy
> as I simply assigned the sip_ext, web_ext, ssh_ext, etc to this adaptor.
> 
> *         eth1 is on the internal subnet and I have added the role of
> sip_int to this adaptor.  


why did you do this? Why not lo interface for this role?

> 
> *         I created a new subscriber and set the preferences to
> authenticate to the peer and set the peer username, password and realm

Did you also check the "peer_auth_register" option?


> 
> *         Despite my best efforts, I cannot get the subscriber external
> to the Sipwise CE system to authenticate to the Asterisk system internal
> to the Sipwise CE system.  
> 

What? An external subscriber to authenticate against the internal asterisk? The
asterisk system running in the spce is just a voicebox endpoint.

Could you please explain what's you scenario and what you are trying to
achieve? Sorry but I didn't understand.

>  
> 
> What am I missing?  I'm sure it's pretty simple but I can't see it.
> 

Please tell us in more detail what your scenario is and what you did.


cheers,

Jon

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130323/42f36788/attachment.asc>

From am at sipalto.com  Sat Mar 23 13:32:11 2013
From: am at sipalto.com (Ahmed Murad)
Date: Sat, 23 Mar 2013 17:32:11 -0000
Subject: [Spce-user] routing calls between peers
Message-ID: <04db01ce27ec$5a3f5c30$0ebe1490$@sipalto.com>

Hello,

 

I am new to SIPWISE, do you have any guides on how to route calls from 1
peer to another peer via the SIPWISE box?

 

Any guidance would be greatly appreciated.

 

Best regards,

 

Ahmed

 

Ahmed Murad | SiPalto Ltd. 

Tel: 0207 148 7525  |  Fax: 0207 148  7526  |  Email:
<mailto:am at sipalto.com> am at sipalto.com
6-16 Arbutus St, London, E8 4DT | Website:  <http://www.sipalto.com/>
www.sipalto.com

 

Description: Description: Description: Description: Description:
Description: Description:
http://www.sipalto.com/images/large-logo.png

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130323/9132c9cd/attachment.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.png
Type: image/png
Size: 5743 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130323/9132c9cd/attachment.png>

From am at sipalto.com  Sat Mar 23 13:36:19 2013
From: am at sipalto.com (Ahmed Murad)
Date: Sat, 23 Mar 2013 17:36:19 -0000
Subject: [Spce-user] Connect softphone to SIPWISE
Message-ID: <04e101ce27ec$ee7b0d90$cb7128b0$@sipalto.com>

Hello,

 

I have been following your guides and successfully created & connected to a
couple of external peers.

 

However I am having trouble connecting a softphone to our first subscriber.
I have created the domain, account and subscriber.

 

But I get a timeout 500 error when I try to connect the softphone to the
subscriber, I have ensured there are no firewall rules causing the timeout.

 

Please could you point me in the right direction in regards to
troubleshooting, is there anything I've missed?

 

Best regards,

 

Ahmed

 

Ahmed Murad | SiPalto Ltd. 

Tel: 0207 148 7525  |  Fax: 0207 148  7526  |  Email:
<mailto:am at sipalto.com> am at sipalto.com
6-16 Arbutus St, London, E8 4DT | Website:  <http://www.sipalto.com/>
www.sipalto.com

 

Description: Description: Description: Description: Description:
Description: Description:
http://www.sipalto.com/images/large-logo.png

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130323/06e1a274/attachment.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.png
Type: image/png
Size: 5743 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130323/06e1a274/attachment.png>

From nh.developer at gmail.com  Sat Mar 23 15:03:01 2013
From: nh.developer at gmail.com (Developing Tomorrow)
Date: Sat, 23 Mar 2013 20:03:01 +0100
Subject: [Spce-user] Sipwise over openvpn
Message-ID: <00b701ce27f9$0b2de460$2189ad20$@gmail.com>

Hi.,

 

I have "successfully" installed sipwise and am able to register users and
make calls using any "normal" providers.

Now

In the same box, I have created a tunnel to another network that has a
gateway.

I am able to ping that gateway from the station that has sipwise also able
to send traffic to the gateway, but I will get no answer because of the ip
used.

Let me put down the layout:

All the following ips are fake but will clearly show the layout.

 

Sipwise external ip: 200.200.200.200

Sipwise internal ip: 10.0.0.1

 

Tunnel router: 10.0.0.10

Gateway across tunel: 192.168.10.10

 

If I use a standard asterisk box, give it ip 10.0.1 and send calls to
gateway, the voice will come back thru tunnel because the gateway is
replying to 10.0.0.1

 

In my sipwise case, it does not work as sipwise is sending the call with ip
200.200.200.200 so the gateway is answering thru the normal internet and not
the tunnel.

 

How can I configure sipwise to send info to that gateway using ip in the
form ttttttttt at 10.0.0.1 <mailto:ttttttttt at 10.0.0.1>  and not
ttttttttt at 200.200.200.200 <mailto:ttttttttt at 200.200.200.200> .

 

The media relay is supposed to do that function but I have no clue on how to
configure it.

I could only create a socket with ip 10.0.0.1 and that allows me to send the
traffic thru the vpn, but does not change the ip that the gateway receives.

 

I would like to receive a call like TTTTTTTTTTTTTTTTTT at yyy.yyy.yyy.yyy
<mailto:TTTTTTTTTTTTTTTTTT at yyy.yyy.yyy.yyy>  and send it to the gateway like
TTTTTTTTTTTTTT at 10.0.0.1 <mailto:TTTTTTTTTTTTTT at 10.0.0.1> ....

So the gateway would answer thru the vpn to 10.0.0.1 and sipwise would send
it back to yyy.yyy.yyy.yyy

 

Media proxy? Media relay?

 

How can I configure that.

Guidance would be mostly appreciated.

Thank you.

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130323/d89d1b7d/attachment.html>

From oliver at oliverv.com  Sat Mar 23 15:12:39 2013
From: oliver at oliverv.com (Oliver Vermeulen)
Date: Sat, 23 Mar 2013 20:12:39 +0100
Subject: [Spce-user] Why does it take 35sec to register
Message-ID: <011701ce27fa$63ef56a0$2bce03e0$@oliverv.com>

Hello All,

 

I tested sipwise yesterday.. install was successful very easy.

 

Does anybody have problems with sip registration ?

Why does it take 35sec to register.

 

Also when it's registered and I dial takes long time to process call.

 

Cheers,

 

Olivier Vermeulen

Description: oliverv.jpg



msn:  <mailto:oliver at oliverv.com> oliver at oliverv.com

 <http://www.wvg-tele.com/> http://www.oliverv.com

****************************************************************************
************ 
This E-mail message, including any attachments, is intended only for the
person or entity to which it is addressed, 
and may contain confidential Information. If you are not the intended
recipient, any review, retransmission, disclosure, copying, 
modification or other use of this E-mail message or attachments is strictly
forbidden. If you have received this E-mail message in error,
please contact the author and delete the message and any attachments from
your computer.
P Think before you print.

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130323/66b7ba3e/attachment.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image002.png
Type: application/octet-stream
Size: 21421 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130323/66b7ba3e/attachment.png>

From jbonilla at sipwise.com  Sat Mar 23 15:21:10 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Sat, 23 Mar 2013 20:21:10 +0100
Subject: [Spce-user] Why does it take 35sec to register
In-Reply-To: <011701ce27fa$63ef56a0$2bce03e0$@oliverv.com>
References: <011701ce27fa$63ef56a0$2bce03e0$@oliverv.com>
Message-ID: <20130323202110.6f71ae8d@quenya>

El Sat, 23 Mar 2013 20:12:39 +0100
"Oliver Vermeulen" <oliver at oliverv.com> escribi?:

> Hello All,
> 
>  
> 
> I tested sipwise yesterday.. install was successful very easy.
> 
>  
> 
> Does anybody have problems with sip registration ?
> 
> Why does it take 35sec to register.
> 
>  
> 
> Also when it's registered and I dial takes long time to process call.
> 
> 

Please make a ngrep-sip capture of a registration and another one of a call. We
will be able to see how long takes.


-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130323/afee13bc/attachment.asc>

From jbonilla at sipwise.com  Sat Mar 23 15:24:46 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Sat, 23 Mar 2013 20:24:46 +0100
Subject: [Spce-user] Sipwise over openvpn
In-Reply-To: <00b701ce27f9$0b2de460$2189ad20$@gmail.com>
References: <00b701ce27f9$0b2de460$2189ad20$@gmail.com>
Message-ID: <20130323202446.1bc5eeed@quenya>

El Sat, 23 Mar 2013 20:03:01 +0100
"Developing Tomorrow" <nh.developer at gmail.com> escribi?:

> Hi.,
> 
>  
> 
> I have "successfully" installed sipwise and am able to register users and
> make calls using any "normal" providers.
> 
> Now
> 
> In the same box, I have created a tunnel to another network that has a
> gateway.
> 
> I am able to ping that gateway from the station that has sipwise also able
> to send traffic to the gateway, but I will get no answer because of the ip
> used.
> 
> Let me put down the layout:
> 
> All the following ips are fake but will clearly show the layout.
> 
>  
> 
> Sipwise external ip: 200.200.200.200
> 
> Sipwise internal ip: 10.0.0.1
> 
>  
> 
> Tunnel router: 10.0.0.10
> 
> Gateway across tunel: 192.168.10.10
> 
>  
> 
> If I use a standard asterisk box, give it ip 10.0.1 and send calls to
> gateway, the voice will come back thru tunnel because the gateway is
> replying to 10.0.0.1
> 
>  
> 
> In my sipwise case, it does not work as sipwise is sending the call with ip
> 200.200.200.200 so the gateway is answering thru the normal internet and not
> the tunnel.
> 
>  
> 
> How can I configure sipwise to send info to that gateway using ip in the
> form ttttttttt at 10.0.0.1 <mailto:ttttttttt at 10.0.0.1>  and not
> ttttttttt at 200.200.200.200 <mailto:ttttttttt at 200.200.200.200> .
> 
>  

Why don't you configure in the other side of the tunnel. "Send the traffic to
20.200.200.200 via 10.0.0.1" ?

You can use the "extra_socket" option in the spce and that will work for you at
sip signaling level but at rtp level it won't work so you'd get signaling via
the tunel and media over internet. If that's what you want it's fine. 

Anyways I'd continue using the external main address and try to fix you issue
at networking level, with routes.


> 
> The media relay is supposed to do that function but I have no clue on how to
> configure it.
> 
> I could only create a socket with ip 10.0.0.1 and that allows me to send the
> traffic thru the vpn, but does not change the ip that the gateway receives.
> 



cheers,

Jon

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130323/d6223bd4/attachment.asc>

From jbonilla at sipwise.com  Sat Mar 23 15:26:05 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Sat, 23 Mar 2013 20:26:05 +0100
Subject: [Spce-user] Connect softphone to SIPWISE
In-Reply-To: <04e101ce27ec$ee7b0d90$cb7128b0$@sipalto.com>
References: <04e101ce27ec$ee7b0d90$cb7128b0$@sipalto.com>
Message-ID: <20130323202605.57ed0cdd@quenya>

El Sat, 23 Mar 2013 17:36:19 -0000
"Ahmed Murad" <am at sipalto.com> escribi?:

> Hello,
> 
>  
> 
> I have been following your guides and successfully created & connected to a
> couple of external peers.
> 
>  
> 
> However I am having trouble connecting a softphone to our first subscriber.
> I have created the domain, account and subscriber.
> 
>  
> 
> But I get a timeout 500 error when I try to connect the softphone to the
> subscriber, I have ensured there are no firewall rules causing the timeout.
> 
>  
> 
> Please could you point me in the right direction in regards to
> troubleshooting, is there anything I've missed?
> 
>  
> 

Please do a ngre-sip capture in the systeem and let us check the
incoming/outgoing traffic.

Also, you can check /var/log/ngcp/kamaiio-proxy.log to check if REGISTER
requests are arriving and what's going on with them.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130323/8d7f84f5/attachment.asc>

From jbonilla at sipwise.com  Sat Mar 23 15:40:18 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Sat, 23 Mar 2013 20:40:18 +0100
Subject: [Spce-user] routing calls between peers
In-Reply-To: <04db01ce27ec$5a3f5c30$0ebe1490$@sipalto.com>
References: <04db01ce27ec$5a3f5c30$0ebe1490$@sipalto.com>
Message-ID: <20130323204018.49f4428a@quenya>

El Sat, 23 Mar 2013 17:32:11 -0000
"Ahmed Murad" <am at sipalto.com> escribi?:

> Hello,
> 
>  
> 
> I am new to SIPWISE, do you have any guides on how to route calls from 1
> peer to another peer via the SIPWISE box?
> 
>  
> 
> Any guidance would be greatly appreciated.
> 
>  
> 
> Best regards,
> 
>  
>

Hi Ahmed

By default, the spce does not allow peer-peer calls. It only allows
peer-subscriber, subscriber-subscriber and subscriber-peer.

But you can indeed send calls from one peer to another. There are two ways. Let
me explain them and their differences:


- You have the option "force_outbound_calls_to_peer" you can set in the proxy
  preferences in the admin interface. If this option is set, any incoming call
  coming from this peer will be routed to pstn (to another or the same peer)
  based on the peer rules. This will be done even if the callee is a local
  subscriber.


- There's another option in config.yml which is called "allow_peer_relay".
  This is global and will work like this: When a call comes from a peer it
  will terminate the call locally if the calee is local or will send the call
  to the pstn following the peer rules if not.


The first method has the advantage of being specific for a single peer and you
can enable it per peer. I use it for sbc scenarios or pure peer-peer scenarios,
where you don't have local subscriber or those are "sbc subscribers with upper
registration". 

The second one is global, it will be enabled for all you peers. The advantage
is that it won't send out you local calls. I use this for migrations, where you
still need to send the traffic of legacy class5 systems via the new Sipwise
Class5.


In case you want to sell minutes to other peers and change them for it, the
above options have a problem: The rating engine is not prepared atm to bill and
charge peering servers. You'll get the cdrs and you will need to use an
external billing system for the charges. If this is your case you still have a
third option:

- Create your pstn gateways as peers. But other providers, you clients, the
  ones you want to charge create them as *subscribers*. Creating them as
  subscribers will allow you to rate them, apply ncos, prepaid (in case of
  pro) and lots of features not present in peer options. You just need to
  create a permanent contact for them, set them as trusted form the ip
  addresses they use and allow all the clis (if you want).


Please let me know which scenario you're looking for and if any of these three
options match it.

cheers,

Jon

 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130323/2ea74561/attachment.asc>

From jbonilla at sipwise.com  Sat Mar 23 15:44:02 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Sat, 23 Mar 2013 20:44:02 +0100
Subject: [Spce-user] Invoicing and Billing Peers/Trunks
In-Reply-To: <514C59A3.7090101@gmail.com>
References: <514C59A3.7090101@gmail.com>
Message-ID: <20130323204402.4eeaf025@quenya>

El Fri, 22 Mar 2013 15:16:19 +0200
Deon Vermeulen <vermeulen.deon at gmail.com> escribi?:

> Good Day
> 
> Is there any possibility that one could generate an Invoice to customers 
> with a Trunk to SPCE 2.7?


You can get the cdrs, but not an invoice as PDF is you mean that. Any customer,
has to be created as subscriber and it will be rated if you don't disable it.

> 
> Is there any possibility to Generate a Bill to a Peering Partner for 
> calls sent to SPCE 2.7. i.e Reverse billing?
> 

Not at the moment. Peers can't be charged in this version of the spce.


cheers,

Jon
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130323/d3643d13/attachment.asc>

From m.rys at tri-lakes.net  Sat Mar 23 16:21:46 2013
From: m.rys at tri-lakes.net (Marc Rys)
Date: Sat, 23 Mar 2013 15:21:46 -0500 (CDT)
Subject: [Spce-user] 483 Too Many Hops
In-Reply-To: <20130323204402.4eeaf025@quenya>
Message-ID: <c6ac07e3-43b0-4127-8a7d-8f562e0eef4c@usbsnpzimbra01v.servers.rystec.com>

I've been evaluating SPCE over the last week.  I've already setup a couple test subscribers and setup a peer with a provider we work with for SIP LD Term.  All of those test have worked very successful.

We also have our own media gateway which is interconnected with the local PSTN via TDM trunk, but I send incoming calls from the PSTN through our MediaGateway to SPCE, SPCE is responding back with 483 "Too Many Hops".  Attached is the wireshark cap.

Any help is appreciated.

Marc Rys
http://www.tri-lakes.net
http://www.rystec.com
-------------- next part --------------
A non-text attachment was scrubbed...
Name: SPCE-INBOUND-Taqua-483TooManyHops.pcap
Type: application/octet-stream
Size: 2220 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130323/3e72b68f/attachment.pcap>

From jbonilla at sipwise.com  Sat Mar 23 16:45:10 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Sat, 23 Mar 2013 21:45:10 +0100
Subject: [Spce-user] 483 Too Many Hops
In-Reply-To: <c6ac07e3-43b0-4127-8a7d-8f562e0eef4c@usbsnpzimbra01v.servers.rystec.com>
References: <20130323204402.4eeaf025@quenya>
	<c6ac07e3-43b0-4127-8a7d-8f562e0eef4c@usbsnpzimbra01v.servers.rystec.com>
Message-ID: <20130323214510.5498f5dc@quenya>

El Sat, 23 Mar 2013 15:21:46 -0500 (CDT)
Marc Rys <m.rys at tri-lakes.net> escribi?:

> En-Respuesta-A: <20130323204402.4eeaf025 at quenya>


Please don't answer a mail to create a new thread. Use the "new mail" button
instead. 


Cheers,

Jon

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130323/df575ddc/attachment.asc>

From lorenzo.mangani at gmail.com  Sat Mar 23 16:55:40 2013
From: lorenzo.mangani at gmail.com (Lorenzo Mangani)
Date: Sat, 23 Mar 2013 21:55:40 +0100
Subject: [Spce-user] 483 Too Many Hops
In-Reply-To: <c6ac07e3-43b0-4127-8a7d-8f562e0eef4c@usbsnpzimbra01v.servers.rystec.com>
References: <20130323204402.4eeaf025@quenya>
	<c6ac07e3-43b0-4127-8a7d-8f562e0eef4c@usbsnpzimbra01v.servers.rystec.com>
Message-ID: <CAKN+pHYO2a=jA5rJ=y52ONYW75Duh97o5b-RN4qK+kbSQo9NuA@mail.gmail.com>

Marc,

Your Tarqua invite has Max-Forwards set to 1.
Try increasing the allowed hops and the call will terminate.


Lorenzo Mangani

HOMER DEV TEAM
QXIP - Network Engineering

On Sat, Mar 23, 2013 at 9:21 PM, Marc Rys <m.rys at tri-lakes.net> wrote:

> I've been evaluating SPCE over the last week.  I've already setup a couple
> test subscribers and setup a peer with a provider we work with for SIP LD
> Term.  All of those test have worked very successful.
>
> We also have our own media gateway which is interconnected with the local
> PSTN via TDM trunk, but I send incoming calls from the PSTN through our
> MediaGateway to SPCE, SPCE is responding back with 483 "Too Many Hops".
>  Attached is the wireshark cap.
>
> Any help is appreciated.
>
> Marc Rys
> http://www.tri-lakes.net
> http://www.rystec.com
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
>
>


--
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130323/c66d0c64/attachment.html>

From m.rys at tri-lakes.net  Sat Mar 23 17:33:36 2013
From: m.rys at tri-lakes.net (Marc Rys)
Date: Sat, 23 Mar 2013 16:33:36 -0500 (CDT)
Subject: [Spce-user] 404 Not Found
In-Reply-To: <dd2ad9e2-70fc-4987-b767-fc7de6d5bb13@usbsnpzimbra01v.servers.rystec.com>
Message-ID: <396bde5e-b30d-42cb-ad5d-092d77b22c90@usbsnpzimbra01v.servers.rystec.com>

Ok, I adjusted my Taqua/Gateway to 4 hops and now I dont get the 483 anymore..  Now I get a 404 Not Found.

I'm so close..

pcap attached.

Marc Rys
http://www.tri-lakes.net
http://www.rystec.com
-------------- next part --------------
A non-text attachment was scrubbed...
Name: SPCE-INBOUND-Taqua-404NotFound.pcap
Type: application/octet-stream
Size: 6396 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130323/a7d2aebf/attachment.pcap>

From lorenzo.mangani at gmail.com  Sat Mar 23 17:39:50 2013
From: lorenzo.mangani at gmail.com (Lorenzo Mangani)
Date: Sat, 23 Mar 2013 22:39:50 +0100
Subject: [Spce-user] 483 Too Many Hops
In-Reply-To: <CAKN+pHYO2a=jA5rJ=y52ONYW75Duh97o5b-RN4qK+kbSQo9NuA@mail.gmail.com>
References: <20130323204402.4eeaf025@quenya>
	<c6ac07e3-43b0-4127-8a7d-8f562e0eef4c@usbsnpzimbra01v.servers.rystec.com>
	<CAKN+pHYO2a=jA5rJ=y52ONYW75Duh97o5b-RN4qK+kbSQo9NuA@mail.gmail.com>
Message-ID: <CAKN+pHaUiEuDtj9k=yKUcqSqWbHKCC=iTSoN5ZAOL6d7wF9fgA@mail.gmail.com>

Marc,

Please don't fork the messages by creating a new thread for each step of
the discussion and consult the documentation.
You need an inbound rewrite rule applied to strip the + from the INVITE in
order to match the local user, as well described in the Handbook, actually
this is exactly the example shown there:
http://www.sipwise.com/doc/2.7/spce/ar01s06.html#dialplans

Lorenzo Mangani

HOMER DEV TEAM
QXIP - Network Engineering

On Sat, Mar 23, 2013 at 9:55 PM, Lorenzo Mangani
<lorenzo.mangani at gmail.com>wrote:

> Marc,
>
> Your Tarqua invite has Max-Forwards set to 1.
> Try increasing the allowed hops and the call will terminate.
>
>
> Lorenzo Mangani
>
> HOMER DEV TEAM
> QXIP - Network Engineering
>
> On Sat, Mar 23, 2013 at 9:21 PM, Marc Rys <m.rys at tri-lakes.net> wrote:
>
>> I've been evaluating SPCE over the last week.  I've already setup a
>> couple test subscribers and setup a peer with a provider we work with for
>> SIP LD Term.  All of those test have worked very successful.
>>
>> We also have our own media gateway which is interconnected with the local
>> PSTN via TDM trunk, but I send incoming calls from the PSTN through our
>> MediaGateway to SPCE, SPCE is responding back with 483 "Too Many Hops".
>>  Attached is the wireshark cap.
>>
>> Any help is appreciated.
>>
>> Marc Rys
>> http://www.tri-lakes.net
>> http://www.rystec.com
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/listinfo/spce-user
>>
>>
>
>
> --
>
>
>
>


--
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130323/8385ad4c/attachment.html>

From jbonilla at sipwise.com  Sat Mar 23 17:40:18 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Sat, 23 Mar 2013 22:40:18 +0100
Subject: [Spce-user] 404 Not Found
In-Reply-To: <396bde5e-b30d-42cb-ad5d-092d77b22c90@usbsnpzimbra01v.servers.rystec.com>
References: <dd2ad9e2-70fc-4987-b767-fc7de6d5bb13@usbsnpzimbra01v.servers.rystec.com>
	<396bde5e-b30d-42cb-ad5d-092d77b22c90@usbsnpzimbra01v.servers.rystec.com>
Message-ID: <20130323224018.4cc3eab0@quenya>

El Sat, 23 Mar 2013 16:33:36 -0500 (CDT)
Marc Rys <m.rys at tri-lakes.net> escribi?:

> Ok, I adjusted my Taqua/Gateway to 4 hops and now I dont get the 483
> anymore..  Now I get a 404 Not Found.
> 


Please check the kamailio-proxy logs for that call. Most probably the requested
number was not local and the system returns a 404.

Are you using rewrite rules to normalize the call to E164 format?

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130323/b0dba2a9/attachment.asc>

From wfulton at thirdhatch.com  Sat Mar 23 17:34:07 2013
From: wfulton at thirdhatch.com (William Fulton)
Date: Sat, 23 Mar 2013 14:34:07 -0700
Subject: [Spce-user] Sipwise CE 2.7 SBC config revisited
In-Reply-To: <20130323111316.773fab5b@quenya>
References: <0F449623D8AAB145A3278882C750BA5831C713@sea2k3ex06.thirdhatch.corp>
	<20130323111316.773fab5b@quenya>
Message-ID: <0F449623D8AAB145A3278882C750BA5831C719@sea2k3ex06.thirdhatch.corp>

Jon,

The SPCE system is multi-homed with eth0 running on a public routable ip address and eth1 running on a private class c network.  We have an asterisk server running in the class c network as well.  We are trying to use the SPCE system to protect the asterisk server while allowing access through the public internet to authenticated users.

So, let?s say SPCE looks like this:
Eth0: x.x.x.x
Eth1: y.y.y.y

The Asterisk Server:
y.y.y.x

The client:
x.x.x.y

I want the client to authenticate against the SPCE, then the SPCE authenticates the client to the Asterisk server.  I can get the client to authenticate against the SPCE but I cannot get the SPCE to authenticate the client against the Asterisk server.

I have set peer_auth_register flag and also provided the credentials to the SPCE server for the client's account on the asterisk server along with the realm address for the asterisk server.  The SPCE never even tries to authenticate to the asterisk server.

Thank you,
William

-----Original Message-----
From: spce-user-bounces at lists.sipwise.com [mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of Jon Bonilla (Manwe)
Sent: Saturday, March 23, 2013 3:13 AM
To: spce-user at lists.sipwise.com
Subject: Re: [Spce-user] Sipwise CE 2.7 SBC config revisited

El Fri, 22 Mar 2013 16:27:39 -0700
"William Fulton" <wfulton at thirdhatch.com> escribi?:

> Guys,
> 
>  
> 
> I have spent a couple of weeks trying to make this work with 
> absolutely no luck.  I have read the manual cover to cover and there 
> is clearly a section missing.  Section 6 references a section in 
> Section 7 about advanced network configuration that does not exist.
> 
>  
> 
> I have two interfaces:
> 
>  
> 
> eth0 and eth1
> 
>  
> 
> *         eth0 is on the external subnet and this configuration was easy
> as I simply assigned the sip_ext, web_ext, ssh_ext, etc to this adaptor.
> 
> *         eth1 is on the internal subnet and I have added the role of
> sip_int to this adaptor.  


why did you do this? Why not lo interface for this role?

> 
> *         I created a new subscriber and set the preferences to
> authenticate to the peer and set the peer username, password and realm

Did you also check the "peer_auth_register" option?


> 
> *         Despite my best efforts, I cannot get the subscriber external
> to the Sipwise CE system to authenticate to the Asterisk system 
> internal to the Sipwise CE system.
> 

What? An external subscriber to authenticate against the internal asterisk? The
asterisk system running in the spce is just a voicebox endpoint.

Could you please explain what's you scenario and what you are trying to
achieve? Sorry but I didn't understand.

>  
> 
> What am I missing?  I'm sure it's pretty simple but I can't see it.
> 

Please tell us in more detail what your scenario is and what you did.


cheers,

Jon


From jbonilla at sipwise.com  Sat Mar 23 18:57:10 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Sat, 23 Mar 2013 23:57:10 +0100
Subject: [Spce-user] Sipwise CE 2.7 SBC config revisited
In-Reply-To: <0F449623D8AAB145A3278882C750BA5831C719@sea2k3ex06.thirdhatch.corp>
References: <0F449623D8AAB145A3278882C750BA5831C713@sea2k3ex06.thirdhatch.corp>
	<20130323111316.773fab5b@quenya>
	<0F449623D8AAB145A3278882C750BA5831C719@sea2k3ex06.thirdhatch.corp>
Message-ID: <20130323235710.275098f4@quenya>

El Sat, 23 Mar 2013 14:34:07 -0700
"William Fulton" <wfulton at thirdhatch.com> escribi?:

> Jon,
> 
> The SPCE system is multi-homed with eth0 running on a public routable ip
> address and eth1 running on a private class c network.  We have an asterisk
> server running in the class c network as well.  We are trying to use the SPCE
> system to protect the asterisk server while allowing access through the
> public internet to authenticated users.
> 
> So, let?s say SPCE looks like this:
> Eth0: x.x.x.x
> Eth1: y.y.y.y
> 
> The Asterisk Server:
> y.y.y.x
> 
> The client:
> x.x.x.y
> 
> I want the client to authenticate against the SPCE, then the SPCE
> authenticates the client to the Asterisk server.  I can get the client to
> authenticate against the SPCE but I cannot get the SPCE to authenticate the
> client against the Asterisk server.
> 
> I have set peer_auth_register flag and also provided the credentials to the
> SPCE server for the client's account on the asterisk server along with the
> realm address for the asterisk server.  The SPCE never even tries to
> authenticate to the asterisk server.
> 
> 

Th spce will always use sip_ext as incoming and outgoing interface for the sip
traffic. sip_int is only for internal communication of the spce elements.

If you want the spce to communicate to the asterisk server and register there,
it will use x.x.x.x ip to go to y.y.y.x


Do you have the network and the routes prepared for that? You router should
have the routes for the spce to reach the asterisk server in the internal
network.

I think this is more a networking issue here.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130323/bcca0a19/attachment.asc>

From nh.developer at gmail.com  Sat Mar 23 20:19:16 2013
From: nh.developer at gmail.com (Developing Tomorrow)
Date: Sun, 24 Mar 2013 01:19:16 +0100
Subject: [Spce-user] Spce-user Digest, Vol 28, Issue 52
In-Reply-To: <mailman.1220.1364071519.2547.spce-user@lists.sipwise.com>
References: <mailman.1220.1364071519.2547.spce-user@lists.sipwise.com>
Message-ID: <00d101ce2825$390cbc40$ab2634c0$@gmail.com>


Hi Jon 

Thanks

When I discovered the fact that the rtp was sent thru the normal internet,
that is what I thought
For the router to send the packets thru the vpn.
The problem that came is that, I did not know how to do that.

Technically I know the command to force a host thru the vpn, but I cannot
apply that because the vpn is created thru the internet
Now, if I tell the router to force the vpn host also thru the vpn, then it
cannot create itself and also go thru its own tunnel.

I than thought that, any udp packet sent from gateway should go thru the vpn
tunnel.
Anyone know how to translate that to iptables command?
Udp packets from 192.168.10.10 all thru route 10.0.0.2 (tunnel on router
side of gateway)

thx


_________________________________________________________________________
Please consider the environment before printing this email.
?
Notice: Please do not include customer identifying information (i.e.,
account number, social security, etc.) in email communications.
?
Privacy Statement: This message may contain confidential information that is
protected by law. If you received this message in error, please notify the
sender by fax or email and delete this message. If you properly received
this message, you may use its contents only in strictaccordance with our
instructions and privacy policy.
If any person makes a false or misleading representation to obtain customer
information, that person may have committed a federal crime, and we may
report that person to the proper authorities.

-----Original Message-----
From: spce-user-bounces at lists.sipwise.com
[mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of
spce-user-request at lists.sipwise.com
Sent: 23 March 2013 21:45
To: spce-user at lists.sipwise.com
Subject: Spce-user Digest, Vol 28, Issue 52

Send Spce-user mailing list submissions to
	spce-user at lists.sipwise.com

To subscribe or unsubscribe via the World Wide Web, visit
	http://lists.sipwise.com/listinfo/spce-user
or, via email, send a message with subject or body 'help' to
	spce-user-request at lists.sipwise.com

You can reach the person managing the list at
	spce-user-owner at lists.sipwise.com

When replying, please edit your Subject line so it is more specific than
"Re: Contents of Spce-user digest..."


Today's Topics:

   1. Re: Why does it take 35sec to register (Jon Bonilla (Manwe))
   2. Re: Sipwise over openvpn (Jon Bonilla (Manwe))
   3. Re: Connect softphone to SIPWISE (Jon Bonilla (Manwe))
   4. Re: routing calls between peers (Jon Bonilla (Manwe))
   5. Re: Invoicing and Billing Peers/Trunks (Jon Bonilla (Manwe))
   6. 483 Too Many Hops (Marc Rys)
   7. Re: 483 Too Many Hops (Jon Bonilla (Manwe))


----------------------------------------------------------------------

Message: 1
Date: Sat, 23 Mar 2013 20:21:10 +0100
From: Jon Bonilla (Manwe) <jbonilla at sipwise.com>
Subject: Re: [Spce-user] Why does it take 35sec to register
To: spce-user at lists.sipwise.com
Message-ID: <20130323202110.6f71ae8d at quenya>
Content-Type: text/plain; charset="utf-8"

El Sat, 23 Mar 2013 20:12:39 +0100
"Oliver Vermeulen" <oliver at oliverv.com> escribi?:

> Hello All,
> 
>  
> 
> I tested sipwise yesterday.. install was successful very easy.
> 
>  
> 
> Does anybody have problems with sip registration ?
> 
> Why does it take 35sec to register.
> 
>  
> 
> Also when it's registered and I dial takes long time to process call.
> 
> 

Please make a ngrep-sip capture of a registration and another one of a call.
We will be able to see how long takes.


-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130323/afee13bc/
attachment-0001.pgp>

------------------------------

Message: 2
Date: Sat, 23 Mar 2013 20:24:46 +0100
From: Jon Bonilla (Manwe) <jbonilla at sipwise.com>
Subject: Re: [Spce-user] Sipwise over openvpn
To: spce-user at lists.sipwise.com
Message-ID: <20130323202446.1bc5eeed at quenya>
Content-Type: text/plain; charset="utf-8"

El Sat, 23 Mar 2013 20:03:01 +0100
"Developing Tomorrow" <nh.developer at gmail.com> escribi?:

> Hi.,
> 
>  
> 
> I have "successfully" installed sipwise and am able to register users 
> and make calls using any "normal" providers.
> 
> Now
> 
> In the same box, I have created a tunnel to another network that has a 
> gateway.
> 
> I am able to ping that gateway from the station that has sipwise also 
> able to send traffic to the gateway, but I will get no answer because 
> of the ip used.
> 
> Let me put down the layout:
> 
> All the following ips are fake but will clearly show the layout.
> 
>  
> 
> Sipwise external ip: 200.200.200.200
> 
> Sipwise internal ip: 10.0.0.1
> 
>  
> 
> Tunnel router: 10.0.0.10
> 
> Gateway across tunel: 192.168.10.10
> 
>  
> 
> If I use a standard asterisk box, give it ip 10.0.1 and send calls to 
> gateway, the voice will come back thru tunnel because the gateway is 
> replying to 10.0.0.1
> 
>  
> 
> In my sipwise case, it does not work as sipwise is sending the call 
> with ip
> 200.200.200.200 so the gateway is answering thru the normal internet 
> and not the tunnel.
> 
>  
> 
> How can I configure sipwise to send info to that gateway using ip in 
> the form ttttttttt at 10.0.0.1 <mailto:ttttttttt at 10.0.0.1>  and not
> ttttttttt at 200.200.200.200 <mailto:ttttttttt at 200.200.200.200> .
> 
>  

Why don't you configure in the other side of the tunnel. "Send the traffic
to
20.200.200.200 via 10.0.0.1" ?

You can use the "extra_socket" option in the spce and that will work for you
at sip signaling level but at rtp level it won't work so you'd get signaling
via the tunel and media over internet. If that's what you want it's fine. 

Anyways I'd continue using the external main address and try to fix you
issue at networking level, with routes.


> 
> The media relay is supposed to do that function but I have no clue on 
> how to configure it.
> 
> I could only create a socket with ip 10.0.0.1 and that allows me to 
> send the traffic thru the vpn, but does not change the ip that the gateway
receives.
> 



cheers,

Jon

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130323/d6223bd4/
attachment-0001.pgp>

------------------------------

Message: 3
Date: Sat, 23 Mar 2013 20:26:05 +0100
From: Jon Bonilla (Manwe) <jbonilla at sipwise.com>
Subject: Re: [Spce-user] Connect softphone to SIPWISE
To: spce-user at lists.sipwise.com
Message-ID: <20130323202605.57ed0cdd at quenya>
Content-Type: text/plain; charset="utf-8"

El Sat, 23 Mar 2013 17:36:19 -0000
"Ahmed Murad" <am at sipalto.com> escribi?:

> Hello,
> 
>  
> 
> I have been following your guides and successfully created & connected 
> to a couple of external peers.
> 
>  
> 
> However I am having trouble connecting a softphone to our first
subscriber.
> I have created the domain, account and subscriber.
> 
>  
> 
> But I get a timeout 500 error when I try to connect the softphone to 
> the subscriber, I have ensured there are no firewall rules causing the
timeout.
> 
>  
> 
> Please could you point me in the right direction in regards to 
> troubleshooting, is there anything I've missed?
> 
>  
> 

Please do a ngre-sip capture in the systeem and let us check the
incoming/outgoing traffic.

Also, you can check /var/log/ngcp/kamaiio-proxy.log to check if REGISTER
requests are arriving and what's going on with them.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130323/8d7f84f5/
attachment-0001.pgp>

------------------------------

Message: 4
Date: Sat, 23 Mar 2013 20:40:18 +0100
From: Jon Bonilla (Manwe) <jbonilla at sipwise.com>
Subject: Re: [Spce-user] routing calls between peers
To: spce-user at lists.sipwise.com
Message-ID: <20130323204018.49f4428a at quenya>
Content-Type: text/plain; charset="utf-8"

El Sat, 23 Mar 2013 17:32:11 -0000
"Ahmed Murad" <am at sipalto.com> escribi?:

> Hello,
> 
>  
> 
> I am new to SIPWISE, do you have any guides on how to route calls from 
> 1 peer to another peer via the SIPWISE box?
> 
>  
> 
> Any guidance would be greatly appreciated.
> 
>  
> 
> Best regards,
> 
>  
>

Hi Ahmed

By default, the spce does not allow peer-peer calls. It only allows
peer-subscriber, subscriber-subscriber and subscriber-peer.

But you can indeed send calls from one peer to another. There are two ways.
Let me explain them and their differences:


- You have the option "force_outbound_calls_to_peer" you can set in the
proxy
  preferences in the admin interface. If this option is set, any incoming
call
  coming from this peer will be routed to pstn (to another or the same peer)
  based on the peer rules. This will be done even if the callee is a local
  subscriber.


- There's another option in config.yml which is called "allow_peer_relay".
  This is global and will work like this: When a call comes from a peer it
  will terminate the call locally if the calee is local or will send the
call
  to the pstn following the peer rules if not.


The first method has the advantage of being specific for a single peer and
you can enable it per peer. I use it for sbc scenarios or pure peer-peer
scenarios, where you don't have local subscriber or those are "sbc
subscribers with upper registration". 

The second one is global, it will be enabled for all you peers. The
advantage is that it won't send out you local calls. I use this for
migrations, where you still need to send the traffic of legacy class5
systems via the new Sipwise Class5.


In case you want to sell minutes to other peers and change them for it, the
above options have a problem: The rating engine is not prepared atm to bill
and charge peering servers. You'll get the cdrs and you will need to use an
external billing system for the charges. If this is your case you still have
a third option:

- Create your pstn gateways as peers. But other providers, you clients, the
  ones you want to charge create them as *subscribers*. Creating them as
  subscribers will allow you to rate them, apply ncos, prepaid (in case of
  pro) and lots of features not present in peer options. You just need to
  create a permanent contact for them, set them as trusted form the ip
  addresses they use and allow all the clis (if you want).


Please let me know which scenario you're looking for and if any of these
three options match it.

cheers,

Jon

 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130323/2ea74561/
attachment-0001.pgp>

------------------------------

Message: 5
Date: Sat, 23 Mar 2013 20:44:02 +0100
From: Jon Bonilla (Manwe) <jbonilla at sipwise.com>
Subject: Re: [Spce-user] Invoicing and Billing Peers/Trunks
To: spce-user at lists.sipwise.com
Message-ID: <20130323204402.4eeaf025 at quenya>
Content-Type: text/plain; charset="utf-8"

El Fri, 22 Mar 2013 15:16:19 +0200
Deon Vermeulen <vermeulen.deon at gmail.com> escribi?:

> Good Day
> 
> Is there any possibility that one could generate an Invoice to 
> customers with a Trunk to SPCE 2.7?


You can get the cdrs, but not an invoice as PDF is you mean that. Any
customer, has to be created as subscriber and it will be rated if you don't
disable it.

> 
> Is there any possibility to Generate a Bill to a Peering Partner for 
> calls sent to SPCE 2.7. i.e Reverse billing?
> 

Not at the moment. Peers can't be charged in this version of the spce.


cheers,

Jon
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130323/d3643d13/
attachment-0001.pgp>

------------------------------

Message: 6
Date: Sat, 23 Mar 2013 15:21:46 -0500 (CDT)
From: Marc Rys <m.rys at tri-lakes.net>
Subject: [Spce-user] 483 Too Many Hops
To: spce-user at lists.sipwise.com
Message-ID:
	
<c6ac07e3-43b0-4127-8a7d-8f562e0eef4c at usbsnpzimbra01v.servers.rystec.com>
	
Content-Type: text/plain; charset="utf-8"

I've been evaluating SPCE over the last week.  I've already setup a couple
test subscribers and setup a peer with a provider we work with for SIP LD
Term.  All of those test have worked very successful.

We also have our own media gateway which is interconnected with the local
PSTN via TDM trunk, but I send incoming calls from the PSTN through our
MediaGateway to SPCE, SPCE is responding back with 483 "Too Many Hops".
Attached is the wireshark cap.

Any help is appreciated.

Marc Rys
http://www.tri-lakes.net
http://www.rystec.com
-------------- next part --------------
A non-text attachment was scrubbed...
Name: SPCE-INBOUND-Taqua-483TooManyHops.pcap
Type: application/octet-stream
Size: 2220 bytes
Desc: not available
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130323/3e72b68f/
attachment-0001.obj>

------------------------------

Message: 7
Date: Sat, 23 Mar 2013 21:45:10 +0100
From: Jon Bonilla (Manwe) <jbonilla at sipwise.com>
Subject: Re: [Spce-user] 483 Too Many Hops
To: spce-user at lists.sipwise.com
Message-ID: <20130323214510.5498f5dc at quenya>
Content-Type: text/plain; charset="utf-8"

El Sat, 23 Mar 2013 15:21:46 -0500 (CDT) Marc Rys <m.rys at tri-lakes.net>
escribi?:

> En-Respuesta-A: <20130323204402.4eeaf025 at quenya>


Please don't answer a mail to create a new thread. Use the "new mail" button
instead. 


Cheers,

Jon

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130323/df575ddc/
attachment.pgp>

------------------------------

_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com
http://lists.sipwise.com/listinfo/spce-user


End of Spce-user Digest, Vol 28, Issue 52
*****************************************




From wfulton at thirdhatch.com  Sun Mar 24 01:21:55 2013
From: wfulton at thirdhatch.com (William Fulton)
Date: Sat, 23 Mar 2013 22:21:55 -0700
Subject: [Spce-user] Sipwise CE 2.7 SBC config revisited
In-Reply-To: <20130323235710.275098f4@quenya>
References: <0F449623D8AAB145A3278882C750BA5831C713@sea2k3ex06.thirdhatch.corp><20130323111316.773fab5b@quenya><0F449623D8AAB145A3278882C750BA5831C719@sea2k3ex06.thirdhatch.corp>
	<20130323235710.275098f4@quenya>
Message-ID: <0F449623D8AAB145A3278882C750BA5831C71B@sea2k3ex06.thirdhatch.corp>


>Th spce will always use sip_ext as incoming and outgoing interface for the sip traffic. sip_int is only for internal communication of the spce elements.

>If you want the spce to communicate to the asterisk server and register there, it will use x.x.x.x ip to go to y.y.y.x


>Do you have the network and the routes prepared for that? You router should have the routes for the spce to reach the asterisk server in the internal >network.

>I think this is more a networking issue here.
So the SPCE does not know to use eth1 to access servers on that subnet?  Is there a way to designate both eth0 and eth1 for sip_ext?

Thank you,
William

From jbonilla at sipwise.com  Sun Mar 24 05:50:35 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Sun, 24 Mar 2013 10:50:35 +0100
Subject: [Spce-user] Sipwise CE 2.7 SBC config revisited
In-Reply-To: <0F449623D8AAB145A3278882C750BA5831C71B@sea2k3ex06.thirdhatch.corp>
References: <0F449623D8AAB145A3278882C750BA5831C713@sea2k3ex06.thirdhatch.corp>
	<20130323111316.773fab5b@quenya>
	<0F449623D8AAB145A3278882C750BA5831C719@sea2k3ex06.thirdhatch.corp>
	<20130323235710.275098f4@quenya>
	<0F449623D8AAB145A3278882C750BA5831C71B@sea2k3ex06.thirdhatch.corp>
Message-ID: <20130324105035.3ca73296@quenya>

El Sat, 23 Mar 2013 22:21:55 -0700
"William Fulton" <wfulton at thirdhatch.com> escribi?:

> 
> >Th spce will always use sip_ext as incoming and outgoing interface for the
> >sip traffic. sip_int is only for internal communication of the spce elements.
> 
> >If you want the spce to communicate to the asterisk server and register
> >there, it will use x.x.x.x ip to go to y.y.y.x
> 
> 
> >Do you have the network and the routes prepared for that? You router should
> >have the routes for the spce to reach the asterisk server in the internal
> >>network.
> 
> >I think this is more a networking issue here.
> So the SPCE does not know to use eth1 to access servers on that subnet?  Is
> there a way to designate both eth0 and eth1 for sip_ext?


Hi William.

No, the spce will listen only a single ip address. From that address, it's up
to the network to provide routes to the internet and to any internal network
you want to reach.

Can't your router provide that? Can't your internal elements reach the spce's
public ip address?

You can always do it via "route commands" and "ip_forward" enabled on the
system. But the "normal" way of doing this is network routing in a router.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130324/12266e3e/attachment.asc>

From m.rys at tri-lakes.net  Sun Mar 24 11:54:57 2013
From: m.rys at tri-lakes.net (Marc Rys)
Date: Sun, 24 Mar 2013 10:54:57 -0500 (CDT)
Subject: [Spce-user] 483 Too Many Hops
In-Reply-To: <CAKN+pHaUiEuDtj9k=yKUcqSqWbHKCC=iTSoN5ZAOL6d7wF9fgA@mail.gmail.com>
Message-ID: <9dbde9c8-a122-472d-841f-f7b03069dd21@usbsnpzimbra01v.servers.rystec.com>

ok I'll stop changing the subject, even though I regret not naming it SPCE Taqua Integration.. 

Anyways, I've heeded your advice and I normalized the patterns and I'm now routing the calls from my Taqua PSTN gateway to my IP phones.  Now the newest problem to reveal itself is the incoming calls to my Phones fail to setup completely.  It appears all the call routing is working properly now,  I can call my DID from the PSTN, and my IP phone rings, but when I pickup the call, I hear no audio, and it appears the phone never completely sets up the call.  The screen still shows the phone as ringing.

The wireshark cap shows RTP moving between my gateway & SPCE, and RTP moving between SPCE and my IP phone, but the call never appears to setup completely on the IP phone.

Any thoughts?

Marc Rys 
http://www.tri-lakes.net 
http://www.rystec.com 


----- Original Message -----
From: "Lorenzo Mangani" <lorenzo.mangani at gmail.com>
To: "Marc Rys" <m.rys at tri-lakes.net>
Cc: spce-user at lists.sipwise.com
Sent: Saturday, March 23, 2013 4:39:50 PM
Subject: Re: [Spce-user] 483 Too Many Hops

Marc, 


Please don't fork the messages by creating a new thread for each step of the discussion and consult the documentation. 
You need an inbound rewrite rule applied to strip the + from the INVITE in order to match the local user, as well described in the Handbook, actually this is exactly the example shown there: http://www.sipwise.com/doc/2.7/spce/ar01s06.html#dialplans 



Lorenzo Mangani 



HOMER DEV TEAM 
QXIP - Network Engineering 


On Sat, Mar 23, 2013 at 9:55 PM, Lorenzo Mangani < lorenzo.mangani at gmail.com > wrote: 


Marc, 


Your Tarqua invite has Max-Forwards set to 1. 
Try increasing the allowed hops and the call will terminate. 





Lorenzo Mangani 



HOMER DEV TEAM 
QXIP - Network Engineering 



On Sat, Mar 23, 2013 at 9:21 PM, Marc Rys < m.rys at tri-lakes.net > wrote: 




I've been evaluating SPCE over the last week. I've already setup a couple test subscribers and setup a peer with a provider we work with for SIP LD Term. All of those test have worked very successful. 

We also have our own media gateway which is interconnected with the local PSTN via TDM trunk, but I send incoming calls from the PSTN through our MediaGateway to SPCE, SPCE is responding back with 483 "Too Many Hops". Attached is the wireshark cap. 

Any help is appreciated. 

Marc Rys 
http://www.tri-lakes.net 
http://www.rystec.com 
_______________________________________________ 
Spce-user mailing list 
Spce-user at lists.sipwise.com 
http://lists.sipwise.com/listinfo/spce-user 





-- 










-- 








-------------- next part --------------
A non-text attachment was scrubbed...
Name: SPCE-InboundFromTaqua-NoAudio1.pcap
Type: application/octet-stream
Size: 452641 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130324/c866f247/attachment.pcap>

From m.rys at tri-lakes.net  Sun Mar 24 12:19:02 2013
From: m.rys at tri-lakes.net (Marc Rys)
Date: Sun, 24 Mar 2013 11:19:02 -0500 (CDT)
Subject: [Spce-user] 483 Too Many Hops
In-Reply-To: <9dbde9c8-a122-472d-841f-f7b03069dd21@usbsnpzimbra01v.servers.rystec.com>
Message-ID: <1c5d4ad6-1793-41ee-bc29-e46c581f0c55@usbsnpzimbra01v.servers.rystec.com>

Also, just as an FYI,  Both of my test IP Phone subscribers can all each other flawlessly.  The calls between them setup just as expected.  (video worked too!)

Marc Rys 
http://www.tri-lakes.net 
http://www.rystec.com 

----- Original Message -----
From: "Marc Rys" <m.rys at tri-lakes.net>
To: "Marc Rys" <m.rys at zimbra.rystec.com>
Cc: spce-user at lists.sipwise.com
Sent: Sunday, March 24, 2013 10:54:57 AM
Subject: Re: [Spce-user] 483 Too Many Hops

ok I'll stop changing the subject, even though I regret not naming it SPCE Taqua Integration.. 

Anyways, I've heeded your advice and I normalized the patterns and I'm now routing the calls from my Taqua PSTN gateway to my IP phones.  Now the newest problem to reveal itself is the incoming calls to my Phones fail to setup completely.  It appears all the call routing is working properly now,  I can call my DID from the PSTN, and my IP phone rings, but when I pickup the call, I hear no audio, and it appears the phone never completely sets up the call.  The screen still shows the phone as ringing.

The wireshark cap shows RTP moving between my gateway & SPCE, and RTP moving between SPCE and my IP phone, but the call never appears to setup completely on the IP phone.

Any thoughts?

Marc Rys 
http://www.tri-lakes.net 
http://www.rystec.com 


----- Original Message -----
From: "Lorenzo Mangani" <lorenzo.mangani at gmail.com>
To: "Marc Rys" <m.rys at tri-lakes.net>
Cc: spce-user at lists.sipwise.com
Sent: Saturday, March 23, 2013 4:39:50 PM
Subject: Re: [Spce-user] 483 Too Many Hops

Marc, 


Please don't fork the messages by creating a new thread for each step of the discussion and consult the documentation. 
You need an inbound rewrite rule applied to strip the + from the INVITE in order to match the local user, as well described in the Handbook, actually this is exactly the example shown there: http://www.sipwise.com/doc/2.7/spce/ar01s06.html#dialplans 



Lorenzo Mangani 



HOMER DEV TEAM 
QXIP - Network Engineering 


On Sat, Mar 23, 2013 at 9:55 PM, Lorenzo Mangani < lorenzo.mangani at gmail.com > wrote: 


Marc, 


Your Tarqua invite has Max-Forwards set to 1. 
Try increasing the allowed hops and the call will terminate. 





Lorenzo Mangani 



HOMER DEV TEAM 
QXIP - Network Engineering 



On Sat, Mar 23, 2013 at 9:21 PM, Marc Rys < m.rys at tri-lakes.net > wrote: 




I've been evaluating SPCE over the last week. I've already setup a couple test subscribers and setup a peer with a provider we work with for SIP LD Term. All of those test have worked very successful. 

We also have our own media gateway which is interconnected with the local PSTN via TDM trunk, but I send incoming calls from the PSTN through our MediaGateway to SPCE, SPCE is responding back with 483 "Too Many Hops". Attached is the wireshark cap. 

Any help is appreciated. 

Marc Rys 
http://www.tri-lakes.net 
http://www.rystec.com 
_______________________________________________ 
Spce-user mailing list 
Spce-user at lists.sipwise.com 
http://lists.sipwise.com/listinfo/spce-user 





-- 










-- 









_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com
http://lists.sipwise.com/listinfo/spce-user



From lorenzo.mangani at gmail.com  Sun Mar 24 12:25:03 2013
From: lorenzo.mangani at gmail.com (Lorenzo Mangani)
Date: Sun, 24 Mar 2013 17:25:03 +0100
Subject: [Spce-user] 483 Too Many Hops
In-Reply-To: <9dbde9c8-a122-472d-841f-f7b03069dd21@usbsnpzimbra01v.servers.rystec.com>
References: <CAKN+pHaUiEuDtj9k=yKUcqSqWbHKCC=iTSoN5ZAOL6d7wF9fgA@mail.gmail.com>
	<9dbde9c8-a122-472d-841f-f7b03069dd21@usbsnpzimbra01v.servers.rystec.com>
Message-ID: <CAKN+pHYD6tE55RxhbMoTx92cuOoVcn-GDRPpNv=GeUoFifK+dQ@mail.gmail.com>

Hi Marc,

As you can see from your trace, the call is never completely established,
as the 200 OK's ACK never makes it back to the user-agent.
After a brief look this seems to be caused by the gateway always using the
contact header to respond instead of the from header, as for the BYE
message also being rejected for the same reason - just my 2 cents, might be
wrong - maybe someone else on the list will have different input,

Best,

Lorenzo Mangani

HOMER DEV TEAM
QXIP - Network Engineering



On Sun, Mar 24, 2013 at 4:54 PM, Marc Rys <m.rys at tri-lakes.net> wrote:

> ok I'll stop changing the subject, even though I regret not naming it SPCE
> Taqua Integration..
>
> Anyways, I've heeded your advice and I normalized the patterns and I'm now
> routing the calls from my Taqua PSTN gateway to my IP phones.  Now the
> newest problem to reveal itself is the incoming calls to my Phones fail to
> setup completely.  It appears all the call routing is working properly now,
>  I can call my DID from the PSTN, and my IP phone rings, but when I pickup
> the call, I hear no audio, and it appears the phone never completely sets
> up the call.  The screen still shows the phone as ringing.
>
> The wireshark cap shows RTP moving between my gateway & SPCE, and RTP
> moving between SPCE and my IP phone, but the call never appears to setup
> completely on the IP phone.
>
> Any thoughts?
>
> Marc Rys
> http://www.tri-lakes.net
> http://www.rystec.com
>
>
> ----- Original Message -----
> From: "Lorenzo Mangani" <lorenzo.mangani at gmail.com>
> To: "Marc Rys" <m.rys at tri-lakes.net>
> Cc: spce-user at lists.sipwise.com
> Sent: Saturday, March 23, 2013 4:39:50 PM
> Subject: Re: [Spce-user] 483 Too Many Hops
>
> Marc,
>
>
> Please don't fork the messages by creating a new thread for each step of
> the discussion and consult the documentation.
> You need an inbound rewrite rule applied to strip the + from the INVITE in
> order to match the local user, as well described in the Handbook, actually
> this is exactly the example shown there:
> http://www.sipwise.com/doc/2.7/spce/ar01s06.html#dialplans
>
>
>
> Lorenzo Mangani
>
>
>
> HOMER DEV TEAM
> QXIP - Network Engineering
>
>
> On Sat, Mar 23, 2013 at 9:55 PM, Lorenzo Mangani <
> lorenzo.mangani at gmail.com > wrote:
>
>
> Marc,
>
>
> Your Tarqua invite has Max-Forwards set to 1.
> Try increasing the allowed hops and the call will terminate.
>
>
>
>
>
> Lorenzo Mangani
>
>
>
> HOMER DEV TEAM
> QXIP - Network Engineering
>
>
>
> On Sat, Mar 23, 2013 at 9:21 PM, Marc Rys < m.rys at tri-lakes.net > wrote:
>
>
>
>
> I've been evaluating SPCE over the last week. I've already setup a couple
> test subscribers and setup a peer with a provider we work with for SIP LD
> Term. All of those test have worked very successful.
>
> We also have our own media gateway which is interconnected with the local
> PSTN via TDM trunk, but I send incoming calls from the PSTN through our
> MediaGateway to SPCE, SPCE is responding back with 483 "Too Many Hops".
> Attached is the wireshark cap.
>
> Any help is appreciated.
>
> Marc Rys
> http://www.tri-lakes.net
> http://www.rystec.com
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
>
>
>
>
>
> --
>
>
>
>
>
>
>
>
>
>
> --
>
>
>
>
>
>
>
>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
>
>


--
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130324/20afc139/attachment.html>

From agranig at sipwise.com  Sun Mar 24 12:43:18 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Sun, 24 Mar 2013 17:43:18 +0100
Subject: [Spce-user] 483 Too Many Hops
In-Reply-To: <CAKN+pHYD6tE55RxhbMoTx92cuOoVcn-GDRPpNv=GeUoFifK+dQ@mail.gmail.com>
References: <CAKN+pHaUiEuDtj9k=yKUcqSqWbHKCC=iTSoN5ZAOL6d7wF9fgA@mail.gmail.com>
	<9dbde9c8-a122-472d-841f-f7b03069dd21@usbsnpzimbra01v.servers.rystec.com>
	<CAKN+pHYD6tE55RxhbMoTx92cuOoVcn-GDRPpNv=GeUoFifK+dQ@mail.gmail.com>
Message-ID: <514F2D26.3030103@sipwise.com>

Hi,

On 03/24/2013 05:25 PM, Lorenzo Mangani wrote:
> As you can see from your trace, the call is never completely
> established, as the 200 OK's ACK never makes it back to the user-agent.
> After a brief look this seems to be caused by the gateway always using
> the contact header to respond instead of the from header, as for the BYE
> message also being rejected for the same reason - just my 2 cents, might
> be wrong - maybe someone else on the list will have different input,
>

The Contact handling is as expected, but the 208.73.198.12 doesn't 
convert the Record-Route headers added by the SPCE in the 200-OK (packet 
#14) into Route headers of the ACK (packet #15), so the proxy has no 
information where to relay the ACK to (note that ACKs to 200 are relayed 
end-to-end, so all Route information must be preserved).

The sender of the ACK *must* take the Record-Route headers of the 
previous 200-OK, convert them to Routes and add them to the message in 
reverse order, and send the request to the URI found in the top-most Route.

Andreas



From oliver at oliverv.com  Sun Mar 24 18:57:05 2013
From: oliver at oliverv.com (Oliver Vermeulen)
Date: Sun, 24 Mar 2013 23:57:05 +0100
Subject: [Spce-user] Question of features
Message-ID: <02c601ce28e2$e86a8ce0$b93fa6a0$@oliverv.com>

Hello ,

 

I'm testing sipwise , had some questions if anybody can answer me.

 

User CE version

 

1)      Does is support simple sip (messaging)

2)      Xcap 

3)      Presence

4)      Push notifications 

5)      Is possible to use external billing system via Radius ?

 

What clients are supported ? For all features ?

 

Best Regards,

 

Olivier Vermeulen

Description: oliverv.jpg


USA:  +1 (305) 722-1457
BELGIUM: + (32) 9-3955620
SPAIN: + (34) 931-845025
RO: + (40) 21-569-4703
msn:  <mailto:oliver at oliverv.com> oliver at oliverv.com

 <http://www.wvg-tele.com/> http://www.oliverv.com

****************************************************************************
************ 
This E-mail message, including any attachments, is intended only for the
person or entity to which it is addressed, 
and may contain confidential Information. If you are not the intended
recipient, any review, retransmission, disclosure, copying, 
modification or other use of this E-mail message or attachments is strictly
forbidden. If you have received this E-mail message in error,
please contact the author and delete the message and any attachments from
your computer.
P Think before you print.

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130324/78edfec5/attachment.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.png
Type: application/octet-stream
Size: 21421 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130324/78edfec5/attachment.png>

From skchopperguy at gmail.com  Mon Mar 25 04:35:36 2013
From: skchopperguy at gmail.com (Skyler)
Date: Mon, 25 Mar 2013 01:35:36 -0700
Subject: [Spce-user] Question of features
In-Reply-To: <02c601ce28e2$e86a8ce0$b93fa6a0$@oliverv.com>
References: <02c601ce28e2$e86a8ce0$b93fa6a0$@oliverv.com>
Message-ID: <51500C58.6080902@gmail.com>

Hi,

  1. yes
  2. yes
  3. yes
  4. yes with work
  5. yes
  6. All that send REGISTER
  7. yes

  --Skyler

On 3/24/2013 3:57 PM, Oliver Vermeulen wrote:
> Hello ,
>
> I?m testing sipwise , had some questions if anybody can answer me.
>
> User CE version
>
> 1)Does is support simple sip (messaging)
>
> 2)Xcap
>
> 3)Presence
>
> 4)Push notifications
>
> 5)Is possible to use external billing system via Radius ?
>
> What clients are supported ? For all features ?
>
> *Best Regards,*
>
> *Olivier Vermeulen**
>
> *Description: oliverv.jpg
>
> *
> USA:*  +1 (305) 722-1457
> *BELGIUM: *+ (32) 9-3955620
> *SPAIN: *+ (34) 931-845025
> *RO:* + (40) 21-569-4703*
> msn:*oliver at oliverv.com <mailto:oliver at oliverv.com>
>
> http://www.oliverv.com <http://www.wvg-tele.com/>
>
> ****************************************************************************************
>
> This E-mail message, including any attachments, is intended only for the
> person or entity to which it is addressed,
> and may contain confidential Information. If you are not the intended
> recipient, any review, retransmission, disclosure, copying,
> modification or other use of this E-mail message or attachments is
> strictly forbidden. If you have received this E-mail message in error,
> please contact the author and delete the message and any attachments
> from your computer.
> P***Think before you print.*
>
>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
>



From mario.contreras at innovasur.es  Mon Mar 25 06:21:19 2013
From: mario.contreras at innovasur.es (Mario Contreras)
Date: Mon, 25 Mar 2013 11:21:19 +0100
Subject: [Spce-user] Outgoing calls in subscriber's cdr (admin panel)
Message-ID: <5150251F.10504@innovasur.es>

Hi all,

I've realized that since I upgraded to 2.7, from certain subscribers(I 
think all subscribers are configured in asterisk), outgoing calls are 
not showing in web admin panel, and that's not happening in csc.

I've seen that the file with this output is 
/usr/share/ngcp-www-admin/lib/admin/root/tt/subscriber_call_data.tt, 
right? But I don't know exactly how provisioning works, so I can't solve 
it. It's not a big deal, really...

Anyway, with the new details link, when the subscriber have many calls, 
it takes very long to load. Can it be done using AJAX? It's only an idea.

Thanks for your great job!



From am at sipalto.com  Mon Mar 25 07:19:20 2013
From: am at sipalto.com (Ahmed Murad)
Date: Mon, 25 Mar 2013 11:19:20 -0000
Subject: [Spce-user] Connect softphone to SIPWISE (Jon Bonilla (Manwe))
Message-ID: <058901ce294a$9919d9d0$cb4d8d70$@sipalto.com>

Hi,

Thanks for the info. I received help on connecting a UA to SIPWISE from
another user who advised to run the below commands, after running them UA's
connected without any issues, is it worth adding this to your documentation?

ngcp-network --set-interface=eth0 --ip=auto --netmask=auto
ngcp-network --move-from=lo --move-to=eth0 --type=web_ext --type=sip_ext
--type=rtp_ext --  -type=ssh_ext
Ngcpfg apply

Many thanks for your quick response anyway.

Best regards,

Ahmed

Ahmed Murad | SiPalto Ltd. 
Tel: 0207 148 7525? |? Fax: 0207 148? 7526? |? Email: am at sipalto.com
6-16 Arbutus St, London, E8 4DT | Website: www.sipalto.com
?



-----Original Message-----
From: spce-user-bounces at lists.sipwise.com
[mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of
spce-user-request at lists.sipwise.com
Sent: 23 March 2013 20:45
To: spce-user at lists.sipwise.com
Subject: Spce-user Digest, Vol 28, Issue 52

Send Spce-user mailing list submissions to
	spce-user at lists.sipwise.com

To subscribe or unsubscribe via the World Wide Web, visit
	http://lists.sipwise.com/listinfo/spce-user
or, via email, send a message with subject or body 'help' to
	spce-user-request at lists.sipwise.com

You can reach the person managing the list at
	spce-user-owner at lists.sipwise.com

When replying, please edit your Subject line so it is more specific than
"Re: Contents of Spce-user digest..."


Today's Topics:

   1. Re: Why does it take 35sec to register (Jon Bonilla (Manwe))
   2. Re: Sipwise over openvpn (Jon Bonilla (Manwe))
   3. Re: Connect softphone to SIPWISE (Jon Bonilla (Manwe))
   4. Re: routing calls between peers (Jon Bonilla (Manwe))
   5. Re: Invoicing and Billing Peers/Trunks (Jon Bonilla (Manwe))
   6. 483 Too Many Hops (Marc Rys)
   7. Re: 483 Too Many Hops (Jon Bonilla (Manwe))


----------------------------------------------------------------------

Message: 1
Date: Sat, 23 Mar 2013 20:21:10 +0100
From: Jon Bonilla (Manwe) <jbonilla at sipwise.com>
Subject: Re: [Spce-user] Why does it take 35sec to register
To: spce-user at lists.sipwise.com
Message-ID: <20130323202110.6f71ae8d at quenya>
Content-Type: text/plain; charset="utf-8"

El Sat, 23 Mar 2013 20:12:39 +0100
"Oliver Vermeulen" <oliver at oliverv.com> escribi?:

> Hello All,
> 
>  
> 
> I tested sipwise yesterday.. install was successful very easy.
> 
>  
> 
> Does anybody have problems with sip registration ?
> 
> Why does it take 35sec to register.
> 
>  
> 
> Also when it's registered and I dial takes long time to process call.
> 
> 

Please make a ngrep-sip capture of a registration and another one of a call.
We will be able to see how long takes.


-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130323/afee13bc/
attachment-0001.pgp>

------------------------------

Message: 2
Date: Sat, 23 Mar 2013 20:24:46 +0100
From: Jon Bonilla (Manwe) <jbonilla at sipwise.com>
Subject: Re: [Spce-user] Sipwise over openvpn
To: spce-user at lists.sipwise.com
Message-ID: <20130323202446.1bc5eeed at quenya>
Content-Type: text/plain; charset="utf-8"

El Sat, 23 Mar 2013 20:03:01 +0100
"Developing Tomorrow" <nh.developer at gmail.com> escribi?:

> Hi.,
> 
>  
> 
> I have "successfully" installed sipwise and am able to register users 
> and make calls using any "normal" providers.
> 
> Now
> 
> In the same box, I have created a tunnel to another network that has a 
> gateway.
> 
> I am able to ping that gateway from the station that has sipwise also 
> able to send traffic to the gateway, but I will get no answer because 
> of the ip used.
> 
> Let me put down the layout:
> 
> All the following ips are fake but will clearly show the layout.
> 
>  
> 
> Sipwise external ip: 200.200.200.200
> 
> Sipwise internal ip: 10.0.0.1
> 
>  
> 
> Tunnel router: 10.0.0.10
> 
> Gateway across tunel: 192.168.10.10
> 
>  
> 
> If I use a standard asterisk box, give it ip 10.0.1 and send calls to 
> gateway, the voice will come back thru tunnel because the gateway is 
> replying to 10.0.0.1
> 
>  
> 
> In my sipwise case, it does not work as sipwise is sending the call 
> with ip
> 200.200.200.200 so the gateway is answering thru the normal internet 
> and not the tunnel.
> 
>  
> 
> How can I configure sipwise to send info to that gateway using ip in 
> the form ttttttttt at 10.0.0.1 <mailto:ttttttttt at 10.0.0.1>  and not
> ttttttttt at 200.200.200.200 <mailto:ttttttttt at 200.200.200.200> .
> 
>  

Why don't you configure in the other side of the tunnel. "Send the traffic
to
20.200.200.200 via 10.0.0.1" ?

You can use the "extra_socket" option in the spce and that will work for you
at sip signaling level but at rtp level it won't work so you'd get signaling
via the tunel and media over internet. If that's what you want it's fine. 

Anyways I'd continue using the external main address and try to fix you
issue at networking level, with routes.


> 
> The media relay is supposed to do that function but I have no clue on 
> how to configure it.
> 
> I could only create a socket with ip 10.0.0.1 and that allows me to 
> send the traffic thru the vpn, but does not change the ip that the gateway
receives.
> 



cheers,

Jon

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130323/d6223bd4/
attachment-0001.pgp>

------------------------------

Message: 3
Date: Sat, 23 Mar 2013 20:26:05 +0100
From: Jon Bonilla (Manwe) <jbonilla at sipwise.com>
Subject: Re: [Spce-user] Connect softphone to SIPWISE
To: spce-user at lists.sipwise.com
Message-ID: <20130323202605.57ed0cdd at quenya>
Content-Type: text/plain; charset="utf-8"

El Sat, 23 Mar 2013 17:36:19 -0000
"Ahmed Murad" <am at sipalto.com> escribi?:

> Hello,
> 
>  
> 
> I have been following your guides and successfully created & connected 
> to a couple of external peers.
> 
>  
> 
> However I am having trouble connecting a softphone to our first
subscriber.
> I have created the domain, account and subscriber.
> 
>  
> 
> But I get a timeout 500 error when I try to connect the softphone to 
> the subscriber, I have ensured there are no firewall rules causing the
timeout.
> 
>  
> 
> Please could you point me in the right direction in regards to 
> troubleshooting, is there anything I've missed?
> 
>  
> 

Please do a ngre-sip capture in the systeem and let us check the
incoming/outgoing traffic.

Also, you can check /var/log/ngcp/kamaiio-proxy.log to check if REGISTER
requests are arriving and what's going on with them.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130323/8d7f84f5/
attachment-0001.pgp>

------------------------------

Message: 4
Date: Sat, 23 Mar 2013 20:40:18 +0100
From: Jon Bonilla (Manwe) <jbonilla at sipwise.com>
Subject: Re: [Spce-user] routing calls between peers
To: spce-user at lists.sipwise.com
Message-ID: <20130323204018.49f4428a at quenya>
Content-Type: text/plain; charset="utf-8"

El Sat, 23 Mar 2013 17:32:11 -0000
"Ahmed Murad" <am at sipalto.com> escribi?:

> Hello,
> 
>  
> 
> I am new to SIPWISE, do you have any guides on how to route calls from 
> 1 peer to another peer via the SIPWISE box?
> 
>  
> 
> Any guidance would be greatly appreciated.
> 
>  
> 
> Best regards,
> 
>  
>

Hi Ahmed

By default, the spce does not allow peer-peer calls. It only allows
peer-subscriber, subscriber-subscriber and subscriber-peer.

But you can indeed send calls from one peer to another. There are two ways.
Let me explain them and their differences:


- You have the option "force_outbound_calls_to_peer" you can set in the
proxy
  preferences in the admin interface. If this option is set, any incoming
call
  coming from this peer will be routed to pstn (to another or the same peer)
  based on the peer rules. This will be done even if the callee is a local
  subscriber.


- There's another option in config.yml which is called "allow_peer_relay".
  This is global and will work like this: When a call comes from a peer it
  will terminate the call locally if the calee is local or will send the
call
  to the pstn following the peer rules if not.


The first method has the advantage of being specific for a single peer and
you can enable it per peer. I use it for sbc scenarios or pure peer-peer
scenarios, where you don't have local subscriber or those are "sbc
subscribers with upper registration". 

The second one is global, it will be enabled for all you peers. The
advantage is that it won't send out you local calls. I use this for
migrations, where you still need to send the traffic of legacy class5
systems via the new Sipwise Class5.


In case you want to sell minutes to other peers and change them for it, the
above options have a problem: The rating engine is not prepared atm to bill
and charge peering servers. You'll get the cdrs and you will need to use an
external billing system for the charges. If this is your case you still have
a third option:

- Create your pstn gateways as peers. But other providers, you clients, the
  ones you want to charge create them as *subscribers*. Creating them as
  subscribers will allow you to rate them, apply ncos, prepaid (in case of
  pro) and lots of features not present in peer options. You just need to
  create a permanent contact for them, set them as trusted form the ip
  addresses they use and allow all the clis (if you want).


Please let me know which scenario you're looking for and if any of these
three options match it.

cheers,

Jon

 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130323/2ea74561/
attachment-0001.pgp>

------------------------------

Message: 5
Date: Sat, 23 Mar 2013 20:44:02 +0100
From: Jon Bonilla (Manwe) <jbonilla at sipwise.com>
Subject: Re: [Spce-user] Invoicing and Billing Peers/Trunks
To: spce-user at lists.sipwise.com
Message-ID: <20130323204402.4eeaf025 at quenya>
Content-Type: text/plain; charset="utf-8"

El Fri, 22 Mar 2013 15:16:19 +0200
Deon Vermeulen <vermeulen.deon at gmail.com> escribi?:

> Good Day
> 
> Is there any possibility that one could generate an Invoice to 
> customers with a Trunk to SPCE 2.7?


You can get the cdrs, but not an invoice as PDF is you mean that. Any
customer, has to be created as subscriber and it will be rated if you don't
disable it.

> 
> Is there any possibility to Generate a Bill to a Peering Partner for 
> calls sent to SPCE 2.7. i.e Reverse billing?
> 

Not at the moment. Peers can't be charged in this version of the spce.


cheers,

Jon
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130323/d3643d13/
attachment-0001.pgp>

------------------------------

Message: 6
Date: Sat, 23 Mar 2013 15:21:46 -0500 (CDT)
From: Marc Rys <m.rys at tri-lakes.net>
Subject: [Spce-user] 483 Too Many Hops
To: spce-user at lists.sipwise.com
Message-ID:
	
<c6ac07e3-43b0-4127-8a7d-8f562e0eef4c at usbsnpzimbra01v.servers.rystec.com>
	
Content-Type: text/plain; charset="utf-8"

I've been evaluating SPCE over the last week.  I've already setup a couple
test subscribers and setup a peer with a provider we work with for SIP LD
Term.  All of those test have worked very successful.

We also have our own media gateway which is interconnected with the local
PSTN via TDM trunk, but I send incoming calls from the PSTN through our
MediaGateway to SPCE, SPCE is responding back with 483 "Too Many Hops".
Attached is the wireshark cap.

Any help is appreciated.

Marc Rys
http://www.tri-lakes.net
http://www.rystec.com
-------------- next part --------------
A non-text attachment was scrubbed...
Name: SPCE-INBOUND-Taqua-483TooManyHops.pcap
Type: application/octet-stream
Size: 2220 bytes
Desc: not available
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130323/3e72b68f/
attachment-0001.obj>

------------------------------

Message: 7
Date: Sat, 23 Mar 2013 21:45:10 +0100
From: Jon Bonilla (Manwe) <jbonilla at sipwise.com>
Subject: Re: [Spce-user] 483 Too Many Hops
To: spce-user at lists.sipwise.com
Message-ID: <20130323214510.5498f5dc at quenya>
Content-Type: text/plain; charset="utf-8"

El Sat, 23 Mar 2013 15:21:46 -0500 (CDT) Marc Rys <m.rys at tri-lakes.net>
escribi?:

> En-Respuesta-A: <20130323204402.4eeaf025 at quenya>


Please don't answer a mail to create a new thread. Use the "new mail" button
instead. 


Cheers,

Jon

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130323/df575ddc/
attachment.pgp>

------------------------------

_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com
http://lists.sipwise.com/listinfo/spce-user


End of Spce-user Digest, Vol 28, Issue 52
*****************************************




From jbonilla at sipwise.com  Mon Mar 25 07:41:46 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Mon, 25 Mar 2013 12:41:46 +0100
Subject: [Spce-user] Connect softphone to SIPWISE (Jon Bonilla (Manwe))
In-Reply-To: <058901ce294a$9919d9d0$cb4d8d70$@sipalto.com>
References: <058901ce294a$9919d9d0$cb4d8d70$@sipalto.com>
Message-ID: <20130325124146.7c2f0804@quenya>

El Mon, 25 Mar 2013 11:19:20 -0000
"Ahmed Murad" <am at sipalto.com> escribi?:

> Hi,
> 
> Thanks for the info. I received help on connecting a UA to SIPWISE from
> another user who advised to run the below commands, after running them UA's
> connected without any issues, is it worth adding this to your documentation?
> 
> ngcp-network --set-interface=eth0 --ip=auto --netmask=auto
> ngcp-network --move-from=lo --move-to=eth0 --type=web_ext --type=sip_ext
> --type=rtp_ext --  -type=ssh_ext
> Ngcpfg apply
> 
> 


Those commands are already present in our documentation.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130325/9d2ba0e7/attachment.asc>

From am at sipalto.com  Mon Mar 25 07:45:23 2013
From: am at sipalto.com (Ahmed Murad)
Date: Mon, 25 Mar 2013 11:45:23 -0000
Subject: [Spce-user] routing calls between peers (Jon Bonilla (Manwe))
Message-ID: <058a01ce294e$3cd0fc40$b672f4c0$@sipalto.com>

Hi Jon,

Many thanks for this info. For peering option 1 appears to be the one we
need. Billing is not an issue in this scenario and this spce is being used
as an SBC. We have 4 peers.

Peer 1 (inbound only), Peer 2(inbound only), Peer 3 & Peer 4.

Peer 1 sends inbound calls to the spce, we want to relay all Peer 1 calls to
Peer 4 which will then handle the call.
Peer 2 also sends inbound calls to the spce, we want to relay all calls
again to Peer 4.
Peer 3 and Peer 4 are our outbound Peers. Which subscriber calls will be
routed (inbound & outbound).

I can see the option ' force_outbound_calls_to_peer', how do I go about
setting the peer I want to force these calls to go to?

Again many thanks for your quick response, highly appreciated.

Best regards,

Ahmed

Ahmed Murad | SiPalto Ltd. 
Tel: 0207 148 7525? |? Fax: 0207 148? 7526? |? Email: am at sipalto.com
6-16 Arbutus St, London, E8 4DT | Website: www.sipalto.com
?



-----Original Message-----
From: spce-user-bounces at lists.sipwise.com
[mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of
spce-user-request at lists.sipwise.com
Sent: 23 March 2013 20:45
To: spce-user at lists.sipwise.com
Subject: Spce-user Digest, Vol 28, Issue 52

Send Spce-user mailing list submissions to
	spce-user at lists.sipwise.com

To subscribe or unsubscribe via the World Wide Web, visit
	http://lists.sipwise.com/listinfo/spce-user
or, via email, send a message with subject or body 'help' to
	spce-user-request at lists.sipwise.com

You can reach the person managing the list at
	spce-user-owner at lists.sipwise.com

When replying, please edit your Subject line so it is more specific than
"Re: Contents of Spce-user digest..."


Today's Topics:

   1. Re: Why does it take 35sec to register (Jon Bonilla (Manwe))
   2. Re: Sipwise over openvpn (Jon Bonilla (Manwe))
   3. Re: Connect softphone to SIPWISE (Jon Bonilla (Manwe))
   4. Re: routing calls between peers (Jon Bonilla (Manwe))
   5. Re: Invoicing and Billing Peers/Trunks (Jon Bonilla (Manwe))
   6. 483 Too Many Hops (Marc Rys)
   7. Re: 483 Too Many Hops (Jon Bonilla (Manwe))


----------------------------------------------------------------------

Message: 1
Date: Sat, 23 Mar 2013 20:21:10 +0100
From: Jon Bonilla (Manwe) <jbonilla at sipwise.com>
Subject: Re: [Spce-user] Why does it take 35sec to register
To: spce-user at lists.sipwise.com
Message-ID: <20130323202110.6f71ae8d at quenya>
Content-Type: text/plain; charset="utf-8"

El Sat, 23 Mar 2013 20:12:39 +0100
"Oliver Vermeulen" <oliver at oliverv.com> escribi?:

> Hello All,
> 
>  
> 
> I tested sipwise yesterday.. install was successful very easy.
> 
>  
> 
> Does anybody have problems with sip registration ?
> 
> Why does it take 35sec to register.
> 
>  
> 
> Also when it's registered and I dial takes long time to process call.
> 
> 

Please make a ngrep-sip capture of a registration and another one of a call.
We will be able to see how long takes.


-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130323/afee13bc/
attachment-0001.pgp>

------------------------------

Message: 2
Date: Sat, 23 Mar 2013 20:24:46 +0100
From: Jon Bonilla (Manwe) <jbonilla at sipwise.com>
Subject: Re: [Spce-user] Sipwise over openvpn
To: spce-user at lists.sipwise.com
Message-ID: <20130323202446.1bc5eeed at quenya>
Content-Type: text/plain; charset="utf-8"

El Sat, 23 Mar 2013 20:03:01 +0100
"Developing Tomorrow" <nh.developer at gmail.com> escribi?:

> Hi.,
> 
>  
> 
> I have "successfully" installed sipwise and am able to register users 
> and make calls using any "normal" providers.
> 
> Now
> 
> In the same box, I have created a tunnel to another network that has a 
> gateway.
> 
> I am able to ping that gateway from the station that has sipwise also 
> able to send traffic to the gateway, but I will get no answer because 
> of the ip used.
> 
> Let me put down the layout:
> 
> All the following ips are fake but will clearly show the layout.
> 
>  
> 
> Sipwise external ip: 200.200.200.200
> 
> Sipwise internal ip: 10.0.0.1
> 
>  
> 
> Tunnel router: 10.0.0.10
> 
> Gateway across tunel: 192.168.10.10
> 
>  
> 
> If I use a standard asterisk box, give it ip 10.0.1 and send calls to 
> gateway, the voice will come back thru tunnel because the gateway is 
> replying to 10.0.0.1
> 
>  
> 
> In my sipwise case, it does not work as sipwise is sending the call 
> with ip
> 200.200.200.200 so the gateway is answering thru the normal internet 
> and not the tunnel.
> 
>  
> 
> How can I configure sipwise to send info to that gateway using ip in 
> the form ttttttttt at 10.0.0.1 <mailto:ttttttttt at 10.0.0.1>  and not
> ttttttttt at 200.200.200.200 <mailto:ttttttttt at 200.200.200.200> .
> 
>  

Why don't you configure in the other side of the tunnel. "Send the traffic
to
20.200.200.200 via 10.0.0.1" ?

You can use the "extra_socket" option in the spce and that will work for you
at sip signaling level but at rtp level it won't work so you'd get signaling
via the tunel and media over internet. If that's what you want it's fine. 

Anyways I'd continue using the external main address and try to fix you
issue at networking level, with routes.


> 
> The media relay is supposed to do that function but I have no clue on 
> how to configure it.
> 
> I could only create a socket with ip 10.0.0.1 and that allows me to 
> send the traffic thru the vpn, but does not change the ip that the gateway
receives.
> 



cheers,

Jon

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130323/d6223bd4/
attachment-0001.pgp>

------------------------------

Message: 3
Date: Sat, 23 Mar 2013 20:26:05 +0100
From: Jon Bonilla (Manwe) <jbonilla at sipwise.com>
Subject: Re: [Spce-user] Connect softphone to SIPWISE
To: spce-user at lists.sipwise.com
Message-ID: <20130323202605.57ed0cdd at quenya>
Content-Type: text/plain; charset="utf-8"

El Sat, 23 Mar 2013 17:36:19 -0000
"Ahmed Murad" <am at sipalto.com> escribi?:

> Hello,
> 
>  
> 
> I have been following your guides and successfully created & connected 
> to a couple of external peers.
> 
>  
> 
> However I am having trouble connecting a softphone to our first
subscriber.
> I have created the domain, account and subscriber.
> 
>  
> 
> But I get a timeout 500 error when I try to connect the softphone to 
> the subscriber, I have ensured there are no firewall rules causing the
timeout.
> 
>  
> 
> Please could you point me in the right direction in regards to 
> troubleshooting, is there anything I've missed?
> 
>  
> 

Please do a ngre-sip capture in the systeem and let us check the
incoming/outgoing traffic.

Also, you can check /var/log/ngcp/kamaiio-proxy.log to check if REGISTER
requests are arriving and what's going on with them.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130323/8d7f84f5/
attachment-0001.pgp>

------------------------------

Message: 4
Date: Sat, 23 Mar 2013 20:40:18 +0100
From: Jon Bonilla (Manwe) <jbonilla at sipwise.com>
Subject: Re: [Spce-user] routing calls between peers
To: spce-user at lists.sipwise.com
Message-ID: <20130323204018.49f4428a at quenya>
Content-Type: text/plain; charset="utf-8"

El Sat, 23 Mar 2013 17:32:11 -0000
"Ahmed Murad" <am at sipalto.com> escribi?:

> Hello,
> 
>  
> 
> I am new to SIPWISE, do you have any guides on how to route calls from 
> 1 peer to another peer via the SIPWISE box?
> 
>  
> 
> Any guidance would be greatly appreciated.
> 
>  
> 
> Best regards,
> 
>  
>

Hi Ahmed

By default, the spce does not allow peer-peer calls. It only allows
peer-subscriber, subscriber-subscriber and subscriber-peer.

But you can indeed send calls from one peer to another. There are two ways.
Let me explain them and their differences:


- You have the option "force_outbound_calls_to_peer" you can set in the
proxy
  preferences in the admin interface. If this option is set, any incoming
call
  coming from this peer will be routed to pstn (to another or the same peer)
  based on the peer rules. This will be done even if the callee is a local
  subscriber.


- There's another option in config.yml which is called "allow_peer_relay".
  This is global and will work like this: When a call comes from a peer it
  will terminate the call locally if the calee is local or will send the
call
  to the pstn following the peer rules if not.


The first method has the advantage of being specific for a single peer and
you can enable it per peer. I use it for sbc scenarios or pure peer-peer
scenarios, where you don't have local subscriber or those are "sbc
subscribers with upper registration". 

The second one is global, it will be enabled for all you peers. The
advantage is that it won't send out you local calls. I use this for
migrations, where you still need to send the traffic of legacy class5
systems via the new Sipwise Class5.


In case you want to sell minutes to other peers and change them for it, the
above options have a problem: The rating engine is not prepared atm to bill
and charge peering servers. You'll get the cdrs and you will need to use an
external billing system for the charges. If this is your case you still have
a third option:

- Create your pstn gateways as peers. But other providers, you clients, the
  ones you want to charge create them as *subscribers*. Creating them as
  subscribers will allow you to rate them, apply ncos, prepaid (in case of
  pro) and lots of features not present in peer options. You just need to
  create a permanent contact for them, set them as trusted form the ip
  addresses they use and allow all the clis (if you want).


Please let me know which scenario you're looking for and if any of these
three options match it.

cheers,

Jon

 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130323/2ea74561/
attachment-0001.pgp>

------------------------------

Message: 5
Date: Sat, 23 Mar 2013 20:44:02 +0100
From: Jon Bonilla (Manwe) <jbonilla at sipwise.com>
Subject: Re: [Spce-user] Invoicing and Billing Peers/Trunks
To: spce-user at lists.sipwise.com
Message-ID: <20130323204402.4eeaf025 at quenya>
Content-Type: text/plain; charset="utf-8"

El Fri, 22 Mar 2013 15:16:19 +0200
Deon Vermeulen <vermeulen.deon at gmail.com> escribi?:

> Good Day
> 
> Is there any possibility that one could generate an Invoice to 
> customers with a Trunk to SPCE 2.7?


You can get the cdrs, but not an invoice as PDF is you mean that. Any
customer, has to be created as subscriber and it will be rated if you don't
disable it.

> 
> Is there any possibility to Generate a Bill to a Peering Partner for 
> calls sent to SPCE 2.7. i.e Reverse billing?
> 

Not at the moment. Peers can't be charged in this version of the spce.


cheers,

Jon
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130323/d3643d13/
attachment-0001.pgp>

------------------------------

Message: 6
Date: Sat, 23 Mar 2013 15:21:46 -0500 (CDT)
From: Marc Rys <m.rys at tri-lakes.net>
Subject: [Spce-user] 483 Too Many Hops
To: spce-user at lists.sipwise.com
Message-ID:
	
<c6ac07e3-43b0-4127-8a7d-8f562e0eef4c at usbsnpzimbra01v.servers.rystec.com>
	
Content-Type: text/plain; charset="utf-8"

I've been evaluating SPCE over the last week.  I've already setup a couple
test subscribers and setup a peer with a provider we work with for SIP LD
Term.  All of those test have worked very successful.

We also have our own media gateway which is interconnected with the local
PSTN via TDM trunk, but I send incoming calls from the PSTN through our
MediaGateway to SPCE, SPCE is responding back with 483 "Too Many Hops".
Attached is the wireshark cap.

Any help is appreciated.

Marc Rys
http://www.tri-lakes.net
http://www.rystec.com
-------------- next part --------------
A non-text attachment was scrubbed...
Name: SPCE-INBOUND-Taqua-483TooManyHops.pcap
Type: application/octet-stream
Size: 2220 bytes
Desc: not available
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130323/3e72b68f/
attachment-0001.obj>

------------------------------

Message: 7
Date: Sat, 23 Mar 2013 21:45:10 +0100
From: Jon Bonilla (Manwe) <jbonilla at sipwise.com>
Subject: Re: [Spce-user] 483 Too Many Hops
To: spce-user at lists.sipwise.com
Message-ID: <20130323214510.5498f5dc at quenya>
Content-Type: text/plain; charset="utf-8"

El Sat, 23 Mar 2013 15:21:46 -0500 (CDT) Marc Rys <m.rys at tri-lakes.net>
escribi?:

> En-Respuesta-A: <20130323204402.4eeaf025 at quenya>


Please don't answer a mail to create a new thread. Use the "new mail" button
instead. 


Cheers,

Jon

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130323/df575ddc/
attachment.pgp>

------------------------------

_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com
http://lists.sipwise.com/listinfo/spce-user


End of Spce-user Digest, Vol 28, Issue 52
*****************************************




From jbonilla at sipwise.com  Mon Mar 25 07:56:49 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Mon, 25 Mar 2013 12:56:49 +0100
Subject: [Spce-user] routing calls between peers (Jon Bonilla (Manwe))
In-Reply-To: <058a01ce294e$3cd0fc40$b672f4c0$@sipalto.com>
References: <058a01ce294e$3cd0fc40$b672f4c0$@sipalto.com>
Message-ID: <20130325125649.5e328acf@quenya>

El Mon, 25 Mar 2013 11:45:23 -0000
"Ahmed Murad" <am at sipalto.com> escribi?:

> Hi Jon,
> 
> Many thanks for this info. For peering option 1 appears to be the one we
> need. Billing is not an issue in this scenario and this spce is being used
> as an SBC. We have 4 peers.
> 
> Peer 1 (inbound only), Peer 2(inbound only), Peer 3 & Peer 4.
> 
> Peer 1 sends inbound calls to the spce, we want to relay all Peer 1 calls to
> Peer 4 which will then handle the call.
> Peer 2 also sends inbound calls to the spce, we want to relay all calls
> again to Peer 4.
> Peer 3 and Peer 4 are our outbound Peers. Which subscriber calls will be
> routed (inbound & outbound).
> 
> I can see the option ' force_outbound_calls_to_peer', how do I go about
> setting the peer I want to force these calls to go to?
> 
> Again many thanks for your quick response, highly appreciated.
> 
>

If a call comes from a subscriber which has upper registration (sbc mode) the
call will go to the upper registration peer.

For any other call, peering rules will be used. 

In this particular case, for
calls coming form peers 1 and 2 I guess they will always have the same domain
(domain1 and domain2). If you set a peering rule in peer_group4 which matches
all the destinations but only domain1 and domain2 from domains you'll get that.
It's just using the peering rules, nothing out of the ordinary.


Sample:
Caller_Pattern: ^.*@domain1\.com$

cheers,

Jon
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130325/b0eb295e/attachment.asc>

From sn at fest.dk  Mon Mar 25 08:17:39 2013
From: sn at fest.dk (Sylvester Nielsen)
Date: Mon, 25 Mar 2013 12:17:39 +0000
Subject: [Spce-user] Incoming calls from provider does not work
Message-ID: <E50575DD48D9B743925AC3BCE6310A4B243687F2@DB3PRD0710MB355.eurprd07.prod.outlook.com>

Hi,

I just installed and configured a new server. I have setup the routing to my provider.
Outbound call works as expected but inbound does not work. The trunk is without registration.

At first I got a message that the domain wasen't supported. I added to IP to the domain list, but then the server send a 401 Unauthorized message.
I tried to trace the kamailio logs and see this in the proxy log.
New request - M=INVITE R=sip:+4578788471 at 87.116.27.246 F=sip:+4522771051 at 212.98.66.168;user=phone T=sip:+4578788471 at 87.116.27.246 IP=212.98.66.168:11054 (127.0.0.1:5060) ID=BW131042756250313-108955888 at 212.98.65.15<mailto:ID=BW131042756250313-108955888 at 212.98.65.15>
Authentication failed, no credentials - R=sip:+4578788471 at 87.116.27.246 ID=BW131042756250313-108955888 at 212.98.65.15

The invite I get from the peer look like this:

T 212.98.66.168:11054 -> 87.116.27.246:5060 [AP]
INVITE sip:+4578788471 at 87.116.27.246 SIP/2.0.
Via: SIP/2.0/TCP 212.98.66.168:5060;branch=z9hG4bKvea1ve006o0hvpov25t0.1.
From: "+4522771051"<sip:+4522771051 at 212.98.66.168;user=phone>;tag=659644293-1364213720121-.
To: "4578788471 4578788471"<sip:+4578788471 at 87.116.27.246>.
Call-ID: BW131520121250313-842633845 at 212.98.65.15.
CSeq: 280801821 INVITE.
Contact: <sip:+4522771051 at 212.98.66.168:5060;transport=tcp>.
P-Asserted-Identity: "+4522771051"<sip:+4522771051 at 212.98.65.15;user=phone>.
Privacy: none.
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE.
Accept: application/dtmf,application/dtmf-relay,application/media_control+xml,application/sdp,multipart/mixed.
Supported: .
Max-Forwards: 29.
Content-Type: application/sdp.
Content-Length: 348.
.
v=0.
o=BroadWorks 5699654 1 IN IP4 212.98.66.168.
s=-.
c=IN IP4 212.98.66.168.
t=0 0.
m=audio 23772 RTP/AVP 8 0 18 99 100.
a=rtpmap:99 telephone-event/8000.
a=rtpmap:100 X-NSE/8000.
a=fmtp:99 0-15.
a=fmtp:100 200-202.
a=X-sqn:0.
a=X-cap:1 audio RTP/AVP 100.
a=X-cpar:a=rtpmap:100 X-NSE/8000.
a=X-cpar:a=fmtp:100 200-202.
a=X-cap:2 image udptl t38.

Med venlig hilsen

   Sylvester Nielsen

---------------------------------------------------------------

Fest Data A/S
Generatorvej 8D, 1.th
2730 Herlev

e-mail : sn at fest.dk<mailto:sn at fest.dk>

Tlf:     +45 4582 7389
Fax:   +45 4582 7387
Dir:    +45 6996 2011

http://www.festdata.dk

[Beskrivelse: Description: Description: Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse: cid:image003.jpg at 01CBCE06.BAABD4A0]




-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130325/e92f6b42/attachment.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.jpg
Type: image/jpeg
Size: 11834 bytes
Desc: image001.jpg
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130325/e92f6b42/attachment.jpg>

From apogrebennyk at sipwise.com  Mon Mar 25 08:55:14 2013
From: apogrebennyk at sipwise.com (Andrew Pogrebennyk)
Date: Mon, 25 Mar 2013 13:55:14 +0100
Subject: [Spce-user] Incoming calls from provider does not work
In-Reply-To: <E50575DD48D9B743925AC3BCE6310A4B243687F2@DB3PRD0710MB355.eurprd07.prod.outlook.com>
References: <E50575DD48D9B743925AC3BCE6310A4B243687F2@DB3PRD0710MB355.eurprd07.prod.outlook.com>
Message-ID: <51504932.8030108@sipwise.com>

Hello Sylvester,
it could be that peer is using different IP addresses for origination
and termination. The calls are originated from this IP address:
212.98.66.168. Do you have a peer with this IP address?

On 03/25/2013 01:17 PM, Sylvester Nielsen wrote:
> Hi,
> 
>  
> 
> I just installed and configured a new server. I have setup the routing
> to my provider.
> 
> Outbound call works as expected but inbound does not work. The trunk is
> without registration.
> 
>  
> 
> At first I got a message that the domain wasen?t supported. I added to
> IP to the domain list, but then the server send a 401 Unauthorized message.
> 
> I tried to trace the kamailio logs and see this in the proxy log.
> 
> New request - M=INVITE R=sip:+4578788471 at 87.116.27.246
> F=sip:+4522771051 at 212.98.66.168;user=phone
> T=sip:+4578788471 at 87.116.27.246 IP=212.98.66.168:11054 (127.0.0.1:5060)
> ID=BW131042756250313-108955888 at 212.98.65.15
> <mailto:ID=BW131042756250313-108955888 at 212.98.65.15>
> 
> Authentication failed, no credentials - R=sip:+4578788471 at 87.116.27.246
> ID=BW131042756250313-108955888 at 212.98.65.15
> 
>  
> 
> The invite I get from the peer look like this:
> 
>  
> 
> T 212.98.66.168:11054 -> 87.116.27.246:5060 [AP]
> 
> INVITE sip:+4578788471 at 87.116.27.246 SIP/2.0.
> 
> Via: SIP/2.0/TCP 212.98.66.168:5060;branch=z9hG4bKvea1ve006o0hvpov25t0.1.
> 
> From:
> "+4522771051"<sip:+4522771051 at 212.98.66.168;user=phone>;tag=659644293-1364213720121-.
> 
> To: "4578788471 4578788471"<sip:+4578788471 at 87.116.27.246>.
> 
> Call-ID: BW131520121250313-842633845 at 212.98.65.15.
> 
> CSeq: 280801821 INVITE.
> 
> Contact: <sip:+4522771051 at 212.98.66.168:5060;transport=tcp>.
> 
> P-Asserted-Identity: "+4522771051"<sip:+4522771051 at 212.98.65.15;user=phone>.
> 
> Privacy: none.
> 
> Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE.
> 
> Accept:
> application/dtmf,application/dtmf-relay,application/media_control+xml,application/sdp,multipart/mixed.
> 
> Supported: .
> 
> Max-Forwards: 29.
> 
> Content-Type: application/sdp.
> 
> Content-Length: 348.
> 
> .
> 
> v=0.
> 
> o=BroadWorks 5699654 1 IN IP4 212.98.66.168.
> 
> s=-.
> 
> c=IN IP4 212.98.66.168.
> 
> t=0 0.
> 
> m=audio 23772 RTP/AVP 8 0 18 99 100.
> 
> a=rtpmap:99 telephone-event/8000.
> 
> a=rtpmap:100 X-NSE/8000.
> 
> a=fmtp:99 0-15.
> 
> a=fmtp:100 200-202.
> 
> a=X-sqn:0.
> 
> a=X-cap:1 audio RTP/AVP 100.
> 
> a=X-cpar:a=rtpmap:100 X-NSE/8000.
> 
> a=X-cpar:a=fmtp:100 200-202.
> 
> a=X-cap:2 image udptl t38.
> 
>  
> 
> Med venlig hilsen
> 
>  
> 
>    Sylvester Nielsen
> 
>  
> 
> ---------------------------------------------------------------
> 
> 	
> 
> Fest Data A/S
> Generatorvej 8D, 1.th
> 2730 Herlev
> 
> e-mail : sn at fest.dk <mailto:sn at fest.dk>
> 
> 	
> 
> Tlf:     +45 4582 7389
> Fax:   +45 4582 7387
> Dir:    +45 6996 2011
> 
> http://www.festdata.dk
> 
> 	
> 
> Beskrivelse: Description: Description: Beskrivelse: Beskrivelse:
> Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse:
> cid:image003.jpg at 01CBCE06.BAABD4A0
> 
> 		
> 
>  
> 
>  
> 
> 
> 
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
> 




From m.rys at tri-lakes.net  Mon Mar 25 20:09:49 2013
From: m.rys at tri-lakes.net (Marc Rys)
Date: Mon, 25 Mar 2013 19:09:49 -0500 (CDT)
Subject: [Spce-user] 483 Too Many Hops
In-Reply-To: <514F2D26.3030103@sipwise.com>
Message-ID: <ef498e89-2477-4559-bd77-a272859a326e@usbsnpzimbra01v.servers.rystec.com>

Hi Everybody,  Well I have news.. I switched my test phone from a grandstream to a yeahlink and now i can receive calls from my gateway..  Hooray!!!

Now the only problem I see is that when I try to send a call from my UA to my gateway i get a 403 forbidden from my gateway.  Not sure what its seeing and not liking.  Perhaps is this the same problem the AudioCodes have?

Attached is the pcap.

I truely appreciate all your help.

Marc Rys
http://www.tri-lakes.net
http://www.rystec.com



----- Original Message -----
From: "Andreas Granig" <agranig at sipwise.com>
To: spce-user at lists.sipwise.com
Sent: Sunday, March 24, 2013 11:43:18 AM
Subject: Re: [Spce-user] 483 Too Many Hops

Hi,

On 03/24/2013 05:25 PM, Lorenzo Mangani wrote:
> As you can see from your trace, the call is never completely
> established, as the 200 OK's ACK never makes it back to the user-agent.
> After a brief look this seems to be caused by the gateway always using
> the contact header to respond instead of the from header, as for the BYE
> message also being rejected for the same reason - just my 2 cents, might
> be wrong - maybe someone else on the list will have different input,
>

The Contact handling is as expected, but the 208.73.198.12 doesn't 
convert the Record-Route headers added by the SPCE in the 200-OK (packet 
#14) into Route headers of the ACK (packet #15), so the proxy has no 
information where to relay the ACK to (note that ACKs to 200 are relayed 
end-to-end, so all Route information must be preserved).

The sender of the ACK *must* take the Record-Route headers of the 
previous 200-OK, convert them to Routes and add them to the message in 
reverse order, and send the request to the URI found in the top-most Route.

Andreas

_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com
http://lists.sipwise.com/listinfo/spce-user
-------------- next part --------------
A non-text attachment was scrubbed...
Name: SPCE-OUTBOUND-Taqua-403Forbidden2.pcap
Type: application/octet-stream
Size: 10137 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130325/0cf0e1ce/attachment.pcap>

From m.rys at tri-lakes.net  Mon Mar 25 20:15:06 2013
From: m.rys at tri-lakes.net (Marc Rys)
Date: Mon, 25 Mar 2013 19:15:06 -0500 (CDT)
Subject: [Spce-user] 483 Too Many Hops
In-Reply-To: <514F2D26.3030103@sipwise.com>
Message-ID: <eeb0a182-2a8c-4330-b200-a1bdd1b873cd@usbsnpzimbra01v.servers.rystec.com>

Hi Everybody,  Well I have news.. I switched my test phone from a grandstream to a yeahlink and now i can receive calls from my gateway..  Hooray!!!

Now the only problem I see is that when I try to send a call from my UA to my gateway i get a 403 forbidden from my gateway.
Attached is the pcap.

I truely appreciate all your help.

Marc Rys
http://www.tri-lakes.net
http://www.rystec.com



----- Original Message -----
From: "Andreas Granig" <agranig at sipwise.com>
To: spce-user at lists.sipwise.com
Sent: Sunday, March 24, 2013 11:43:18 AM
Subject: Re: [Spce-user] 483 Too Many Hops

Hi,

On 03/24/2013 05:25 PM, Lorenzo Mangani wrote:
> As you can see from your trace, the call is never completely
> established, as the 200 OK's ACK never makes it back to the user-agent.
> After a brief look this seems to be caused by the gateway always using
> the contact header to respond instead of the from header, as for the BYE
> message also being rejected for the same reason - just my 2 cents, might
> be wrong - maybe someone else on the list will have different input,
>

The Contact handling is as expected, but the 208.73.198.12 doesn't 
convert the Record-Route headers added by the SPCE in the 200-OK (packet 
#14) into Route headers of the ACK (packet #15), so the proxy has no 
information where to relay the ACK to (note that ACKs to 200 are relayed 
end-to-end, so all Route information must be preserved).

The sender of the ACK *must* take the Record-Route headers of the 
previous 200-OK, convert them to Routes and add them to the message in 
reverse order, and send the request to the URI found in the top-most Route.

Andreas

_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com
http://lists.sipwise.com/listinfo/spce-user



From jbonilla at sipwise.com  Tue Mar 26 05:05:29 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Tue, 26 Mar 2013 10:05:29 +0100
Subject: [Spce-user] Mailing list rule changes: Daily digest
Message-ID: <20130326100529.22401a25@quenya>

Hi all! This is for our beloved community

We've been using the spce-user mailing list for free, best effort support
during the last years. It has been more successful than expected in terms of
questions, patches and quality in general. Thanks to all of you for that! We
(Sipwise) find that the mailing list archive is a useful place to find answers
to common questions or find people that had the same issues in the past. Many
bug reports have been reported here too. That's why the mailing list is our
primary support method for our community.

To preserve the mailing list archive as a resource where people can search in
the future, we try to keep track of all threads and finish them for good or
bad. In that goal, we find that using the "daily digest" option to answer the
list for a particular thread breaks the thread and breaks our workflow where
every thread should be followed and closed via answers to a single mail. We
find that the spce-user mailing list is not a high traffic list. Receiving
every mail shouldn't be painful at all. For that reason, we plan to remove the
"receive a daily digest" option form the mailing list and make users to receive
a every single mail. This change would be effective in the following days.


As side notes, and for this same goal of preserving the list clean and useful,
I'd like to add some tips regarding the mailing list usage:

- Please *do not* use the "answer" button to create a new thread. Even if you
  delete the body and the subject of the mail, the mail you send will contain a
  "in-reply-to" header and will be ordered with the old thread in many
  mail-clients.

- Please use the "answer" button to answer an open thread. Don't use the
  forward button or "new mail" or any other buttons.

- Please consider that some people will read emails in plain text and not html.
  "Find my answer in blue" is not a good idea :)

- Please, try to keep the mailing list in CC instead of only answering the
  person. That makes the mailing list archive to have the full information. If
  there's sensible data and it can be replaced by another string do so.

In general, we're not too restrictive with the rules. RFC1855, the old
"netiquette" rules are more or less valid even nowadays.

Tip: To search our mailing list you can just google for 

$KEYWORD site:http://lists.sipwise.com/pipermail/spce-user/


As always, any comments are more than welcome. This is *your* mailing list and
we try to keep it useful for you.


cheers,

Jon

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130326/aba99bef/attachment.asc>

From apogrebennyk at sipwise.com  Tue Mar 26 06:09:06 2013
From: apogrebennyk at sipwise.com (Andrew Pogrebennyk)
Date: Tue, 26 Mar 2013 11:09:06 +0100
Subject: [Spce-user] 483 Too Many Hops
In-Reply-To: <eeb0a182-2a8c-4330-b200-a1bdd1b873cd@usbsnpzimbra01v.servers.rystec.com>
References: <eeb0a182-2a8c-4330-b200-a1bdd1b873cd@usbsnpzimbra01v.servers.rystec.com>
Message-ID: <515173C2.1020506@sipwise.com>

Hi Marc,
well, the reason is outside of sip:provider. In your pcap the called
gateway/UA is sending 403 back to sip:provider. I can only guess that
maybe it needs a specific number in From or PAI or the given number
format is wrong.

On 03/26/2013 01:15 AM, Marc Rys wrote:
> Hi Everybody,  Well I have news.. I switched my test phone from a grandstream to a yeahlink and now i can receive calls from my gateway..  Hooray!!!
> 
> Now the only problem I see is that when I try to send a call from my UA to my gateway i get a 403 forbidden from my gateway.
> Attached is the pcap.
> 
> I truely appreciate all your help.
> 
> Marc Rys
> http://www.tri-lakes.net
> http://www.rystec.com




From rea1ity at kaist.ac.kr  Tue Mar 26 06:52:23 2013
From: rea1ity at kaist.ac.kr (=?UTF-8?B?7ZeI7Jyk?=)
Date: Tue, 26 Mar 2013 19:52:23 +0900 (KST)
Subject: [Spce-user] =?utf-8?q?PSTN_GW_can_register_SIPWISE_server_but_GW_?=
	=?utf-8?q?peer_status_always_busy=2E?=
Message-ID: <51517eaf3fe8_@_imoxion.com>

Hi,
I succeed register with SIP phones and PSTN Gateway on SIPWISE.

So it is working well each SIP phones it located in my private network.

But, When I tried to call telephone number as PSTN using by SIP phone, it returned busy signal as sound.

So now I trying to solve it could communicate with PSTN.

If you were me, how to find the problem that couldn't communicate with PSTN?

If you were me, where to configure in SIPWISE?

I'm also saw the log files it located /var/log/ngcp/ but still I don't know what is problem.

And please let me know what needed the file if you could check what is exactly problem of point.

David Ho
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130326/82abb898/attachment.html>

From jbonilla at sipwise.com  Tue Mar 26 07:05:43 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Tue, 26 Mar 2013 12:05:43 +0100
Subject: [Spce-user] PSTN GW can register SIPWISE server but GW peer
 status always busy.
In-Reply-To: <51517eaf3fe8_@_imoxion.com>
References: <51517eaf3fe8_@_imoxion.com>
Message-ID: <20130326120543.1d753acd@quenya>

El Tue, 26 Mar 2013 19:52:23 +0900 (KST)
??<rea1ity at kaist.ac.kr> escribi?:

> Hi,
> I succeed register with SIP phones and PSTN Gateway on SIPWISE.

Why would you rgister a pstn gw to the spce? The Peers do not register on the
spce.

> 
> So it is working well each SIP phones it located in my private network.
> 
> But, When I tried to call telephone number as PSTN using by SIP phone, it
> returned busy signal as sound.
> 

That's not informational. What's the nwtwork capture? ngrep-sip? kamailio-proxy
logs?


> So now I trying to solve it could communicate with PSTN.
> 
> If you were me, how to find the problem that couldn't communicate with PSTN?
> 
> If you were me, where to configure in SIPWISE?
> 

Please read the handbook carefully and follow the steps.


> I'm also saw the log files it located /var/log/ngcp/ but still I don't know
> what is problem.
> 
> And please let me know what needed the file if you could check what is
> exactly problem of point.
> 


cheers,

Jon
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130326/7a992dd6/attachment.asc>

From agranig at sipwise.com  Tue Mar 26 09:09:50 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Tue, 26 Mar 2013 14:09:50 +0100
Subject: [Spce-user] PSTN GW can register SIPWISE server but GW peer
 status always busy.
In-Reply-To: <51517eaf3fe8_@_imoxion.com>
References: <51517eaf3fe8_@_imoxion.com>
Message-ID: <51519E1E.3070300@sipwise.com>

Hi,

On 03/26/2013 11:52 AM, ?? wrote:
> So now I trying to solve it could communicate with PSTN.
>
> If you were me, how to find the problem that couldn't communicate with PSTN?
>
> If you were me, where to configure in SIPWISE?

Well, do you already have a SIP trunk to some peering provider? If yes, 
did you configure it in the "SIP Peering" section of the admin panel 
(check the handbook for the how-to)? If yes, what did you put there and 
where, and what does the kamailio-proxy.log say for a call towards this 
peer?

Andreas



From jbonilla at sipwise.com  Tue Mar 26 09:16:26 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Tue, 26 Mar 2013 14:16:26 +0100
Subject: [Spce-user] Mailing list rule changes: Daily digest
In-Reply-To: <20130326100529.22401a25@quenya>
References: <20130326100529.22401a25@quenya>
Message-ID: <20130326141626.5436c526@quenya>

El Tue, 26 Mar 2013 10:05:29 +0100
Jon Bonilla (Manwe) <jbonilla at sipwise.com> escribi?:
 
> For that reason, we plan to remove the
> "receive a daily digest" option form the mailing list and make users to
> receive a every single mail. This change would be effective in the following
> days.


Actually, I have been told that the digest option has already been
disabled. Prepare/adjust your inbox filters. 

:)



-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130326/a7438c8b/attachment.asc>

From rea1ity at kaist.ac.kr  Tue Mar 26 21:15:08 2013
From: rea1ity at kaist.ac.kr (=?UTF-8?B?RGF2aWQgSG8=?=)
Date: Wed, 27 Mar 2013 10:15:08 +0900 (KST)
Subject: [Spce-user]
	=?utf-8?q?PSTN_GW_can_register_SIPWISE_server_but_GW_?=
	=?utf-8?q?peerstatus_always_busy=2E?=
Message-ID: <515248863fe3_@_imoxion.com>

I have already a PSTN Gateway it is not a SIP trunk but it can convert SIP to PSTN.
And I the PSTN Gateway IP set private network. i.e. 192.168.102.254
In "SIP Peering" section of admin panel, I made "test group" and there have "test-gw-1" server it configured same like following;
-------------------
Name : test-gw-1
IP Address : 192.168.102.254 ->it is same IP address with PSTN Gateway because I want to use this PSTN.
Hostname : " "(empty)
Port : UDP
Weight : 1
-------------------
Also Peering Rules values it configured all empty. Callee Prefix, Callee Pattern, Caller Pattern, Description.
And as I opened the log file Kamailio-proxy.log, there are so many logs made at just one call.
So I attached two log files one is a call from SIP phone to SIP phone as locally and the other one is a call from SIP Phone to PSTN Gateway.
I don't know exactly what is point of problem in the log files why PSTN status is always busy.
But as I had inference, it is point why didn't call from SIP phone to PSTN Gateway and it status is always busy.
------------------
Mar 27 01:21:25 spce /usr/sbin/kamailio[1994]: INFO: <script>: No matching rewrite rules for '0112796171' found - R=sip:0112796171 at 192.168.102.254:5060;transport=udp ID=d8bb68a825c52d88dc2813b4e77ace43 at 192.168.102.60
Mar 27 01:21:25 spce /usr/sbin/kamailio[1994]: INFO: <script>: Setting P-Called-Party-ID '<sip:0112796171 at 192.168.102.254>' - R=sip:0112796171 at 192.168.102.254:5060;transport=udp ID=d8bb68a825c52d88dc2813b4e77ace43 at 192.168.102.60
Mar 27 01:21:25 spce /usr/sbin/kamailio[1994]: INFO: <script>: Writing sbc parameters  ;aleg_sst_enable=no;sst_enable=no - R=sip:0112796171 at 192.168.102.254:5060;transport=udp ID=d8bb68a825c52d88dc2813b4e77ace43 at 192.168.102.60
Mar 27 01:21:25 spce /usr/sbin/kamailio[1994]: INFO: <script>: Setting 'sip:192.168.102.254:5060' taken from R-URI as next hop after lb for PSTN call - R=sip:0112796171 at 192.168.102.254:5060;transport=udp ID=d8bb68a825c52d88dc2813b4e77ace43 at 192.168.102.60
Mar 27 01:21:25 spce /usr/sbin/kamailio[1994]: INFO: <script>: Appending P-D-URI 'sip:127.0.0.1:5060;received='sip:192.168.102.254:5060;transport=udp'' - R=sip:0112796171 at 192.168.102.254:5060;transport=udp ID=d8bb68a825c52d88dc2813b4e77ace43 at 192.168.102.60
Mar 27 01:21:25 spce /usr/sbin/kamailio[1994]: INFO: <script>: Forcing request via B2BUA 'sip:127.0.0.1:5080' - R=sip:0112796171 at 192.168.102.254:5060;transport=udp ID=d8bb68a825c52d88dc2813b4e77ace43 at 192.168.102.60
Mar 27 01:21:25 spce /usr/sbin/kamailio[1994]: INFO: <script>: Setting P-Caller-UUID to '0cbfd478-f7eb-445d-9f3c-6061b0f96eb2' - R=sip:0112796171 at 192.168.102.254:5060;transport=udp ID=d8bb68a825c52d88dc2813b4e77ace43 at 192.168.102.60
Mar 27 01:21:25 spce /usr/sbin/kamailio[1994]: INFO: <script>: Setting P-Callee-UUID to '0' - R=sip:0112796171 at 192.168.102.254:5060;transport=udp ID=d8bb68a825c52d88dc2813b4e77ace43 at 192.168.102.60
Mar 27 01:21:25 spce /usr/sbin/kamailio[1994]: INFO: <script>: Request leaving server, D-URI='sip:127.0.0.1:5080' - R=sip:0112796171 at 192.168.102.254:5060;transport=udp ID=d8bb68a825c52d88dc2813b4e77ace43 at 192.168.102.60
Mar 27 01:21:25 spce /usr/sbin/kamailio[1995]: INFO: <script>: NAT-Reply - S=100 - Connecting M=INVITE IP=192.168.102.60:5060 (127.0.0.1:5080) ID=d8bb68a825c52d88dc2813b4e77ace43 at 192.168.102.60
Mar 27 01:21:25 spce /usr/sbin/kamailio[1992]: INFO: <script>: NAT-Reply - S=486 - Busy Here M=INVITE IP=192.168.102.60:5060 (127.0.0.1:5080) ID=d8bb68a825c52d88dc2813b4e77ace43 at 192.168.102.60
Mar 27 01:21:25 spce /usr/sbin/kamailio[1992]: INFO: <script>: Failure route for PSTN call - R=sip:0112796171 at sip.baekdoo.local:5060;transport=udp ID=d8bb68a825c52d88dc2813b4e77ace43 at 192.168.102.60
------------------------


So I don't know exactly but the SIP Server tried to call the PSTN Gateway as the SIPWISE server IP 127.0.0.1:5080 as local SIP it referenced following a log in upper logs.
-------------------------------
Forcing request via B2BUA 'sip:127.0.0.1:5080' - R=sip:0112796171 at 192.168.102.254:5060;transport=udp
-------------------------------

But SIPWISE server couldn't reached to PSTN gateway because IP networks are different each others.
So SIPWISE can't reached the network 192.168.102.254:5060 that is only my opinion.


Please let me know if you find what was problem from connect with from SIP phone to PSTN network.


David Ho




----------------------????-----------------------
???? : Andreas Granig <agranig at sipwise.com>
????: spce-user at lists.sipwise.com
????: 2013-03-26 22:09:50 GMT +0900 (Asia/Seoul)
??: Re: [Spce-user] PSTN GW can register SIPWISE server but GW peerstatus always busy.

Hi,

On 03/26/2013 11:52 AM, ?? wrote:
> So now I trying to solve it could communicate with PSTN.
>
> If you were me, how to find the problem that couldn't communicate with PSTN?
>
> If you were me, where to configure in SIPWISE?

Well, do you already have a SIP trunk to some peering provider? If yes, 
did you configure it in the "SIP Peering" section of the admin panel 
(check the handbook for the how-to)? If yes, what did you put there and 
where, and what does the kamailio-proxy.log say for a call towards this 
peer?

Andreas

_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com
http://lists.sipwise.com/listinfo/spce-user


-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130327/1ca17207/attachment.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: SIP_to_SIP_Locally.txt
Type: application/octet-stream
Size: 17754 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130327/1ca17207/attachment.txt>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: SIP_to_PSTN_GW.txt
Type: application/octet-stream
Size: 16571 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130327/1ca17207/attachment-0001.txt>

From agranig at sipwise.com  Wed Mar 27 04:34:36 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Wed, 27 Mar 2013 09:34:36 +0100
Subject: [Spce-user] PSTN GW can register SIPWISE server but GW
 peerstatus always busy.
In-Reply-To: <515248833fda_@_imoxion.com>
References: <515248833fda_@_imoxion.com>
Message-ID: <5152AF1C.5000601@mail.sipwise.com>

Hi,

On 03/27/2013 02:15 AM, David Ho wrote:
> But SIPWISE server couldn't reached to PSTN gateway because IP
> networks are different each others.
>
> So SIPWISE can't reached the network 192.168.102.254:5060 that is only
> my opinion.

Well, since you get this line in the logs:

NAT-Reply - S=486 - Busy Here M=INVITE IP=192.168.102.60:5060

it clearly indicates that your GW is in fact getting the request, but as 
logged, answers with Busy. You need to check the logs of your peer/gw to 
see why it does so.

Maybe it doesn't like some headers, number formats or similar, but we 
can just guess without knowing what the GW says in its logs.

Andreas



From sn at fest.dk  Wed Mar 27 10:56:37 2013
From: sn at fest.dk (Sylvester Nielsen)
Date: Wed, 27 Mar 2013 14:56:37 +0000
Subject: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
Message-ID: <E50575DD48D9B743925AC3BCE6310A4B2436E488@DB3PRD0710MB355.eurprd07.prod.outlook.com>

Hi,

When we send an invite to the peer we have ngcpct='sip:127.0.0.1:5080' in the contact header.
Can someone tell me what this is and if we can remove it? For some reason the peer think it should send BYE and CANCEL to this address which makes calls hang.

Med venlig hilsen

   Sylvester Nielsen

---------------------------------------------------------------

Fest Data A/S
Generatorvej 8D, 1.th
2730 Herlev

e-mail : sn at fest.dk<mailto:sn at fest.dk>

Tlf:     +45 4582 7389
Fax:   +45 4582 7387
Dir:    +45 6996 2011

http://www.festdata.dk

[Beskrivelse: Description: Description: Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse: cid:image003.jpg at 01CBCE06.BAABD4A0]




-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130327/7e2a023b/attachment.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.jpg
Type: image/jpeg
Size: 11834 bytes
Desc: image001.jpg
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130327/7e2a023b/attachment.jpg>

From agranig at sipwise.com  Wed Mar 27 11:01:23 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Wed, 27 Mar 2013 16:01:23 +0100
Subject: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
In-Reply-To: <E50575DD48D9B743925AC3BCE6310A4B2436E488@DB3PRD0710MB355.eurprd07.prod.outlook.com>
References: <E50575DD48D9B743925AC3BCE6310A4B2436E488@DB3PRD0710MB355.eurprd07.prod.outlook.com>
Message-ID: <515309C3.1080601@mail.sipwise.com>

Hi,

Any URI params in the Contact header must to be preserved for correct 
routing, otherwise you're opening a huge can of worms.

Andreas

On 03/27/2013 03:56 PM, Sylvester Nielsen wrote:
> Hi,
>
> When we send an invite to the peer we have ngcpct='sip:127.0.0.1:5080'
> in the contact header.
>
> Can someone tell me what this is and if we can remove it? For some
> reason the peer think it should send BYE and CANCEL to this address
> which makes calls hang.
>
> Med venlig hilsen
>
>     Sylvester Nielsen
>
> ---------------------------------------------------------------
>
> 	
>
> Fest Data A/S
> Generatorvej 8D, 1.th
> 2730 Herlev
>
> e-mail : sn at fest.dk <mailto:sn at fest.dk>
>
> 	
>
> Tlf:     +45 4582 7389
> Fax:   +45 4582 7387
> Dir:    +45 6996 2011
>
> http://www.festdata.dk
>
> 	
>
> Beskrivelse: Description: Description: Beskrivelse: Beskrivelse:
> Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse:
> cid:image003.jpg at 01CBCE06.BAABD4A0
>
> 		
>
>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
>



From john.murray at skyracktelecom.com  Wed Mar 27 11:11:48 2013
From: john.murray at skyracktelecom.com (John Murray)
Date: Wed, 27 Mar 2013 15:11:48 -0000
Subject: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
In-Reply-To: <515309C3.1080601@mail.sipwise.com>
References: <E50575DD48D9B743925AC3BCE6310A4B2436E488@DB3PRD0710MB355.eurprd07.prod.outlook.com>
	<515309C3.1080601@mail.sipwise.com>
Message-ID: <51530c35.a266b40a.3a4f.ffffb1e3@mx.google.com>

Andreas,

With a peer you are typically in some sort of relationship and can negotiate
any fixes (not always the case of course). But with a subscriber you are
dependent on the code in the device.
In my experience with PBXs ( naming no names (but NEC is terrible)) are very
bad with this header so it would be very helpful if there were some switch
against the subscriber to remove this as a workaround.
Getting the vendor to do something about this has always proven very
difficult/painful/slow.

Regards

John 

-----Original Message-----
From: spce-user-bounces at lists.sipwise.com
[mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of Andreas Granig
Sent: 27 March 2013 15:01
To: spce-user at lists.sipwise.com
Subject: Re: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header

Hi,

Any URI params in the Contact header must to be preserved for correct
routing, otherwise you're opening a huge can of worms.

Andreas

On 03/27/2013 03:56 PM, Sylvester Nielsen wrote:
> Hi,
>
> When we send an invite to the peer we have ngcpct='sip:127.0.0.1:5080'
> in the contact header.
>
> Can someone tell me what this is and if we can remove it? For some 
> reason the peer think it should send BYE and CANCEL to this address 
> which makes calls hang.
>
> Med venlig hilsen
>
>     Sylvester Nielsen
>
> ---------------------------------------------------------------
>
> 	
>
> Fest Data A/S
> Generatorvej 8D, 1.th
> 2730 Herlev
>
> e-mail : sn at fest.dk <mailto:sn at fest.dk>
>
> 	
>
> Tlf:     +45 4582 7389
> Fax:   +45 4582 7387
> Dir:    +45 6996 2011
>
> http://www.festdata.dk
>
> 	
>
> Beskrivelse: Description: Description: Beskrivelse: Beskrivelse:
> Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse:
> cid:image003.jpg at 01CBCE06.BAABD4A0
>
> 		
>
>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
>

_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com
http://lists.sipwise.com/listinfo/spce-user




From gary at netdiverse.com  Wed Mar 27 11:39:20 2013
From: gary at netdiverse.com (Gary Nieboer)
Date: Wed, 27 Mar 2013 09:39:20 -0600
Subject: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
In-Reply-To: <515309C3.1080601@mail.sipwise.com>
References: <E50575DD48D9B743925AC3BCE6310A4B2436E488@DB3PRD0710MB355.eurprd07.prod.outlook.com>
	<515309C3.1080601@mail.sipwise.com>
Message-ID: <CAMXUi2TGRE3b8FaRZSi_nmtAfKg_ii7Q-JYsPebsvauC0iM_Xg@mail.gmail.com>

Andreas,

We have the same issue with Level 3 Communications / Global Crossing on
their Enterprise SIP Platform.   As background, they have a Acme Packet SBC
in front of their Broadsoft App Server.

When the party on the Peer/Level3 side of the network sends a BYE, it never
reaches the subscriber on the SPCE side and hangs.  If the SPCE subscriber
sends the BYE, then it appears to work properly.

>From everything we can see and with feedback from Level , their SBC or App
Server appears to be getting confused by the address SPCE puts in the
contact ID field.   Is there a setting that will put an additional or
different IP address in the Contact ID field?    I likely can't get a
packet capture until Thursday night (US-EDT), but would be happy to provide
one privately later in the week.

-Gary


--
Gary Nieboer


877-763-1014 Toll Free  |  801-657-5484 Utah  |  615-800-6946 Tennessee  |
801-694-6106 Mobile
gary at netdiverse.com  |   www.netdiverse.com


On Wed, Mar 27, 2013 at 9:01 AM, Andreas Granig <agranig at sipwise.com> wrote:

> Hi,
>
> Any URI params in the Contact header must to be preserved for correct
> routing, otherwise you're opening a huge can of worms.
>
> Andreas
>
>
> On 03/27/2013 03:56 PM, Sylvester Nielsen wrote:
>
>> Hi,
>>
>> When we send an invite to the peer we have ngcpct='sip:127.0.0.1:5080'
>> in the contact header.
>>
>> Can someone tell me what this is and if we can remove it? For some
>> reason the peer think it should send BYE and CANCEL to this address
>> which makes calls hang.
>>
>> Med venlig hilsen
>>
>>     Sylvester Nielsen
>>
>> ------------------------------**------------------------------**---
>>
>>
>>
>> Fest Data A/S
>> Generatorvej 8D, 1.th
>> 2730 Herlev
>>
>> e-mail : sn at fest.dk <mailto:sn at fest.dk>
>>
>>
>>
>>
>> Tlf:     +45 4582 7389
>> Fax:   +45 4582 7387
>> Dir:    +45 6996 2011
>>
>> http://www.festdata.dk
>>
>>
>>
>> Beskrivelse: Description: Description: Beskrivelse: Beskrivelse:
>> Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse:
>> cid:image003.jpg at 01CBCE06.**BAABD4A0
>>
>>
>>
>>
>>
>> ______________________________**_________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/**listinfo/spce-user<http://lists.sipwise.com/listinfo/spce-user>
>>
>>
> ______________________________**_________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/**listinfo/spce-user<http://lists.sipwise.com/listinfo/spce-user>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130327/ae023c36/attachment.html>

From agranig at sipwise.com  Wed Mar 27 11:41:14 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Wed, 27 Mar 2013 16:41:14 +0100
Subject: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
In-Reply-To: <51530c35.a266b40a.3a4f.ffffb1e3@mx.google.com>
References: <E50575DD48D9B743925AC3BCE6310A4B2436E488@DB3PRD0710MB355.eurprd07.prod.outlook.com>
	<515309C3.1080601@mail.sipwise.com>
	<51530c35.a266b40a.3a4f.ffffb1e3@mx.google.com>
Message-ID: <5153131A.7060002@mail.sipwise.com>

Hi,

On 03/27/2013 04:11 PM, John Murray wrote:
> With a peer you are typically in some sort of relationship and can negotiate
> any fixes (not always the case of course). But with a subscriber you are
> dependent on the code in the device.
> In my experience with PBXs ( naming no names (but NEC is terrible)) are very
> bad with this header so it would be very helpful if there were some switch
> against the subscriber to remove this as a workaround.
> Getting the vendor to do something about this has always proven very
> difficult/painful/slow.

Well, the reason this has been added was to mask the local sems contact 
of 127.0.0.1:5080, because some peers are doing far-end nat traversal 
(like the SPCE itself), and if some peer tries to fix this, the SIP flow 
is again broken.

So in one way or another, this information needs to be coded into the 
message. I'm open for any better suggestions of course :)

Andreas



From john.murray at skyracktelecom.com  Wed Mar 27 11:52:40 2013
From: john.murray at skyracktelecom.com (John Murray)
Date: Wed, 27 Mar 2013 15:52:40 -0000
Subject: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
In-Reply-To: <CAMXUi2TGRE3b8FaRZSi_nmtAfKg_ii7Q-JYsPebsvauC0iM_Xg@mail.gmail.com>
References: <E50575DD48D9B743925AC3BCE6310A4B2436E488@DB3PRD0710MB355.eurprd07.prod.outlook.com>
	<515309C3.1080601@mail.sipwise.com>
	<CAMXUi2TGRE3b8FaRZSi_nmtAfKg_ii7Q-JYsPebsvauC0iM_Xg@mail.gmail.com>
Message-ID: <515315ca.2560b40a.32bb.ffffc590@mx.google.com>

Gary,

 

Check out the Audiocodes workaround in the handbook section 5.5.1 if the
problem in the address itself being 'loopback'

We use our external address with the first octet replaced with '10' for
internal communications. It makes debugging easier.

This fixes a number of systems getting confused.

 

Regards

 

John

 

From: spce-user-bounces at lists.sipwise.com
[mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of Gary Nieboer
Sent: 27 March 2013 15:39
To: spce-user at lists.sipwise.com
Subject: Re: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header

 

Andreas, 

 

We have the same issue with Level 3 Communications / Global Crossing on
their Enterprise SIP Platform.   As background, they have a Acme Packet SBC
in front of their Broadsoft App Server.

 

When the party on the Peer/Level3 side of the network sends a BYE, it never
reaches the subscriber on the SPCE side and hangs.  If the SPCE subscriber
sends the BYE, then it appears to work properly. 

 

>From everything we can see and with feedback from Level , their SBC or App
Server appears to be getting confused by the address SPCE puts in the
contact ID field.   Is there a setting that will put an additional or
different IP address in the Contact ID field?    I likely can't get a packet
capture until Thursday night (US-EDT), but would be happy to provide one
privately later in the week.

 

-Gary

 

 

--
Gary Nieboer

  <http://www.netdiverse.com/content/images/NewNDLogo180wide.png> 


877-763-1014 Toll Free  |  801-657-5484 Utah  |  615-800-6946 Tennessee  |
801-694-6106 Mobile
gary at netdiverse.com  |   www.netdiverse.com <http://www.netdiverse.com/> 

 

On Wed, Mar 27, 2013 at 9:01 AM, Andreas Granig <agranig at sipwise.com> wrote:

Hi,

Any URI params in the Contact header must to be preserved for correct
routing, otherwise you're opening a huge can of worms.

Andreas



On 03/27/2013 03:56 PM, Sylvester Nielsen wrote:

Hi,

When we send an invite to the peer we have ngcpct='sip:127.0.0.1:5080'
in the contact header.

Can someone tell me what this is and if we can remove it? For some
reason the peer think it should send BYE and CANCEL to this address
which makes calls hang.

Med venlig hilsen

    Sylvester Nielsen

---------------------------------------------------------------

        

Fest Data A/S
Generatorvej 8D, 1.th
2730 Herlev

e-mail : sn at fest.dk <mailto:sn at fest.dk>



        

Tlf:     +45 4582 7389 <tel:%2B45%204582%207389> 
Fax:   +45 4582 7387 <tel:%2B45%204582%207387> 
Dir:    +45 6996 2011 <tel:%2B45%206996%202011> 

http://www.festdata.dk

        

Beskrivelse: Description: Description: Beskrivelse: Beskrivelse:
Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse:
cid:image003.jpg at 01CBCE06.BAABD4A0

                




_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com
http://lists.sipwise.com/listinfo/spce-user


_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com
http://lists.sipwise.com/listinfo/spce-user

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130327/26d8e552/attachment.html>

From skchopperguy at gmail.com  Wed Mar 27 11:55:20 2013
From: skchopperguy at gmail.com (Skyler)
Date: Wed, 27 Mar 2013 08:55:20 -0700
Subject: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
In-Reply-To: <5153131A.7060002@mail.sipwise.com>
References: <E50575DD48D9B743925AC3BCE6310A4B2436E488@DB3PRD0710MB355.eurprd07.prod.outlook.com>
	<515309C3.1080601@mail.sipwise.com>
	<51530c35.a266b40a.3a4f.ffffb1e3@mx.google.com>
	<5153131A.7060002@mail.sipwise.com>
Message-ID: <51531668.2090106@gmail.com>

Hi,

On 3/27/2013 8:41 AM, Andreas Granig wrote:
> Hi,
>
> On 03/27/2013 04:11 PM, John Murray wrote:
>> With a peer you are typically in some sort of relationship and can
>> negotiate
>> any fixes (not always the case of course). But with a subscriber you are
>> dependent on the code in the device.
>> In my experience with PBXs ( naming no names (but NEC is terrible))
>> are very
>> bad with this header so it would be very helpful if there were some
>> switch
>> against the subscriber to remove this as a workaround.
>> Getting the vendor to do something about this has always proven very
>> difficult/painful/slow.
>
> Well, the reason this has been added was to mask the local sems contact
> of 127.0.0.1:5080, because some peers are doing far-end nat traversal
> (like the SPCE itself), and if some peer tries to fix this, the SIP flow
> is again broken.
>
> So in one way or another, this information needs to be coded into the
> message. I'm open for any better suggestions of course :)
>
> Andreas
>

  I created an lo:0 with a private IP address (10.10.xxx.xxx) for lb > 
b2buas: & proxies > sip:. That may help in this case as its likely the 
127.0.0.1 which is confusing the upstream proxy.

Skyler



From john.murray at skyracktelecom.com  Wed Mar 27 12:03:21 2013
From: john.murray at skyracktelecom.com (John Murray)
Date: Wed, 27 Mar 2013 16:03:21 -0000
Subject: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
In-Reply-To: <51531668.2090106@gmail.com>
References: <E50575DD48D9B743925AC3BCE6310A4B2436E488@DB3PRD0710MB355.eurprd07.prod.outlook.com>
	<515309C3.1080601@mail.sipwise.com>
	<51530c35.a266b40a.3a4f.ffffb1e3@mx.google.com>
	<5153131A.7060002@mail.sipwise.com> <51531668.2090106@gmail.com>
Message-ID: <5153184b.8a0db40a.5bf5.ffffc4e1@mx.google.com>

Yes one problem with some vendors is the loopback address of 127.0.0.1 which
as you say can be fixed by changing the internal address as described in
handbook section 5.5.1.
The other problem some vendors have is with the 'ngcpct=' section of the
Contact: header as:

Contact: <sip:ngcp-lb at 10.23.11.18:5060;ngcpct='sip:10.23.11.18:5080'>'

Some vendors 'forget' this and reply to sip:ngcp-lb at 10.23.11.18:5060 and
some get stuck and don't reply at all when they see this.

Regards

John

-----Original Message-----
From: Skyler [mailto:skchopperguy at gmail.com] 
Sent: 27 March 2013 15:55
To: Andreas Granig
Cc: John Murray; spce-user at lists.sipwise.com
Subject: Re: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header

Hi,

On 3/27/2013 8:41 AM, Andreas Granig wrote:
> Hi,
>
> On 03/27/2013 04:11 PM, John Murray wrote:
>> With a peer you are typically in some sort of relationship and can 
>> negotiate any fixes (not always the case of course). But with a 
>> subscriber you are dependent on the code in the device.
>> In my experience with PBXs ( naming no names (but NEC is terrible)) 
>> are very bad with this header so it would be very helpful if there 
>> were some switch against the subscriber to remove this as a 
>> workaround.
>> Getting the vendor to do something about this has always proven very 
>> difficult/painful/slow.
>
> Well, the reason this has been added was to mask the local sems 
> contact of 127.0.0.1:5080, because some peers are doing far-end nat 
> traversal (like the SPCE itself), and if some peer tries to fix this, 
> the SIP flow is again broken.
>
> So in one way or another, this information needs to be coded into the 
> message. I'm open for any better suggestions of course :)
>
> Andreas
>

  I created an lo:0 with a private IP address (10.10.xxx.xxx) for lb >
b2buas: & proxies > sip:. That may help in this case as its likely the
127.0.0.1 which is confusing the upstream proxy.

Skyler




From agranig at sipwise.com  Wed Mar 27 12:07:11 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Wed, 27 Mar 2013 17:07:11 +0100
Subject: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
In-Reply-To: <51531668.2090106@gmail.com>
References: <E50575DD48D9B743925AC3BCE6310A4B2436E488@DB3PRD0710MB355.eurprd07.prod.outlook.com>
	<515309C3.1080601@mail.sipwise.com>
	<51530c35.a266b40a.3a4f.ffffb1e3@mx.google.com>
	<5153131A.7060002@mail.sipwise.com> <51531668.2090106@gmail.com>
Message-ID: <5153192F.4050106@mail.sipwise.com>

Hi,

On 03/27/2013 04:55 PM, Skyler wrote:
>   I created an lo:0 with a private IP address (10.10.xxx.xxx) for lb >
> b2buas: & proxies > sip:. That may help in this case as its likely the
> 127.0.0.1 which is confusing the upstream proxy.

I've heard of endpoints disliking the SIP uri with the quotes... so one 
option could be to convert the "ngcpct='sip:127.0.0.1:5080'" and 
base64-encode the "sip:127.0.0.1:5080", so it gets something like 
"ngcpct=c2lwOjEyNy4wLjAuMTo1MDgwCg==" instead.

Andreas



From skchopperguy at gmail.com  Wed Mar 27 12:09:03 2013
From: skchopperguy at gmail.com (Skyler)
Date: Wed, 27 Mar 2013 09:09:03 -0700
Subject: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
In-Reply-To: <5153184b.8a0db40a.5bf5.ffffc4e1@mx.google.com>
References: <E50575DD48D9B743925AC3BCE6310A4B2436E488@DB3PRD0710MB355.eurprd07.prod.outlook.com>
	<515309C3.1080601@mail.sipwise.com>
	<51530c35.a266b40a.3a4f.ffffb1e3@mx.google.com>
	<5153131A.7060002@mail.sipwise.com> <51531668.2090106@gmail.com>
	<5153184b.8a0db40a.5bf5.ffffc4e1@mx.google.com>
Message-ID: <5153199F.2020608@gmail.com>

Ah good to know, I haven't seen that be a problem 'yet' (knock wood). 
I'll step away :)

Skyler

On 3/27/2013 9:03 AM, John Murray wrote:
> Yes one problem with some vendors is the loopback address of 127.0.0.1 which
> as you say can be fixed by changing the internal address as described in
> handbook section 5.5.1.
> The other problem some vendors have is with the 'ngcpct=' section of the
> Contact: header as:
>
> Contact: <sip:ngcp-lb at 10.23.11.18:5060;ngcpct='sip:10.23.11.18:5080'>'
>
> Some vendors 'forget' this and reply to sip:ngcp-lb at 10.23.11.18:5060 and
> some get stuck and don't reply at all when they see this.
>
> Regards
>
> John
>
> -----Original Message-----
> From: Skyler [mailto:skchopperguy at gmail.com]
> Sent: 27 March 2013 15:55
> To: Andreas Granig
> Cc: John Murray; spce-user at lists.sipwise.com
> Subject: Re: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
>
> Hi,
>
> On 3/27/2013 8:41 AM, Andreas Granig wrote:
>> Hi,
>>
>> On 03/27/2013 04:11 PM, John Murray wrote:
>>> With a peer you are typically in some sort of relationship and can
>>> negotiate any fixes (not always the case of course). But with a
>>> subscriber you are dependent on the code in the device.
>>> In my experience with PBXs ( naming no names (but NEC is terrible))
>>> are very bad with this header so it would be very helpful if there
>>> were some switch against the subscriber to remove this as a
>>> workaround.
>>> Getting the vendor to do something about this has always proven very
>>> difficult/painful/slow.
>>
>> Well, the reason this has been added was to mask the local sems
>> contact of 127.0.0.1:5080, because some peers are doing far-end nat
>> traversal (like the SPCE itself), and if some peer tries to fix this,
>> the SIP flow is again broken.
>>
>> So in one way or another, this information needs to be coded into the
>> message. I'm open for any better suggestions of course :)
>>
>> Andreas
>>
>
>    I created an lo:0 with a private IP address (10.10.xxx.xxx) for lb >
> b2buas: & proxies > sip:. That may help in this case as its likely the
> 127.0.0.1 which is confusing the upstream proxy.
>
> Skyler
>



From skchopperguy at gmail.com  Wed Mar 27 12:31:33 2013
From: skchopperguy at gmail.com (Skyler)
Date: Wed, 27 Mar 2013 09:31:33 -0700
Subject: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
In-Reply-To: <5153192F.4050106@mail.sipwise.com>
References: <E50575DD48D9B743925AC3BCE6310A4B2436E488@DB3PRD0710MB355.eurprd07.prod.outlook.com>
	<515309C3.1080601@mail.sipwise.com>
	<51530c35.a266b40a.3a4f.ffffb1e3@mx.google.com>
	<5153131A.7060002@mail.sipwise.com> <51531668.2090106@gmail.com>
	<5153192F.4050106@mail.sipwise.com>
Message-ID: <51531EE5.1010802@gmail.com>

That makes sense, also might just be that the ngcpct sip uri is the last 
uri read in the contact.

  Maybe Sylvester and Gary could test this out as they have the OP.

What would a good test look like to base64-encode that ngcpct in a 
customtt.tt2?

Skyler

On 3/27/2013 9:07 AM, Andreas Granig wrote:
> Hi,
>
> On 03/27/2013 04:55 PM, Skyler wrote:
>>   I created an lo:0 with a private IP address (10.10.xxx.xxx) for lb >
>> b2buas: & proxies > sip:. That may help in this case as its likely the
>> 127.0.0.1 which is confusing the upstream proxy.
>
> I've heard of endpoints disliking the SIP uri with the quotes... so one
> option could be to convert the "ngcpct='sip:127.0.0.1:5080'" and
> base64-encode the "sip:127.0.0.1:5080", so it gets something like
> "ngcpct=c2lwOjEyNy4wLjAuMTo1MDgwCg==" instead.
>
> Andreas



From gary at netdiverse.com  Wed Mar 27 12:42:12 2013
From: gary at netdiverse.com (Gary Nieboer)
Date: Wed, 27 Mar 2013 10:42:12 -0600
Subject: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
In-Reply-To: <51531EE5.1010802@gmail.com>
References: <E50575DD48D9B743925AC3BCE6310A4B2436E488@DB3PRD0710MB355.eurprd07.prod.outlook.com>
	<515309C3.1080601@mail.sipwise.com>
	<51530c35.a266b40a.3a4f.ffffb1e3@mx.google.com>
	<5153131A.7060002@mail.sipwise.com> <51531668.2090106@gmail.com>
	<5153192F.4050106@mail.sipwise.com> <51531EE5.1010802@gmail.com>
Message-ID: <CAMXUi2QV2vOAtpSW_CJU73oy=iScpKV0cx+YdShSk6Gi64_pnA@mail.gmail.com>

I will give a couple of the items advised a try.

Should there be more than one uri in the Contact field?  If so, that may be
the quick fix.  The contact field I captured a few weeks ago only has one,
and it is the loopback/proxy address of 127.0.0.1:5080

-Gary
--
Gary Nieboer


877-763-1014 Toll Free  |  801-657-5484 Utah  |  615-800-6946 Tennessee  |
801-694-6106 Mobile
gary at netdiverse.com  |   www.netdiverse.com


On Wed, Mar 27, 2013 at 10:31 AM, Skyler <skchopperguy at gmail.com> wrote:

> That makes sense, also might just be that the ngcpct sip uri is the last
> uri read in the contact.
>
>  Maybe Sylvester and Gary could test this out as they have the OP.
>
> What would a good test look like to base64-encode that ngcpct in a
> customtt.tt2?
>
> Skyler
>
>
> On 3/27/2013 9:07 AM, Andreas Granig wrote:
>
>> Hi,
>>
>> On 03/27/2013 04:55 PM, Skyler wrote:
>>
>>>   I created an lo:0 with a private IP address (10.10.xxx.xxx) for lb >
>>> b2buas: & proxies > sip:. That may help in this case as its likely the
>>> 127.0.0.1 which is confusing the upstream proxy.
>>>
>>
>> I've heard of endpoints disliking the SIP uri with the quotes... so one
>> option could be to convert the "ngcpct='sip:127.0.0.1:5080'" and
>> base64-encode the "sip:127.0.0.1:5080", so it gets something like
>> "ngcpct=**c2lwOjEyNy4wLjAuMTo1MDgwCg==" instead.
>>
>> Andreas
>>
>
> ______________________________**_________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/**listinfo/spce-user<http://lists.sipwise.com/listinfo/spce-user>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130327/8e2bce26/attachment.html>

From skchopperguy at gmail.com  Wed Mar 27 12:48:44 2013
From: skchopperguy at gmail.com (Skyler)
Date: Wed, 27 Mar 2013 09:48:44 -0700
Subject: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
In-Reply-To: <CAMXUi2QV2vOAtpSW_CJU73oy=iScpKV0cx+YdShSk6Gi64_pnA@mail.gmail.com>
References: <E50575DD48D9B743925AC3BCE6310A4B2436E488@DB3PRD0710MB355.eurprd07.prod.outlook.com>
	<515309C3.1080601@mail.sipwise.com>
	<51530c35.a266b40a.3a4f.ffffb1e3@mx.google.com>
	<5153131A.7060002@mail.sipwise.com> <51531668.2090106@gmail.com>
	<5153192F.4050106@mail.sipwise.com> <51531EE5.1010802@gmail.com>
	<CAMXUi2QV2vOAtpSW_CJU73oy=iScpKV0cx+YdShSk6Gi64_pnA@mail.gmail.com>
Message-ID: <515322EC.5080908@gmail.com>

My guess (as I've not actually had this issue) is that your contact 
would look something like:

Contact: <sip:ngcp-lb at your.ext.ip.address:5060;ngcpct='sip:127.0.0.1:5080'>

Skyler

On 3/27/2013 9:42 AM, Gary Nieboer wrote:
> I will give a couple of the items advised a try.
>
> Should there be more than one uri in the Contact field?  If so, that may
> be the quick fix.  The contact field I captured a few weeks ago only has
> one, and it is the loopback/proxy address of 127.0.0.1:5080
> <http://127.0.0.1:5080>
>
> -Gary
> --
> Gary Nieboer
>
>
> 877-763-1014 Toll Free  |  801-657-5484 Utah  |  615-800-6946 Tennessee
> | 801-694-6106 Mobile
> gary at netdiverse.com <mailto:gary at netdiverse.com>  | www.netdiverse.com
> <http://www.netdiverse.com/>
>
>
> On Wed, Mar 27, 2013 at 10:31 AM, Skyler <skchopperguy at gmail.com
> <mailto:skchopperguy at gmail.com>> wrote:
>
>     That makes sense, also might just be that the ngcpct sip uri is the
>     last uri read in the contact.
>
>       Maybe Sylvester and Gary could test this out as they have the OP.
>
>     What would a good test look like to base64-encode that ngcpct in a
>     customtt.tt2?
>
>     Skyler
>
>
>     On 3/27/2013 9:07 AM, Andreas Granig wrote:
>
>         Hi,
>
>         On 03/27/2013 04:55 PM, Skyler wrote:
>
>                I created an lo:0 with a private IP address
>             (10.10.xxx.xxx) for lb >
>             b2buas: & proxies > sip:. That may help in this case as its
>             likely the
>             127.0.0.1 which is confusing the upstream proxy.
>
>
>         I've heard of endpoints disliking the SIP uri with the quotes...
>         so one
>         option could be to convert the "ngcpct='sip:127.0.0.1:5080
>         <http://127.0.0.1:5080>'" and
>         base64-encode the "sip:127.0.0.1:5080 <http://127.0.0.1:5080>",
>         so it gets something like
>         "ngcpct=__c2lwOjEyNy4wLjAuMTo1MDgwCg==" instead.
>
>         Andreas
>
>
>     _________________________________________________
>     Spce-user mailing list
>     Spce-user at lists.sipwise.com <mailto:Spce-user at lists.sipwise.com>
>     http://lists.sipwise.com/__listinfo/spce-user
>     <http://lists.sipwise.com/listinfo/spce-user>
>
>



From gary at netdiverse.com  Wed Mar 27 13:17:23 2013
From: gary at netdiverse.com (Gary Nieboer)
Date: Wed, 27 Mar 2013 11:17:23 -0600
Subject: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
In-Reply-To: <515322EC.5080908@gmail.com>
References: <E50575DD48D9B743925AC3BCE6310A4B2436E488@DB3PRD0710MB355.eurprd07.prod.outlook.com>
	<515309C3.1080601@mail.sipwise.com>
	<51530c35.a266b40a.3a4f.ffffb1e3@mx.google.com>
	<5153131A.7060002@mail.sipwise.com> <51531668.2090106@gmail.com>
	<5153192F.4050106@mail.sipwise.com> <51531EE5.1010802@gmail.com>
	<CAMXUi2QV2vOAtpSW_CJU73oy=iScpKV0cx+YdShSk6Gi64_pnA@mail.gmail.com>
	<515322EC.5080908@gmail.com>
Message-ID: <CAMXUi2QrywYL5wEcXk-A2hDx_4tcDgtAcx9wpdxU1LREVwYFtw@mail.gmail.com>

I found an old pcap with an invite showing the Contact.

That could be the issue.   On the SPCE Invite to the Peer, the only thing
in the capture is:

Contact: <sip:127.0.0.1:5080>


and nothing in the Contact field about our ip:  x.x.x.x:5060 or with
 "ngcp-lb" or "ngcpct"




-Gary
--
Gary Nieboer


877-763-1014 Toll Free  |  801-657-5484 Utah  |  615-800-6946 Tennessee  |
801-694-6106 Mobile
gary at netdiverse.com  |   www.netdiverse.com


On Wed, Mar 27, 2013 at 10:48 AM, Skyler <skchopperguy at gmail.com> wrote:

> My guess (as I've not actually had this issue) is that your contact would
> look something like:
>
> Contact: <sip:ngcp-lb at your.ext.ip.**address:5060;ngcpct='sip:127.**
> 0.0.1:5080 <http://127.0.0.1:5080>'>
>
> Skyler
>
>
> On 3/27/2013 9:42 AM, Gary Nieboer wrote:
>
>> I will give a couple of the items advised a try.
>>
>> Should there be more than one uri in the Contact field?  If so, that may
>> be the quick fix.  The contact field I captured a few weeks ago only has
>> one, and it is the loopback/proxy address of 127.0.0.1:5080
>> <http://127.0.0.1:5080>
>>
>> -Gary
>> --
>> Gary Nieboer
>>
>>
>> 877-763-1014 Toll Free  |  801-657-5484 Utah  |  615-800-6946 Tennessee
>> | 801-694-6106 Mobile
>> gary at netdiverse.com <mailto:gary at netdiverse.com>  | www.netdiverse.com
>> <http://www.netdiverse.com/>
>>
>>
>>
>> On Wed, Mar 27, 2013 at 10:31 AM, Skyler <skchopperguy at gmail.com
>> <mailto:skchopperguy at gmail.com**>> wrote:
>>
>>     That makes sense, also might just be that the ngcpct sip uri is the
>>     last uri read in the contact.
>>
>>       Maybe Sylvester and Gary could test this out as they have the OP.
>>
>>     What would a good test look like to base64-encode that ngcpct in a
>>     customtt.tt2?
>>
>>     Skyler
>>
>>
>>     On 3/27/2013 9:07 AM, Andreas Granig wrote:
>>
>>         Hi,
>>
>>         On 03/27/2013 04:55 PM, Skyler wrote:
>>
>>                I created an lo:0 with a private IP address
>>             (10.10.xxx.xxx) for lb >
>>             b2buas: & proxies > sip:. That may help in this case as its
>>             likely the
>>             127.0.0.1 which is confusing the upstream proxy.
>>
>>
>>         I've heard of endpoints disliking the SIP uri with the quotes...
>>         so one
>>         option could be to convert the "ngcpct='sip:127.0.0.1:5080
>>         <http://127.0.0.1:5080>'" and
>>         base64-encode the "sip:127.0.0.1:5080 <http://127.0.0.1:5080>",
>>
>>         so it gets something like
>>         "ngcpct=__**c2lwOjEyNy4wLjAuMTo1MDgwCg==" instead.
>>
>>         Andreas
>>
>>
>>     ______________________________**___________________
>>     Spce-user mailing list
>>     Spce-user at lists.sipwise.com <mailto:Spce-user at lists.**sipwise.com<Spce-user at lists.sipwise.com>
>> >
>>     http://lists.sipwise.com/__**listinfo/spce-user<http://lists.sipwise.com/__listinfo/spce-user>
>>     <http://lists.sipwise.com/**listinfo/spce-user<http://lists.sipwise.com/listinfo/spce-user>
>> >
>>
>>
>>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130327/29e262db/attachment.html>

From m.rys at tri-lakes.net  Wed Mar 27 13:30:08 2013
From: m.rys at tri-lakes.net (Marc Rys)
Date: Wed, 27 Mar 2013 12:30:08 -0500 (CDT)
Subject: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
In-Reply-To: <515322EC.5080908@gmail.com>
Message-ID: <c44cce95-50a6-41bf-9ca3-931723259972@usbsnpzimbra01v.servers.rystec.com>

Ok, Im not sure if this is related, but I've been having struggles with our Taqua Gateway peering to SPCE.  OUr Gateway kept replying back with a 403 Forbidden whenever SPCE tried to send a call to it.

After seeing your dialog about the contact headers, i looked at one our wireshark captures and our Invites were showing the same thing...
sip:ngcp-lb at your.ext.ip.address:5060;ngcpct='sip:127.0.0.1:5080'

I followed the instructions for the audiocodes fix and changed our dummy address to 10.10.10.1  On our Taqua Gateway I had to add the dummy IP to the IP control list.. (10.10.10.1:5080) and now the Taqua accepts calls from SPCE.

I'm not sure if this is solution for you to use with the other providers listed in the thread.

In my attached wireshark cap, it appears that SPCE is sending an ACK back from my dummy address 10.10.10.1 to my gateway.  Thats not right is it?  at this point now, my calls to our gateway dont stay up for longer than 32Seconds then the audio drops out of the call and the person on the PSTN side of the call is disconnected completely, and my SPCE UA appears to have the call up still with no audio until I end the call.



Marc Rys
http://www.tri-lakes.net
http://www.rystec.com

----- Original Message -----
From: "Skyler" <skchopperguy at gmail.com>
To: "Gary Nieboer" <gary at netdiverse.com>
Cc: spce-user at lists.sipwise.com
Sent: Wednesday, March 27, 2013 11:48:44 AM
Subject: Re: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header

My guess (as I've not actually had this issue) is that your contact 
would look something like:

Contact: <sip:ngcp-lb at your.ext.ip.address:5060;ngcpct='sip:127.0.0.1:5080'>

Skyler

On 3/27/2013 9:42 AM, Gary Nieboer wrote:
> I will give a couple of the items advised a try.
>
> Should there be more than one uri in the Contact field?  If so, that may
> be the quick fix.  The contact field I captured a few weeks ago only has
> one, and it is the loopback/proxy address of 127.0.0.1:5080
> <http://127.0.0.1:5080>
>
> -Gary
> --
> Gary Nieboer
>
>
> 877-763-1014 Toll Free  |  801-657-5484 Utah  |  615-800-6946 Tennessee
> | 801-694-6106 Mobile
> gary at netdiverse.com <mailto:gary at netdiverse.com>  | www.netdiverse.com
> <http://www.netdiverse.com/>
>
>
> On Wed, Mar 27, 2013 at 10:31 AM, Skyler <skchopperguy at gmail.com
> <mailto:skchopperguy at gmail.com>> wrote:
>
>     That makes sense, also might just be that the ngcpct sip uri is the
>     last uri read in the contact.
>
>       Maybe Sylvester and Gary could test this out as they have the OP.
>
>     What would a good test look like to base64-encode that ngcpct in a
>     customtt.tt2?
>
>     Skyler
>
>
>     On 3/27/2013 9:07 AM, Andreas Granig wrote:
>
>         Hi,
>
>         On 03/27/2013 04:55 PM, Skyler wrote:
>
>                I created an lo:0 with a private IP address
>             (10.10.xxx.xxx) for lb >
>             b2buas: & proxies > sip:. That may help in this case as its
>             likely the
>             127.0.0.1 which is confusing the upstream proxy.
>
>
>         I've heard of endpoints disliking the SIP uri with the quotes...
>         so one
>         option could be to convert the "ngcpct='sip:127.0.0.1:5080
>         <http://127.0.0.1:5080>'" and
>         base64-encode the "sip:127.0.0.1:5080 <http://127.0.0.1:5080>",
>         so it gets something like
>         "ngcpct=__c2lwOjEyNy4wLjAuMTo1MDgwCg==" instead.
>
>         Andreas
>
>
>     _________________________________________________
>     Spce-user mailing list
>     Spce-user at lists.sipwise.com <mailto:Spce-user at lists.sipwise.com>
>     http://lists.sipwise.com/__listinfo/spce-user
>     <http://lists.sipwise.com/listinfo/spce-user>
>
>

_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com
http://lists.sipwise.com/listinfo/spce-user
-------------- next part --------------
A non-text attachment was scrubbed...
Name: SPCE-OUTBOUND-Taqua-30SecDisc.pcap
Type: application/octet-stream
Size: 1654240 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130327/d0b68338/attachment.pcap>

From skchopperguy at gmail.com  Wed Mar 27 15:53:13 2013
From: skchopperguy at gmail.com (Skyler)
Date: Wed, 27 Mar 2013 12:53:13 -0700
Subject: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
In-Reply-To: <CAMXUi2QrywYL5wEcXk-A2hDx_4tcDgtAcx9wpdxU1LREVwYFtw@mail.gmail.com>
References: <E50575DD48D9B743925AC3BCE6310A4B2436E488@DB3PRD0710MB355.eurprd07.prod.outlook.com>
	<515309C3.1080601@mail.sipwise.com>
	<51530c35.a266b40a.3a4f.ffffb1e3@mx.google.com>
	<5153131A.7060002@mail.sipwise.com> <51531668.2090106@gmail.com>
	<5153192F.4050106@mail.sipwise.com> <51531EE5.1010802@gmail.com>
	<CAMXUi2QV2vOAtpSW_CJU73oy=iScpKV0cx+YdShSk6Gi64_pnA@mail.gmail.com>
	<515322EC.5080908@gmail.com>
	<CAMXUi2QrywYL5wEcXk-A2hDx_4tcDgtAcx9wpdxU1LREVwYFtw@mail.gmail.com>
Message-ID: <51534E29.6090006@gmail.com>

I would say that the invite you are looking at was internal between the 
proxy and lb. its easy to mix them. Would be impossible for any outside 
peer to contact localhost so SPCE would not insert 127.0.0.1 as a 
contact to an outside peer. Are you sure the invite was from your 
external IP to the external peer IP?

Skyler

On 3/27/2013 10:17 AM, Gary Nieboer wrote:
> I found an old pcap with an invite showing the Contact.
>
> That could be the issue.   On the SPCE Invite to the Peer, the only
> thing in the capture is:
>
>     Contact: <sip:127.0.0.1:5080 <http://127.0.0.1:5080>>
>
>
> and nothing in the Contact field about our ip:  x.x.x.x:5060 or with
>   "ngcp-lb" or "ngcpct"
>
>
>
>
> -Gary
> --
> Gary Nieboer
>
>
> 877-763-1014 Toll Free  |  801-657-5484 Utah  |  615-800-6946 Tennessee
> | 801-694-6106 Mobile
> gary at netdiverse.com <mailto:gary at netdiverse.com>  | www.netdiverse.com
> <http://www.netdiverse.com/>
>
>
> On Wed, Mar 27, 2013 at 10:48 AM, Skyler <skchopperguy at gmail.com
> <mailto:skchopperguy at gmail.com>> wrote:
>
>     My guess (as I've not actually had this issue) is that your contact
>     would look something like:
>
>     Contact:
>     <sip:ngcp-lb at your.ext.ip.__address:5060;ngcpct='sip:127.__0.0.1:5080
>     <http://127.0.0.1:5080>'>
>
>     Skyler
>
>
>     On 3/27/2013 9:42 AM, Gary Nieboer wrote:
>
>         I will give a couple of the items advised a try.
>
>         Should there be more than one uri in the Contact field?  If so,
>         that may
>         be the quick fix.  The contact field I captured a few weeks ago
>         only has
>         one, and it is the loopback/proxy address of 127.0.0.1:5080
>         <http://127.0.0.1:5080>
>         <http://127.0.0.1:5080>
>
>         -Gary
>         --
>         Gary Nieboer
>
>
>         877-763-1014 <tel:877-763-1014> Toll Free  | 801-657-5484
>         <tel:801-657-5484> Utah  | 615-800-6946 <tel:615-800-6946> Tennessee
>         | 801-694-6106 <tel:801-694-6106> Mobile
>         gary at netdiverse.com <mailto:gary at netdiverse.com>
>         <mailto:gary at netdiverse.com <mailto:gary at netdiverse.com>>  |
>         www.netdiverse.com <http://www.netdiverse.com>
>         <http://www.netdiverse.com/>
>
>
>
>         On Wed, Mar 27, 2013 at 10:31 AM, Skyler <skchopperguy at gmail.com
>         <mailto:skchopperguy at gmail.com>
>         <mailto:skchopperguy at gmail.com
>         <mailto:skchopperguy at gmail.com>__>> wrote:
>
>              That makes sense, also might just be that the ngcpct sip
>         uri is the
>              last uri read in the contact.
>
>                Maybe Sylvester and Gary could test this out as they have
>         the OP.
>
>              What would a good test look like to base64-encode that
>         ngcpct in a
>              customtt.tt2?
>
>              Skyler
>
>
>              On 3/27/2013 9:07 AM, Andreas Granig wrote:
>
>                  Hi,
>
>                  On 03/27/2013 04:55 PM, Skyler wrote:
>
>                         I created an lo:0 with a private IP address
>                      (10.10.xxx.xxx) for lb >
>                      b2buas: & proxies > sip:. That may help in this
>         case as its
>                      likely the
>                      127.0.0.1 which is confusing the upstream proxy.
>
>
>                  I've heard of endpoints disliking the SIP uri with the
>         quotes...
>                  so one
>                  option could be to convert the
>         "ngcpct='sip:127.0.0.1:5080 <http://127.0.0.1:5080>
>                  <http://127.0.0.1:5080>'" and
>                  base64-encode the "sip:127.0.0.1:5080
>         <http://127.0.0.1:5080> <http://127.0.0.1:5080>",
>
>                  so it gets something like
>                  "ngcpct=____c2lwOjEyNy4wLjAuMTo1MDgwCg==" instead.
>
>                  Andreas
>
>
>              ___________________________________________________
>              Spce-user mailing list
>         Spce-user at lists.sipwise.com <mailto:Spce-user at lists.sipwise.com>
>         <mailto:Spce-user at lists.__sipwise.com
>         <mailto:Spce-user at lists.sipwise.com>>
>         http://lists.sipwise.com/____listinfo/spce-user
>         <http://lists.sipwise.com/__listinfo/spce-user>
>              <http://lists.sipwise.com/__listinfo/spce-user
>         <http://lists.sipwise.com/listinfo/spce-user>>
>
>
>



From skchopperguy at gmail.com  Wed Mar 27 16:05:30 2013
From: skchopperguy at gmail.com (Skyler)
Date: Wed, 27 Mar 2013 13:05:30 -0700
Subject: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
In-Reply-To: <c44cce95-50a6-41bf-9ca3-931723259972@usbsnpzimbra01v.servers.rystec.com>
References: <c44cce95-50a6-41bf-9ca3-931723259972@usbsnpzimbra01v.servers.rystec.com>
Message-ID: <5153510A.8000400@gmail.com>

I would run..

ngrep-sip b -d any -qt -W byline port 5060 >/tmp/ngcpct_check.txt

and make a test call. Then attach the txt to the list.

Skyler

On 3/27/2013 10:30 AM, Marc Rys wrote:
> Ok, Im not sure if this is related, but I've been having struggles with our Taqua Gateway peering to SPCE.  OUr Gateway kept replying back with a 403 Forbidden whenever SPCE tried to send a call to it.
>
> After seeing your dialog about the contact headers, i looked at one our wireshark captures and our Invites were showing the same thing...
> sip:ngcp-lb at your.ext.ip.address:5060;ngcpct='sip:127.0.0.1:5080'
>
> I followed the instructions for the audiocodes fix and changed our dummy address to 10.10.10.1  On our Taqua Gateway I had to add the dummy IP to the IP control list.. (10.10.10.1:5080) and now the Taqua accepts calls from SPCE.
>
> I'm not sure if this is solution for you to use with the other providers listed in the thread.
>
> In my attached wireshark cap, it appears that SPCE is sending an ACK back from my dummy address 10.10.10.1 to my gateway.  Thats not right is it?  at this point now, my calls to our gateway dont stay up for longer than 32Seconds then the audio drops out of the call and the person on the PSTN side of the call is disconnected completely, and my SPCE UA appears to have the call up still with no audio until I end the call.
>
>
>
> Marc Rys
> http://www.tri-lakes.net
> http://www.rystec.com
>
> ----- Original Message -----
> From: "Skyler" <skchopperguy at gmail.com>
> To: "Gary Nieboer" <gary at netdiverse.com>
> Cc: spce-user at lists.sipwise.com
> Sent: Wednesday, March 27, 2013 11:48:44 AM
> Subject: Re: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
>
> My guess (as I've not actually had this issue) is that your contact
> would look something like:
>
> Contact: <sip:ngcp-lb at your.ext.ip.address:5060;ngcpct='sip:127.0.0.1:5080'>
>
> Skyler
>
> On 3/27/2013 9:42 AM, Gary Nieboer wrote:
>> I will give a couple of the items advised a try.
>>
>> Should there be more than one uri in the Contact field?  If so, that may
>> be the quick fix.  The contact field I captured a few weeks ago only has
>> one, and it is the loopback/proxy address of 127.0.0.1:5080
>> <http://127.0.0.1:5080>
>>
>> -Gary
>> --
>> Gary Nieboer
>>
>>
>> 877-763-1014 Toll Free  |  801-657-5484 Utah  |  615-800-6946 Tennessee
>> | 801-694-6106 Mobile
>> gary at netdiverse.com <mailto:gary at netdiverse.com>  | www.netdiverse.com
>> <http://www.netdiverse.com/>
>>
>>
>> On Wed, Mar 27, 2013 at 10:31 AM, Skyler <skchopperguy at gmail.com
>> <mailto:skchopperguy at gmail.com>> wrote:
>>
>>      That makes sense, also might just be that the ngcpct sip uri is the
>>      last uri read in the contact.
>>
>>        Maybe Sylvester and Gary could test this out as they have the OP.
>>
>>      What would a good test look like to base64-encode that ngcpct in a
>>      customtt.tt2?
>>
>>      Skyler
>>
>>
>>      On 3/27/2013 9:07 AM, Andreas Granig wrote:
>>
>>          Hi,
>>
>>          On 03/27/2013 04:55 PM, Skyler wrote:
>>
>>                 I created an lo:0 with a private IP address
>>              (10.10.xxx.xxx) for lb >
>>              b2buas: & proxies > sip:. That may help in this case as its
>>              likely the
>>              127.0.0.1 which is confusing the upstream proxy.
>>
>>
>>          I've heard of endpoints disliking the SIP uri with the quotes...
>>          so one
>>          option could be to convert the "ngcpct='sip:127.0.0.1:5080
>>          <http://127.0.0.1:5080>'" and
>>          base64-encode the "sip:127.0.0.1:5080 <http://127.0.0.1:5080>",
>>          so it gets something like
>>          "ngcpct=__c2lwOjEyNy4wLjAuMTo1MDgwCg==" instead.
>>
>>          Andreas
>>
>>
>>      _________________________________________________
>>      Spce-user mailing list
>>      Spce-user at lists.sipwise.com <mailto:Spce-user at lists.sipwise.com>
>>      http://lists.sipwise.com/__listinfo/spce-user
>>      <http://lists.sipwise.com/listinfo/spce-user>
>>
>>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
>
>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
>



From jchism2 at gmail.com  Wed Mar 27 17:39:41 2013
From: jchism2 at gmail.com (Jeremie Chism)
Date: Wed, 27 Mar 2013 16:39:41 -0500
Subject: [Spce-user] Bug report for 2.7
Message-ID: <82D1110D-5433-4AB7-A974-9CB8B97E2692@gmail.com>

We have a subscriber that provides a sip trunk to a freepbx instance. We have around 12-14 dids pointed at this instance through alias. When you make any change to the alias the cli which was cleared out before to allow different dids to show populates back with the e164 number. I do not remember this happening on 2.6. Is there a way to correct this. 

Sent from my iPhone


From ljallow at utg.edu.gm  Thu Mar 28 03:24:36 2013
From: ljallow at utg.edu.gm (lamarana jallow)
Date: Thu, 28 Mar 2013 15:24:36 +0800
Subject: [Spce-user]  Packet Routing by SIP server
Message-ID: <CADRbaC6+N+zkruw6YcO29k8USdNbtNvxkU50XDBhZa+myRGsxA@mail.gmail.com>

hello guys,
first of all sorry for the title, i dint know the right title to give it.

i read some papers about SIP but i have a question which i cant seem to
find the answer to,
and i am hoping the gurus in here will help me with it, or direct me to a
resource.

i am actually concerned about the individual packets that is sent from one
UA to another.
Lets say  after the connection has been established by the SIP provider,
and the ACK has been
sent to the callee and the media is now between the two UA's   what happens
to the packets, do
they go via the SIP server or not? So in this case when the call has been
established  and the server is down
does it mean that the call will drop?

Thank you for the usual help.





-- 

-- 
Lamarana Jallow
Graduate Assistant
University of The Gambia
Administration Building
Brikama Campus
P.O. Box 3530
Mob:(+220)7019072
The Gambia. West Africa
E-mail:ljallow at utg.edu.gm <http://goog_1591787375/>
/lamjals<http://goog_1591787375/>
@ <http://goog_1591787375/>yahoo.com
Web: www.utg.edu.gm
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130328/c5ab7422/attachment.html>

From jbonilla at sipwise.com  Thu Mar 28 04:37:27 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Thu, 28 Mar 2013 09:37:27 +0100
Subject: [Spce-user] Packet Routing by SIP server
In-Reply-To: <CADRbaC6+N+zkruw6YcO29k8USdNbtNvxkU50XDBhZa+myRGsxA@mail.gmail.com>
References: <CADRbaC6+N+zkruw6YcO29k8USdNbtNvxkU50XDBhZa+myRGsxA@mail.gmail.com>
Message-ID: <20130328093727.27e82ee5@quenya>

El Thu, 28 Mar 2013 15:24:36 +0800
lamarana jallow <ljallow at utg.edu.gm> escribi?:

> hello guys,
> first of all sorry for the title, i dint know the right title to give it.
> 
> i read some papers about SIP but i have a question which i cant seem to
> find the answer to,
> and i am hoping the gurus in here will help me with it, or direct me to a
> resource.
> 
> i am actually concerned about the individual packets that is sent from one
> UA to another.
> Lets say  after the connection has been established by the SIP provider,
> and the ACK has been
> sent to the callee and the media is now between the two UA's   what happens
> to the packets, do
> they go via the SIP server or not?

Yes, they do. You need them lto keep track of the session and close the
accounting when the call is finished.

> So in this case when the call has been
> established  and the server is down
> does it mean that the call will drop?
> 

Yes, it will.


> Thank you for the usual help.
> 

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130328/eb35a380/attachment.asc>

From mpower at entelit.ca  Thu Mar 28 14:25:19 2013
From: mpower at entelit.ca (Maxwell Power)
Date: Thu, 28 Mar 2013 12:25:19 -0600
Subject: [Spce-user] Passing dialed number to subscriber
Message-ID: <005301ce2be1$9abc49f0$d034ddd0$@entelit.ca>

Hello,

 

I have been trying to wrap my head around how to accomplish this for a
little while now. 

 

A subscriber has a their own PBX. We have multiple number aliases setup for
a subscriber. How do I pass the dialled number to the subscribers PBX so it
can route the call appropriately? It seems no matter what I change and mess
around with the dialled number does not show up in any of the headers.

 

I have messed around in the gui but cannot find an easy way to accomplish
this. Am I missing something? Do I have to set each number up as a
subscriber separately?

Cheers,

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130328/95d9b18e/attachment.html>

From jbonilla at sipwise.com  Thu Mar 28 14:45:31 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Thu, 28 Mar 2013 19:45:31 +0100
Subject: [Spce-user] Passing dialed number to subscriber
In-Reply-To: <005301ce2be1$9abc49f0$d034ddd0$@entelit.ca>
References: <005301ce2be1$9abc49f0$d034ddd0$@entelit.ca>
Message-ID: <20130328194531.5cbfc35c@quenya>

El Thu, 28 Mar 2013 12:25:19 -0600
"Maxwell Power" <mpower at entelit.ca> escribi?:

> Hello,
> 
>  
> 
> I have been trying to wrap my head around how to accomplish this for a
> little while now. 
> 
>  
> 
> A subscriber has a their own PBX. We have multiple number aliases setup for
> a subscriber. How do I pass the dialled number to the subscribers PBX so it
> can route the call appropriately? It seems no matter what I change and mess
> around with the dialled number does not show up in any of the headers.
> 
>  
> 
> I have messed around in the gui but cannot find an easy way to accomplish
> this. Am I missing something? Do I have to set each number up as a
> subscriber separately?
> 
> Cheers,
> 

You have an option for that: "e164_to_ruri"

You can enable it at subscriber level and this is the recommended way of
dealing with multialiased pbx systems. With this option, the system will send
the alias number (normalized via rewrite rules of course) and it will fallback
to the sip username in case a 404 is returned by the subscriber.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130328/86438eb0/attachment.asc>

From sn at fest.dk  Thu Mar 28 14:46:45 2013
From: sn at fest.dk (Sylvester Nielsen)
Date: Thu, 28 Mar 2013 18:46:45 +0000
Subject: [Spce-user] Passing dialed number to subscriber
In-Reply-To: <005301ce2be1$9abc49f0$d034ddd0$@entelit.ca>
References: <005301ce2be1$9abc49f0$d034ddd0$@entelit.ca>
Message-ID: <E50575DD48D9B743925AC3BCE6310A4B2436F2BD@DB3PRD0710MB355.eurprd07.prod.outlook.com>

Hi,

You need to check the e164_to_ruri for that subscriber.
By doing this the invite to the PBX will contain the DID number for you to route the inbound calls.

// Sylvester

Fra: spce-user-bounces at lists.sipwise.com [mailto:spce-user-bounces at lists.sipwise.com] P? vegne af Maxwell Power
Sendt: 28. marts 2013 19:25
Til: spce-user at lists.sipwise.com
Emne: [Spce-user] Passing dialed number to subscriber

Hello,

I have been trying to wrap my head around how to accomplish this for a little while now.

A subscriber has a their own PBX. We have multiple number aliases setup for a subscriber. How do I pass the dialled number to the subscribers PBX so it can route the call appropriately? It seems no matter what I change and mess around with the dialled number does not show up in any of the headers.

I have messed around in the gui but cannot find an easy way to accomplish this. Am I missing something? Do I have to set each number up as a subscriber separately?
Cheers,

________________________________
Denne mail er blevet scannet af fest.dk ApS<http://www.fest.dk/> og indeholder ikke virus!
________________________________
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130328/b18fed7b/attachment.html>

From mpower at entelit.ca  Thu Mar 28 14:59:24 2013
From: mpower at entelit.ca (Maxwell Power)
Date: Thu, 28 Mar 2013 12:59:24 -0600
Subject: [Spce-user] Passing dialed number to subscriber
In-Reply-To: <E50575DD48D9B743925AC3BCE6310A4B2436F2BD@DB3PRD0710MB355.eurprd07.prod.outlook.com>
References: <005301ce2be1$9abc49f0$d034ddd0$@entelit.ca>
	<E50575DD48D9B743925AC3BCE6310A4B2436F2BD@DB3PRD0710MB355.eurprd07.prod.outlook.com>
Message-ID: <007201ce2be6$5e1574e0$1a405ea0$@entelit.ca>

Thanks Jon and Sylvester. I knew it was something really simple I was
overlooking. Worked like a charm!

Cheers,



From: Sylvester Nielsen [mailto:sn at fest.dk] 
Sent: March-28-13 12:47
To: Maxwell Power; spce-user at lists.sipwise.com
Subject: SV: [Spce-user] Passing dialed number to subscriber

 

Hi,

 

You need to check the e164_to_ruri for that subscriber.

By doing this the invite to the PBX will contain the DID number for you to
route the inbound calls.

 

// Sylvester

 

Fra: spce-user-bounces at lists.sipwise.com
[mailto:spce-user-bounces at lists.sipwise.com] P? vegne af Maxwell Power
Sendt: 28. marts 2013 19:25
Til: spce-user at lists.sipwise.com
Emne: [Spce-user] Passing dialed number to subscriber

 

Hello,

 

I have been trying to wrap my head around how to accomplish this for a
little while now. 

 

A subscriber has a their own PBX. We have multiple number aliases setup for
a subscriber. How do I pass the dialled number to the subscribers PBX so it
can route the call appropriately? It seems no matter what I change and mess
around with the dialled number does not show up in any of the headers.

 

I have messed around in the gui but cannot find an easy way to accomplish
this. Am I missing something? Do I have to set each number up as a
subscriber separately?

Cheers,

 

  _____  

Denne mail er blevet scannet af  <http://www.fest.dk/> fest.dk ApS og
indeholder ikke virus! 

  _____  

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130328/a6f2eb0c/attachment.html>

From agranig at sipwise.com  Thu Mar 28 15:39:15 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Thu, 28 Mar 2013 20:39:15 +0100
Subject: [Spce-user] Packet Routing by SIP server
In-Reply-To: <CADRbaC6+N+zkruw6YcO29k8USdNbtNvxkU50XDBhZa+myRGsxA@mail.gmail.com>
References: <CADRbaC6+N+zkruw6YcO29k8USdNbtNvxkU50XDBhZa+myRGsxA@mail.gmail.com>
Message-ID: <51549C63.2000308@mail.sipwise.com>

Hi,

On 03/28/2013 08:24 AM, lamarana jallow wrote:
> i am actually concerned about the individual packets that is sent from
> one UA to another.
> Lets say  after the connection has been established by the SIP provider,
> and the ACK has been
> sent to the callee and the media is now between the two UA's   what
> happens to the packets, do
> they go via the SIP server or not? So in this case when the call has
> been established  and the server is down
> does it mean that the call will drop?

Well, depends :)

If you mean the media streams, then on the SPCE there are user 
preferences "always_use_rtpproxy" and "never_use_rtpproxy" for 
subscriber/domain/peer to control whether the media relay on the SPCE 
will be engaged or not. If not, your media goes end-to-end (if you're 
not behind nat), so it won't matter if the SPCE goes down for a short 
time during a call.

However, UAs also (can) do session keep-alives using session timers on 
SIP level during a call, and if the SPCE is down, the UAs will end the 
call if they are not successful.

And finally, if you want to rely on correct billing, you want your SPCE 
to be always up to write an accounting record for the BYE, otherwise you 
won't get a CDR from these calls and can't bill your users.

Andreas



From agranig at sipwise.com  Fri Mar 29 11:36:27 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Fri, 29 Mar 2013 16:36:27 +0100
Subject: [Spce-user] sip:provider CE v2.8 released
Message-ID: <5155B4FB.5090700@mail.sipwise.com>

Hi guys,

I'm excited to announce the new SPCE v2.8. Please check the official 
announcement at

	http://www.sipwise.com/news/announcements/spce-v2_8-release/

for a list of changes.

This version will be our first LTS version, so we will keep back-porting 
fixes and security updates to this version for the next three years. If 
you want something rock-stable for the years to come, you should go with 
that!

After v2.8 we'll switch to v3.0 (due in ~3 months as always) with a 
completely reworked web panel, and we'll switch to a REST interface for 
3rd party provisioning clients, and no more new features will be exposed 
via the SOAP and XMLRPC (and we'll phase it out completely by end of 
this year).

I'd like to thank you all for your valuable feedback and discussions, 
let's keep it going for the next v3.0 with some awesome changes ahead!

Thank you,
Andreas

PS: Please let us know if there goes something wrong in your upgrade or 
installation. It has been tested extensively via automated jobs and also 
manually a lot of times, but if you find something nonetheless, we'll be 
happy to fix it.



From skchopperguy at gmail.com  Fri Mar 29 19:45:24 2013
From: skchopperguy at gmail.com (Skyler)
Date: Fri, 29 Mar 2013 16:45:24 -0700
Subject: [Spce-user] sip:provider CE v2.8 released
In-Reply-To: <5155B4FB.5090700@mail.sipwise.com>
References: <5155B4FB.5090700@mail.sipwise.com>
Message-ID: <51562794.3070006@gmail.com>

Hi,

  Thank you for the LTS and many great updates this release!

  Wondering if Kamailio 4.0 is slated for SPCE 3.0? Will it also be 
back-ported for 2.8-LTS at that time too?

  Very excited on the move to REST interface in provisioning!

Skyler

On 3/29/2013 8:36 AM, Andreas Granig wrote:
> Hi guys,
>
> I'm excited to announce the new SPCE v2.8. Please check the official
> announcement at
>
>      http://www.sipwise.com/news/announcements/spce-v2_8-release/
>
> for a list of changes.
>
> This version will be our first LTS version, so we will keep back-porting
> fixes and security updates to this version for the next three years. If
> you want something rock-stable for the years to come, you should go with
> that!
>
> After v2.8 we'll switch to v3.0 (due in ~3 months as always) with a
> completely reworked web panel, and we'll switch to a REST interface for
> 3rd party provisioning clients, and no more new features will be exposed
> via the SOAP and XMLRPC (and we'll phase it out completely by end of
> this year).
>
> I'd like to thank you all for your valuable feedback and discussions,
> let's keep it going for the next v3.0 with some awesome changes ahead!
>
> Thank you,
> Andreas
>
> PS: Please let us know if there goes something wrong in your upgrade or
> installation. It has been tested extensively via automated jobs and also
> manually a lot of times, but if you find something nonetheless, we'll be
> happy to fix it.
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user



From sipwise at ics-il.net  Sat Mar 30 11:47:59 2013
From: sipwise at ics-il.net (sipwise at ics-il.net)
Date: Sat, 30 Mar 2013 10:47:59 -0500 (CDT)
Subject: [Spce-user] Intro
In-Reply-To: <18585488.5264.1364657228791.JavaMail.mhammett@ThunderFuck>
Message-ID: <26490533.5272.1364658487949.JavaMail.mhammett@ThunderFuck>

I just thought I'd introduce myself to the list. I am looking to migrate to the SIPWise platform. I'll mostly lurk, but I'll surely ask some questions before I move over. 



----- 
Mike Hammett 
Intelligent Computing Solutions 
http://www.ics-il.com 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130330/7642b32b/attachment.html>

From jbonilla at sipwise.com  Sat Mar 30 15:26:23 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Sat, 30 Mar 2013 20:26:23 +0100
Subject: [Spce-user] Intro
In-Reply-To: <26490533.5272.1364658487949.JavaMail.mhammett@ThunderFuck>
References: <18585488.5264.1364657228791.JavaMail.mhammett@ThunderFuck>
	<26490533.5272.1364658487949.JavaMail.mhammett@ThunderFuck>
Message-ID: <20130330202623.2374e1e1@quenya>

El Sat, 30 Mar 2013 10:47:59 -0500 (CDT)
sipwise at ics-il.net escribi?:

> I just thought I'd introduce myself to the list. I am looking to migrate to
> the SIPWise platform. I'll mostly lurk, but I'll surely ask some questions
> before I move over. 
> 
> 


Welcome to the list!

Happy to see that the family is growing :)

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130330/be7bc1ab/attachment.asc>

From agranig at sipwise.com  Sat Mar 30 15:28:26 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Sat, 30 Mar 2013 20:28:26 +0100
Subject: [Spce-user] Intro
In-Reply-To: <26490533.5272.1364658487949.JavaMail.mhammett@ThunderFuck>
References: <26490533.5272.1364658487949.JavaMail.mhammett@ThunderFuck>
Message-ID: <51573CDA.8060604@sipwise.com>

Hi Mike, welcome!

On 03/30/2013 04:47 PM, sipwise at ics-il.net wrote:
> I just thought I'd introduce myself to the list. I am looking to migrate
> to the SIPWise platform. I'll mostly lurk, but I'll surely ask some
> questions before I move over.

Nice! If anything is unclear from the handbook, ask here. Also, if you 
find there is something missing in the handbook, let us know. We're 
trying to constantly improve it, but if you work with it on a day-to-day 
basis, lots of things become obvious... for beginners though, not so much.

We're also welcoming contributions to the handbook, just post new 
sections/chapters to the spce-dev list. We're also working on moving it 
to github, so it gets easy to send pull requests.

Andreas



From sipwise at ics-il.net  Sat Mar 30 15:35:25 2013
From: sipwise at ics-il.net (sipwise at ics-il.net)
Date: Sat, 30 Mar 2013 14:35:25 -0500 (CDT)
Subject: [Spce-user] Provisioning
Message-ID: <8414863.5386.1364672121268.JavaMail.mhammett@ThunderFuck>

The only references I saw in Google about provisioning in SIPWise were to the raw information the end-user could provision their device with. 

Is there any provisioning system built-in? 

If not, what are people using for this? 

What I'm talking about is the end-user ATA fetches a config from the server that is propagated with username\password, server address, etc. 




----- 
Mike Hammett 
Intelligent Computing Solutions 
http://www.ics-il.com 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130330/a1783dc4/attachment.html>

From jchism2 at gmail.com  Sat Mar 30 15:58:24 2013
From: jchism2 at gmail.com (Jeremie Chism)
Date: Sat, 30 Mar 2013 14:58:24 -0500
Subject: [Spce-user] Provisioning
In-Reply-To: <8414863.5386.1364672121268.JavaMail.mhammett@ThunderFuck>
References: <8414863.5386.1364672121268.JavaMail.mhammett@ThunderFuck>
Message-ID: <3B6E8EC8-9CC7-4BD8-83B1-B9BD44442432@gmail.com>

Ata provisioning is not built into sipwise. You would need to develop your own system. We are looking at several options but have not settled on one. 

Sent from my iPhone

On Mar 30, 2013, at 2:35 PM, sipwise at ics-il.net wrote:

> The only references I saw in Google about provisioning in SIPWise were to the raw information the end-user could provision their device with.
> 
> Is there any provisioning system built-in?
> 
> If not, what are people using for this?
> 
> What I'm talking about is the end-user ATA fetches a config from the server that is propagated with username\password, server address, etc.
> 
> 
> 
> -----
> Mike Hammett
> Intelligent Computing Solutions
> http://www.ics-il.com
> 
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130330/b6b3a08d/attachment.html>

From m.rys at tri-lakes.net  Sat Mar 30 17:22:00 2013
From: m.rys at tri-lakes.net (Marc Rys)
Date: Sat, 30 Mar 2013 16:22:00 -0500 (CDT)
Subject: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
In-Reply-To: <5153510A.8000400@gmail.com>
Message-ID: <3eb26bda-cede-43bc-a8f0-20c907977c71@usbsnpzimbra01v.servers.rystec.com>

Sorry for the delay.  Attached is the file as requested.

Marc

----- Original Message -----
From: "Skyler" <skchopperguy at gmail.com>
To: spce-user at lists.sipwise.com
Sent: Wednesday, March 27, 2013 3:05:30 PM
Subject: Re: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header

I would run..

ngrep-sip b -d any -qt -W byline port 5060 >/tmp/ngcpct_check.txt

and make a test call. Then attach the txt to the list.

Skyler

On 3/27/2013 10:30 AM, Marc Rys wrote:
> Ok, Im not sure if this is related, but I've been having struggles with our Taqua Gateway peering to SPCE.  OUr Gateway kept replying back with a 403 Forbidden whenever SPCE tried to send a call to it.
>
> After seeing your dialog about the contact headers, i looked at one our wireshark captures and our Invites were showing the same thing...
> sip:ngcp-lb at your.ext.ip.address:5060;ngcpct='sip:127.0.0.1:5080'
>
> I followed the instructions for the audiocodes fix and changed our dummy address to 10.10.10.1  On our Taqua Gateway I had to add the dummy IP to the IP control list.. (10.10.10.1:5080) and now the Taqua accepts calls from SPCE.
>
> I'm not sure if this is solution for you to use with the other providers listed in the thread.
>
> In my attached wireshark cap, it appears that SPCE is sending an ACK back from my dummy address 10.10.10.1 to my gateway.  Thats not right is it?  at this point now, my calls to our gateway dont stay up for longer than 32Seconds then the audio drops out of the call and the person on the PSTN side of the call is disconnected completely, and my SPCE UA appears to have the call up still with no audio until I end the call.
>
>
>
> Marc Rys
> http://www.tri-lakes.net
> http://www.rystec.com
>
> ----- Original Message -----
> From: "Skyler" <skchopperguy at gmail.com>
> To: "Gary Nieboer" <gary at netdiverse.com>
> Cc: spce-user at lists.sipwise.com
> Sent: Wednesday, March 27, 2013 11:48:44 AM
> Subject: Re: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
>
> My guess (as I've not actually had this issue) is that your contact
> would look something like:
>
> Contact: <sip:ngcp-lb at your.ext.ip.address:5060;ngcpct='sip:127.0.0.1:5080'>
>
> Skyler
>
> On 3/27/2013 9:42 AM, Gary Nieboer wrote:
>> I will give a couple of the items advised a try.
>>
>> Should there be more than one uri in the Contact field?  If so, that may
>> be the quick fix.  The contact field I captured a few weeks ago only has
>> one, and it is the loopback/proxy address of 127.0.0.1:5080
>> <http://127.0.0.1:5080>
>>
>> -Gary
>> --
>> Gary Nieboer
>>
>>
>> 877-763-1014 Toll Free  |  801-657-5484 Utah  |  615-800-6946 Tennessee
>> | 801-694-6106 Mobile
>> gary at netdiverse.com <mailto:gary at netdiverse.com>  | www.netdiverse.com
>> <http://www.netdiverse.com/>
>>
>>
>> On Wed, Mar 27, 2013 at 10:31 AM, Skyler <skchopperguy at gmail.com
>> <mailto:skchopperguy at gmail.com>> wrote:
>>
>>      That makes sense, also might just be that the ngcpct sip uri is the
>>      last uri read in the contact.
>>
>>        Maybe Sylvester and Gary could test this out as they have the OP.
>>
>>      What would a good test look like to base64-encode that ngcpct in a
>>      customtt.tt2?
>>
>>      Skyler
>>
>>
>>      On 3/27/2013 9:07 AM, Andreas Granig wrote:
>>
>>          Hi,
>>
>>          On 03/27/2013 04:55 PM, Skyler wrote:
>>
>>                 I created an lo:0 with a private IP address
>>              (10.10.xxx.xxx) for lb >
>>              b2buas: & proxies > sip:. That may help in this case as its
>>              likely the
>>              127.0.0.1 which is confusing the upstream proxy.
>>
>>
>>          I've heard of endpoints disliking the SIP uri with the quotes...
>>          so one
>>          option could be to convert the "ngcpct='sip:127.0.0.1:5080
>>          <http://127.0.0.1:5080>'" and
>>          base64-encode the "sip:127.0.0.1:5080 <http://127.0.0.1:5080>",
>>          so it gets something like
>>          "ngcpct=__c2lwOjEyNy4wLjAuMTo1MDgwCg==" instead.
>>
>>          Andreas
>>
>>
>>      _________________________________________________
>>      Spce-user mailing list
>>      Spce-user at lists.sipwise.com <mailto:Spce-user at lists.sipwise.com>
>>      http://lists.sipwise.com/__listinfo/spce-user
>>      <http://lists.sipwise.com/listinfo/spce-user>
>>
>>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
>
>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
>

_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com
http://lists.sipwise.com/listinfo/spce-user
-------------- next part --------------
An embedded and charset-unspecified text was scrubbed...
Name: ngcpct_check.txt
URL: <http://lists.sipwise.com/mailman/private/spce-user_lists.sipwise.com/attachments/20130330/3d24f986/attachment.txt>

From skchopperguy at gmail.com  Sun Mar 31 13:33:14 2013
From: skchopperguy at gmail.com (Skyler)
Date: Sun, 31 Mar 2013 10:33:14 -0700
Subject: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
In-Reply-To: <3eb26bda-cede-43bc-a8f0-20c907977c71@usbsnpzimbra01v.servers.rystec.com>
References: <3eb26bda-cede-43bc-a8f0-20c907977c71@usbsnpzimbra01v.servers.rystec.com>
Message-ID: <5158735A.1030308@gmail.com>

Hi,

  Seems your call is dropping due to confused signalling. You should not 
need to input the 10.10.10.1 into your Taqua gw control list. One thing 
I see which does not look correct is the 100 trying response after the 
initial invite (spce > peer). as below:

U 2013/03/30 16:13:10.848816 208.73.198.12:5060 -> 208.73.199.250:5060
SIP/2.0 100 Trying'
Via: SIP/2.0/UDP 
208.73.199.250;branch=z9hG4bK3642.4b4263c869b06a079cb58096dc4f3039.0,SIP/2.0/UDP 
10.10.10.1:5080;branch=z9hG4bKPfekkaGS;rport=5080'
From: 
<sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
CSeq: 10 INVITE'
Content-Length: 0'
'

  Notice this first response from the peer contains a single Via having 
'SIP/2.0/UDP' twice. As though (assuming) the Taqua gw is appending the 
ngcpct= to the via. This could be a case of needing to code the quirks 
of this Taqua into spce to overcome its (Taqua) needs for signalling.

What is your spce version?
Latest updates?
Do you have any customtt.tt2 files in your lb or proxy?

  My first inclination in troubleshooting is to rule out any individual 
customtt files which may have accidentally broken something in spce. Try 
moving customtt files to /tmp then ngcpcfg apply, do a test-call again 
to see if (or which) customtt may be causing this.

  This is only my suggestion, I cannot duplicate or test this behavior.

Skyler

On 3/30/2013 2:22 PM, Marc Rys wrote:
> Sorry for the delay.  Attached is the file as requested.
>
> Marc
>
> ----- Original Message -----
> From: "Skyler" <skchopperguy at gmail.com>
> To: spce-user at lists.sipwise.com
> Sent: Wednesday, March 27, 2013 3:05:30 PM
> Subject: Re: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
>
> I would run..
>
> ngrep-sip b -d any -qt -W byline port 5060 >/tmp/ngcpct_check.txt
>
> and make a test call. Then attach the txt to the list.
>
> Skyler
>
> On 3/27/2013 10:30 AM, Marc Rys wrote:
>> Ok, Im not sure if this is related, but I've been having struggles with our Taqua Gateway peering to SPCE.  OUr Gateway kept replying back with a 403 Forbidden whenever SPCE tried to send a call to it.
>>
>> After seeing your dialog about the contact headers, i looked at one our wireshark captures and our Invites were showing the same thing...
>> sip:ngcp-lb at your.ext.ip.address:5060;ngcpct='sip:127.0.0.1:5080'
>>
>> I followed the instructions for the audiocodes fix and changed our dummy address to 10.10.10.1  On our Taqua Gateway I had to add the dummy IP to the IP control list.. (10.10.10.1:5080) and now the Taqua accepts calls from SPCE.
>>
>> I'm not sure if this is solution for you to use with the other providers listed in the thread.
>>
>> In my attached wireshark cap, it appears that SPCE is sending an ACK back from my dummy address 10.10.10.1 to my gateway.  Thats not right is it?  at this point now, my calls to our gateway dont stay up for longer than 32Seconds then the audio drops out of the call and the person on the PSTN side of the call is disconnected completely, and my SPCE UA appears to have the call up still with no audio until I end the call.
>>
>>
>>
>> Marc Rys
>> http://www.tri-lakes.net
>> http://www.rystec.com
>>
>> ----- Original Message -----
>> From: "Skyler" <skchopperguy at gmail.com>
>> To: "Gary Nieboer" <gary at netdiverse.com>
>> Cc: spce-user at lists.sipwise.com
>> Sent: Wednesday, March 27, 2013 11:48:44 AM
>> Subject: Re: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
>>
>> My guess (as I've not actually had this issue) is that your contact
>> would look something like:
>>
>> Contact: <sip:ngcp-lb at your.ext.ip.address:5060;ngcpct='sip:127.0.0.1:5080'>
>>
>> Skyler
>>
>> On 3/27/2013 9:42 AM, Gary Nieboer wrote:
>>> I will give a couple of the items advised a try.
>>>
>>> Should there be more than one uri in the Contact field?  If so, that may
>>> be the quick fix.  The contact field I captured a few weeks ago only has
>>> one, and it is the loopback/proxy address of 127.0.0.1:5080
>>> <http://127.0.0.1:5080>
>>>
>>> -Gary
>>> --
>>> Gary Nieboer
>>>
>>>
>>> 877-763-1014 Toll Free  |  801-657-5484 Utah  |  615-800-6946 Tennessee
>>> | 801-694-6106 Mobile
>>> gary at netdiverse.com <mailto:gary at netdiverse.com>  | www.netdiverse.com
>>> <http://www.netdiverse.com/>
>>>
>>>
>>> On Wed, Mar 27, 2013 at 10:31 AM, Skyler <skchopperguy at gmail.com
>>> <mailto:skchopperguy at gmail.com>> wrote:
>>>
>>>       That makes sense, also might just be that the ngcpct sip uri is the
>>>       last uri read in the contact.
>>>
>>>         Maybe Sylvester and Gary could test this out as they have the OP.
>>>
>>>       What would a good test look like to base64-encode that ngcpct in a
>>>       customtt.tt2?
>>>
>>>       Skyler
>>>
>>>
>>>       On 3/27/2013 9:07 AM, Andreas Granig wrote:
>>>
>>>           Hi,
>>>
>>>           On 03/27/2013 04:55 PM, Skyler wrote:
>>>
>>>                  I created an lo:0 with a private IP address
>>>               (10.10.xxx.xxx) for lb >
>>>               b2buas: & proxies > sip:. That may help in this case as its
>>>               likely the
>>>               127.0.0.1 which is confusing the upstream proxy.
>>>
>>>
>>>           I've heard of endpoints disliking the SIP uri with the quotes...
>>>           so one
>>>           option could be to convert the "ngcpct='sip:127.0.0.1:5080
>>>           <http://127.0.0.1:5080>'" and
>>>           base64-encode the "sip:127.0.0.1:5080 <http://127.0.0.1:5080>",
>>>           so it gets something like
>>>           "ngcpct=__c2lwOjEyNy4wLjAuMTo1MDgwCg==" instead.
>>>
>>>           Andreas
>>>
>>>
>>>       _________________________________________________
>>>       Spce-user mailing list
>>>       Spce-user at lists.sipwise.com <mailto:Spce-user at lists.sipwise.com>
>>>       http://lists.sipwise.com/__listinfo/spce-user
>>>       <http://lists.sipwise.com/listinfo/spce-user>
>>>
>>>
>>
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/listinfo/spce-user
>>
>>
>>
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/listinfo/spce-user
>>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
>
>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
>



From agranig at sipwise.com  Fri Mar  1 03:07:43 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Fri, 01 Mar 2013 09:07:43 +0100
Subject: [Spce-user] inbound rewriting rule
In-Reply-To: <512FD66A.6080501@anutel.ca>
References: <060FD34D-F154-43C1-A202-8A98286074BB@optionsdsl.ca>
	<512E75CB.7060309@anutel.ca>
	<CEEE718799582B47A55D7897DCB264FC280134@bvusaex.bestvoipusa.local>
	<512FD66A.6080501@anutel.ca>
Message-ID: <513061CF.80902@sipwise.com>

Hi,

On 02/28/2013 11:12 PM, Blagoja wrote:
> This is the line where i can see the call entering the server.
> New request - M=ACK R=sip:127.0.0.1:5080
> F=sip:025101165 at X.X.X.X:5060;user=phone;cpc=ordinary
> T=sip:5101199 at X.X.X.X:5060;user=phone IP=X.X.X.X:5060 (127.0.0.1:5060)
> ID=418ebe9cl43yuf9oa at 172.16.17.10
>   My rulle is : ^([1-9][0-9]+)$ and i believe is muching the number 5101199
> I tried instead of the above rulle with the number but no luck.
> Where and how the server is geting incoming did i cant findout

The ACK is the peer acknowledging the 200-ok of the CE, so this doesn't 
provide any suitable information. You need to look for INVITE messages.

Andreas



From jbonilla at sipwise.com  Fri Mar  1 04:30:21 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Fri, 1 Mar 2013 10:30:21 +0100
Subject: [Spce-user] inbound PSTN peer routing to local sip uri
In-Reply-To: <013501ce15b3$90327c00$b0977400$@murray@skyracktelecom.com>
References: <013501ce15b3$90327c00$b0977400$@murray@skyracktelecom.com>
Message-ID: <20130301103021.4a56afaf@quenya>

El Thu, 28 Feb 2013 13:00:19 -0000
"John Murray" <john.murray at skyracktelecom.com> escribi?:

> Hello,
> 
>  
> 
> I am trying to get calls coming from a PSTN peer (it's a PBX) to route to a
> local user with the sip uri.
> 
> The local user is a001002 at skyrack.net which is in the To: field but SPCE
> doesn't find it as a local user.
> 
>  
> 
> The RURI contains the user part but the domain part is the ip address of our
> server. Can I route this?
>
> The incoming INVITE
> 
> 
> Feb 28 11:55:43 skt-salem02 /usr/sbin/kamailio[2499]: INFO: <script>: New
> request - M=INVITE R=sip:a001002 at 100.100.100.182:5060
> 
> Feb 28 11:55:43 skt-salem02 /usr/sbin/kamailio[2499]: INFO: <script>: Callee
> is not local - R=sip:a001002 at 100.100.100.182:5060
> ID=BW115543276280213-1824721248 at 192.168.30.10
> 
>  

That should work. I usually route based on the username but as subscriber.
There shouldn't be any differences if the call comes from a peer.

You sure that you have an 'a001002' subscriber?

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130301/22e8ff5e/attachment-0001.asc>

From jbonilla at sipwise.com  Fri Mar  1 04:38:28 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Fri, 1 Mar 2013 10:38:28 +0100
Subject: [Spce-user] SOAP array problem
In-Reply-To: <5119FF27.8020107@hfcnetwork.hu>
References: <5119FF27.8020107@hfcnetwork.hu>
Message-ID: <20130301103828.3b3b31ce@quenya>

El Tue, 12 Feb 2013 09:36:55 +0100
HFC-Network Kft - Szab? J?nos <szabojanos at hfcnetwork.hu> escribi?:

> Hi all!
> 
> I want to create a call forward set with a hunting group via SOAP using 
> php with nuSOAP, but I always get this answer:
> "*[faultcode] => soap:Client [faultstring] => Application failed during 
> request deserialization: Unspecified namespace for type 
> 'unnamed_struct_use_soapval[1]' *"
> 
> Am I have to create a special declaration to use this function?
> 
> 

hi

I see that no one answered your question. Did you manage to solve the issue?

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130301/3734837a/attachment-0001.asc>

From john.murray at skyracktelecom.com  Fri Mar  1 04:52:42 2013
From: john.murray at skyracktelecom.com (John Murray)
Date: Fri, 1 Mar 2013 09:52:42 -0000
Subject: [Spce-user] inbound PSTN peer routing to local sip uri
In-Reply-To: <20130301103021.4a56afaf@quenya>
References: <013501ce15b3$90327c00$b0977400$@murray@skyracktelecom.com>
	<20130301103021.4a56afaf@quenya>
Message-ID: <51307a6a.8665b40a.4f88.ffffa26e@mx.google.com>

Jon hi,

Yes it looks like the subscriber is there.

>From admin interface:

SIP URI: a001002 at skyrack.net

And from mysql:
mysql> use kamailio;
Database changed
mysql> select * from subscriber where username='a001002';
+----+----------+-------------+----------+---------------+----------------------------------+----------------------------------+------+--------------------------------------+----------+---------------------+
| id | username | domain      | password | email_address | ha1                              | ha1b                             | rpid | uuid                                 | timezone | datetime_created    |
+----+----------+-------------+----------+---------------+----------------------------------+----------------------------------+------+--------------------------------------+----------+---------------------+
| 21 | a001002  | skyrack.net | xxxxxxxx |               | 90370930c7c47557f23cb76c4bce0f80 | e64a7fc0f08bc93569e556bc18df0b38 | NULL | 3978f230-c438-4e81-b6a3-7ddb90324777 |          | 2013-02-14 11:52:42 |
+----+----------+-------------+----------+---------------+----------------------------------+----------------------------------+------+--------------------------------------+----------+---------------------+
1 row in set (0.00 sec)

Thanks

John
-----Original Message-----
From: spce-user-bounces at lists.sipwise.com [mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of Jon Bonilla (Manwe)
Sent: 01 March 2013 09:30
To: spce-user at lists.sipwise.com
Subject: Re: [Spce-user] inbound PSTN peer routing to local sip uri

El Thu, 28 Feb 2013 13:00:19 -0000
"John Murray" <john.murray at skyracktelecom.com> escribi?:

> Hello,
> 
>  
> 
> I am trying to get calls coming from a PSTN peer (it's a PBX) to route 
> to a local user with the sip uri.
> 
> The local user is a001002 at skyrack.net which is in the To: field but 
> SPCE doesn't find it as a local user.
> 
>  
> 
> The RURI contains the user part but the domain part is the ip address 
> of our server. Can I route this?
>
> The incoming INVITE
> 
> 
> Feb 28 11:55:43 skt-salem02 /usr/sbin/kamailio[2499]: INFO: <script>: 
> New request - M=INVITE R=sip:a001002 at 100.100.100.182:5060
> 
> Feb 28 11:55:43 skt-salem02 /usr/sbin/kamailio[2499]: INFO: <script>: 
> Callee is not local - R=sip:a001002 at 100.100.100.182:5060
> ID=BW115543276280213-1824721248 at 192.168.30.10
> 
>  

That should work. I usually route based on the username but as subscriber.
There shouldn't be any differences if the call comes from a peer.

You sure that you have an 'a001002' subscriber?





From agranig at sipwise.com  Fri Mar  1 06:11:14 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Fri, 01 Mar 2013 12:11:14 +0100
Subject: [Spce-user] inbound PSTN peer routing to local sip uri
In-Reply-To: <51307a6a.8665b40a.4f88.ffffa26e@mx.google.com>
References: <013501ce15b3$90327c00$b0977400$@murray@skyracktelecom.com>
	<20130301103021.4a56afaf@quenya>
	<51307a6a.8665b40a.4f88.ffffa26e@mx.google.com>
Message-ID: <51308CD2.7040802@sipwise.com>

Hi,
On 03/01/2013 10:52 AM, John Murray wrote:
> SIP URI: a001002 at skyrack.net

Well, the problem here is that the call comes in as 
sip:a001002 at 100.100.100.182, but the subscriber is sip:a001002 at skyrack.net.

There is currently no way to also change the domain part via rewrite 
rules, so you'd have to do that hardcoded in the kamailio-lb config, 
like this:

# after the whole loose-route block:
if($rd == "100.100.100.182") {
   $rd = "skyrack.net";
}

You can use it more elegantly of course by using the template values of 
network.yml, but that's basically the idea for now.

Andreas



From szabojanos at hfcnetwork.hu  Fri Mar  1 06:12:53 2013
From: szabojanos at hfcnetwork.hu (=?ISO-8859-1?Q?HFC-Network_Kft_-_Szab=F3_J=E1nos?=)
Date: Fri, 01 Mar 2013 12:12:53 +0100
Subject: [Spce-user] SOAP array problem
In-Reply-To: <20130301103828.3b3b31ce@quenya>
References: <5119FF27.8020107@hfcnetwork.hu> <20130301103828.3b3b31ce@quenya>
Message-ID: <51308D35.4070804@hfcnetwork.hu>

Hi,

No, it's still under investigation.

?dv?zlettel,

Szab? J?nos
HFC-Network Kft

E-mail:  szabojanos at hfcnetwork.hu
Telefon: +36-20/344-4449

HFC Logo

2013.03.01. 10:38 keltez?ssel, Jon Bonilla (Manwe) ?rta:
> El Tue, 12 Feb 2013 09:36:55 +0100
> HFC-Network Kft - Szab? J?nos <szabojanos at hfcnetwork.hu> escribi?:
>
>> Hi all!
>>
>> I want to create a call forward set with a hunting group via SOAP using
>> php with nuSOAP, but I always get this answer:
>> "*[faultcode] => soap:Client [faultstring] => Application failed during
>> request deserialization: Unspecified namespace for type
>> 'unnamed_struct_use_soapval[1]' *"
>>
>> Am I have to create a special declaration to use this function?
>>
>>
> hi
>
> I see that no one answered your question. Did you manage to solve the issue?
>
>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130301/a1290b94/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: hfc_alairas.jpg
Type: image/jpeg
Size: 13079 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130301/a1290b94/attachment-0001.jpg>

From agranig at sipwise.com  Fri Mar  1 06:21:14 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Fri, 01 Mar 2013 12:21:14 +0100
Subject: [Spce-user] inbound PSTN peer routing to local sip uri
In-Reply-To: <51308CD2.7040802@sipwise.com>
References: <013501ce15b3$90327c00$b0977400$@murray@skyracktelecom.com>
	<20130301103021.4a56afaf@quenya>
	<51307a6a.8665b40a.4f88.ffffa26e@mx.google.com>
	<51308CD2.7040802@sipwise.com>
Message-ID: <51308F2A.2080204@sipwise.com>

Another way would be to disable multi-domain handling at all by setting 
kamailio.proxy.ignore_auth_realm to "yes" in config.yml.

That way, the domain part will be ignored completely for authentication 
and also for looking up the callee.

Not sure if you want that though, it's actually a work-around for 
scenarios where lots of subscribers with arbitrary domains are deployed 
in your environment, and you still want to handle all of them on the 
platform without provisioning them within these arbitrary domains.

Andreas

On 03/01/2013 12:11 PM, Andreas Granig wrote:
> Hi,
> On 03/01/2013 10:52 AM, John Murray wrote:
>> SIP URI: a001002 at skyrack.net
>
> Well, the problem here is that the call comes in as
> sip:a001002 at 100.100.100.182, but the subscriber is sip:a001002 at skyrack.net.
>
> There is currently no way to also change the domain part via rewrite
> rules, so you'd have to do that hardcoded in the kamailio-lb config,
> like this:
>
> # after the whole loose-route block:
> if($rd == "100.100.100.182") {
>    $rd = "skyrack.net";
> }
>
> You can use it more elegantly of course by using the template values of
> network.yml, but that's basically the idea for now.
>
> Andreas
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user



From blagojaanusev at anutel.ca  Fri Mar  1 07:39:27 2013
From: blagojaanusev at anutel.ca (Blagoja)
Date: Fri, 01 Mar 2013 07:39:27 -0500
Subject: [Spce-user]   inbound rewriting rule
In-Reply-To: <CEEE718799582B47A55D7897DCB264FC280153@bvusaex.bestvoipusa.local>
References: <512FD9E2.3060003@anutel.ca>
	<CEEE718799582B47A55D7897DCB264FC28014A@bvusaex.bestvoipusa.local>
	<7B51A6CD-1098-44DE-B67B-35688C683785@anutel.ca>
	<CEEE718799582B47A55D7897DCB264FC280153@bvusaex.bestvoipusa.local>
Message-ID: <5130A17F.9060502@anutel.ca>

On 2/28/2013 6:11 PM, Kevin Masse wrote:
> That should still be ok.  Sipwise will see that the alias_number or
> E.164 number assigned to the Subscriber will match and stay local and
> not forward off to an Asterisk box.
> Kevin
>
>
> -----Original Message-----
> From: Blagoja [mailto:blagojaanusev at anutel.ca]
> Sent: Thursday, February 28, 2013 6:08 PM
> To: Kevin Masse
> Cc: <spce-user at lists.sipwise.com>
> Subject: [Spce-user] inbound rewriting rule
>
> Thanks for reply
> There is no asterisk involved i'm trying to route the call to local user
> and the number in my case is 7 digits eg. 510 1199
>
>
> Sent from my iPhone
>
> On 2013-02-28, at 5:35 PM, "Kevin Masse" <kmasse at questblue.com> wrote:
>
>> Good evening:
>>
>> Try to think of it this way:
>>
>> 15551231234 ->Your PSTN Carrier--> inbound call SIPWISE Outbound
> Rewrite
>> Rules for Callee ^(1|)([1-9][0-9][0-9]+)$               \2
>> -->  SIP Subscriber Asterisk Box where the call will go:
>> This will take a call inbound for your carrier like this 15551231234
> and
>> send it to your asterisk box (Subscriber) as 5551231234  The pattern
>> that your Asterisk Box is looking for is 5551231234 then send the call
>> to a destination.  Notice it dropped the 1 for the destination.
>>
>>
>> Let me know if this helps any.
>>
>> Kevin
>>
>>
>> -----Original Message-----
>> From: spce-user-bounces at lists.sipwise.com
>> [mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of Blagoja
>> Sent: Thursday, February 28, 2013 5:28 PM
>> To: spce-user at lists.sipwise.com
>> Subject: [Spce-user] inbound rewriting rule
>>
>> Hello,
>> Thanks for your help I start to understand but i'm stuck again
>>
>> My peer authenticate to remote server so when i call the number call
> is
>> entering my server and disconect. I did rewrite rulle but in
>> kamailio.proxy.log file saying No matching rewrite rules for 'ngcp-lb'
>> found
>>
>> This is the line where i can see the call entering the server.
>> New request - M=ACK R=sip:127.0.0.1:5080
>> F=sip:025101165 at X.X.X.X:5060;user=phone;cpc=ordinary
>> T=sip:5101199 at X.X.X.X:5060;user=phone IP=X.X.X.X:5060 (127.0.0.1:5060)
>> ID=418ebe9cl43yuf9oa at 172.16.17.10
>>   My rulle is : ^([1-9][0-9]+)$ and i believe is muching the number
>> 5101199 I tried instead of the above rulle with the number but no
> luck.
>> Where and how the server is geting incoming did i cant findout
>>
>> Regards.
>>
>>
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/listinfo/spce-user
>>
>
         I think that SCPE is not reading dialed incomung  number 
properly. The number from my provider is 5101199 and the rulle I set for 
that is ^([1-9][0-9]+)$
I had this problem in asterisk when provider is not sending did number 
in sip header . I understand that asterisk is diferent from spce but if 
I can tell somehow to spce how to get  the did like i did in asterisk. 
This is the link for asterisk to help understend what is my problem.
http://www.freepbx.org/support/documentation/howtos/how-to-get-the-did-of-a-sip-trunk-when-the-provider-doesnt-send-it-and-

Blagoja



From blagojaanusev at anutel.ca  Fri Mar  1 07:57:32 2013
From: blagojaanusev at anutel.ca (Blagoja)
Date: Fri, 01 Mar 2013 07:57:32 -0500
Subject: [Spce-user]  inbound rewriting rule
In-Reply-To: <513061CF.80902@sipwise.com>
References: <513061CF.80902@sipwise.com>
Message-ID: <5130A5BC.3010208@anutel.ca>

How do I do that. How do I look in to INVITE messages



From gary at netdiverse.com  Fri Mar  1 11:34:31 2013
From: gary at netdiverse.com (Gary Nieboer)
Date: Fri, 1 Mar 2013 09:34:31 -0700
Subject: [Spce-user] RTCP, Media Monitoring, Media Stats, MOS, etc..
Message-ID: <CAMXUi2TBTRXLeYm=x=rUOSdBVzvS2dbWDyymsiLrCWi6RbLcAA@mail.gmail.com>

This is a request for an update on media monitoring / performance stats,
RTCP, etc...

Although it would be great to have a reporting engine to capture
information on each call and provide summary stats and details per call, it
will still be good if SPCE stores jitter packet loss, etc.. for each leg of
a call that can be found in a log somewhere. If there is a way to see real
time stats on multiple legs like asterisk >sip show channelstats, that
would be great too.

What is available with SPCE?

Thanks,

Gary

>I saw a message #3 on May 31, 2012,  Vol 18   Issue 44
>
>Message: 3
>Hi Matthew,
>
>On 05/31/2012 12:33 PM, Matthew Ogden wrote:
>> There were som replies to my stream monitor, to use pcap etc, which is
>> helpful, but this is a specific case of being able to view or report on
>> jitter/packet loss per client, or per peer etc.
>
>Not yet. I had an interesting discussion with other kamailio developers
>last week where we decided to redesign the rtpproxy control protocol,
>which would give us the flexibility to pass back from mediaproxy-ng to
>the SIP proxy some RTP statistics when ending a media session, which can
>then for example be added as a SIP header or be processed in any other way.
>
>It won't be ready for prime time before late summer, I'd say. Sorry.
>
>Andreas
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130301/7c38f811/attachment-0001.html>

From blagojaanusev at anutel.ca  Fri Mar  1 16:07:07 2013
From: blagojaanusev at anutel.ca (Blagoja)
Date: Fri, 01 Mar 2013 16:07:07 -0500
Subject: [Spce-user] Inbound DID in SIP header
In-Reply-To: <CEEE718799582B47A55D7897DCB264FC280134@bvusaex.bestvoipusa.local>
References: <060FD34D-F154-43C1-A202-8A98286074BB@optionsdsl.ca>
	<512E75CB.7060309@anutel.ca>
	<CEEE718799582B47A55D7897DCB264FC280134@bvusaex.bestvoipusa.local>
Message-ID: <5131187B.9040208@anutel.ca>



I think that SCPE is not reading dialed incoming  number
properly. The number from my provider is 5101199 and the rulle I set for
  ^([1-9][0-9]+)$
But as I can see in kamailio-proxy.log spce is not reding DID number so there is nothing to rewrite or route.
This is the part of the log where spce is treating that call and in BOLD where is not finding any number to rewrite

Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: New request - M=INVITE R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' F=sip:025101165 at 217.16.67.5:5060;user=phone;cpc=ordinary T=sip:5101199 at 217.16.67.5:5060;user=phone IP=217.16.67.5:5060 (127.0.0.1:5060) ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Call from PSTN - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Load peer preferences for peer host '1' - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Load dialplan IDs for peer host '1' - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Fetched dialplan IDs caller_in='1', callee_in='2' - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Dialog set mark total - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Dialog set mark peer to '1' - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: IP authorization for caller not provisioned, allow call - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: User-Provided CLI '025101165' taken from PAI - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Network-Provided CLI '025101165' taken from PAI - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Applying caller-in peer rewrite rules on user-provided CLI using dpid '1' - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Rewriting user-provided CLI '025101165' to '5101165' - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Applying caller-in peer rewrite rules on network-provided CLI using dpid '1' - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Rewriting network-provided CLI '025101165' to '5101165' - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Setting '5101165 at 172.16.17.10' as initiating user-provided CLI - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Setting '5101165 at 172.16.17.10' as initiating network-provided CLI - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
_*Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Searching for callee - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10*__*
*__*Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Applying callee-in peer rewrite rules using dpid '2' - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10*__*
*__*Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: No matching rewrite rules for 'ngcp-lb' found - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10*__*
*__*Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Clear callee preferences - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10*__*
*__*Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Callee is not local - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10*__*
*_Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Check Administrative NCOS level - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Check User NCOS level - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Checking URI 'ngcp-lb at 24.150.194.162' and CLI 'ngcp-lb' against block out lists - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Admin caller block mode is 'open' - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Caller block mode is 'open' - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Call to unknown user - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Accounting failed request for uuid '0' - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Setting acc source-leg for uuid '0': '0|025101165|217.16.67.5|5101165|||<null>|||0|call|<null>|1362171203.778528' - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11992]: INFO: <script>: Setting acc destination-leg for uuid '0': '0|||<null>|ngcp-lb|0|ngcp-lb|24.150.194.162|ngcp-lb|24.150.194.162' - R=sip:ngcp-lb at 24.150.194.162:5060;ngcpct='sip:127.0.0.1:5080' ID=41a5d5f337v18e8j5 at 172.16.17.10
Mar  1 15:53:23 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[11991]: INFO: <script>: New request - M=ACK R=sip:127.0.0.1:5080 F=sip:025101165 at 217.16.67.5:5060;user=phone;cpc=ordinary T=sip:5101199 at 217.16.67.5:5060;user=phone IP=217.16.67.5:5060 (127.0.0.1:5060) ID=41a5d5f337v18e8j5 at 172.16.17.10

Can anybody helpme how to pach this

  I had this problem in asterisk when provider is not sending did number
in sip header . I understand that asterisk is diferent from spce but if
I can tell somehow to spce how to get  the DID number when provider is not sending in header.
This is the link for asterisk to help understend what is my problem.
http://www.freepbx.org/support/documentation/howtos/how-to-get-the-did-of-a-sip-trunk-when-the-provider-doesnt-send-it-and-


Blagoja

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130301/defe63c4/attachment-0001.html>

From jchism2 at gmail.com  Fri Mar  1 16:37:41 2013
From: jchism2 at gmail.com (Jeremie Chism)
Date: Fri, 1 Mar 2013 15:37:41 -0600
Subject: [Spce-user] SMS
Message-ID: <EBB6CE73-9366-45FD-8E9E-8585458FA2B9@gmail.com>

Does sipwise support SMS and if so is there something that needs to be changed in the config to make it work. 

Sent from my iPhone


From jbonilla at sipwise.com  Fri Mar  1 16:46:15 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Fri, 1 Mar 2013 22:46:15 +0100
Subject: [Spce-user] SMS
In-Reply-To: <EBB6CE73-9366-45FD-8E9E-8585458FA2B9@gmail.com>
References: <EBB6CE73-9366-45FD-8E9E-8585458FA2B9@gmail.com>
Message-ID: <20130301224615.1255f743@quenya>

El Fri, 1 Mar 2013 15:37:41 -0600
Jeremie Chism <jchism2 at gmail.com> escribi?:

> Does sipwise support SMS and if so is there something that needs to be
> changed in the config to make it work. 


Could you tell us what exactly you mean by supporting sms? I guess the answer
is no, but sending an http request on events to a sms gateway shouldn't be too
difficult.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130301/584c21d1/attachment-0001.asc>

From jchism2 at gmail.com  Fri Mar  1 18:11:40 2013
From: jchism2 at gmail.com (Jeremie Chism)
Date: Fri, 1 Mar 2013 17:11:40 -0600
Subject: [Spce-user] SMS
In-Reply-To: <20130301224615.1255f743@quenya>
References: <EBB6CE73-9366-45FD-8E9E-8585458FA2B9@gmail.com>
	<20130301224615.1255f743@quenya>
Message-ID: <CAG-pSShJ3mf8LdKKfwYjTj3=NyZ0gm_NkOBBBvmSin08MEYFrQ@mail.gmail.com>

Yes it is sent by sip message.

On Friday, March 1, 2013, Jon Bonilla wrote:

> El Fri, 1 Mar 2013 15:37:41 -0600
> Jeremie Chism <jchism2 at gmail.com <javascript:;>> escribi?:
>
> > Does sipwise support SMS and if so is there something that needs to be
> > changed in the config to make it work.
>
>
> Could you tell us what exactly you mean by supporting sms? I guess the
> answer
> is no, but sending an http request on events to a sms gateway shouldn't be
> too
> difficult.
>
>

-- 
Jeremie Chism
Triton Communications
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130301/a3639634/attachment-0001.html>

From skchopperguy at gmail.com  Fri Mar  1 20:41:22 2013
From: skchopperguy at gmail.com (Skyler)
Date: Fri, 01 Mar 2013 17:41:22 -0800
Subject: [Spce-user] SMS
In-Reply-To: <CAG-pSShJ3mf8LdKKfwYjTj3=NyZ0gm_NkOBBBvmSin08MEYFrQ@mail.gmail.com>
References: <EBB6CE73-9366-45FD-8E9E-8585458FA2B9@gmail.com>
	<20130301224615.1255f743@quenya>
	<CAG-pSShJ3mf8LdKKfwYjTj3=NyZ0gm_NkOBBBvmSin08MEYFrQ@mail.gmail.com>
Message-ID: <513158C2.1000900@gmail.com>

In theory, if your SMSC can translate MESSAGE it should not be too hard 
to treat your SMSC in the same way peers are handled. ie: lookup local, 
if not found then go to smsc gateway. Mind you I'm sure it would be more 
involved than that.

what are you using as an smsc? I'd be into tinkering for fun to see if a 
stable solution could come of it.

Skyler

On 3/1/2013 3:11 PM, Jeremie Chism wrote:
> Yes it is sent by sip message.
>
> On Friday, March 1, 2013, Jon Bonilla wrote:
>
>     El Fri, 1 Mar 2013 15:37:41 -0600
>     Jeremie Chism <jchism2 at gmail.com <javascript:;>> escribi?:
>
>      > Does sipwise support SMS and if so is there something that needs
>     to be
>      > changed in the config to make it work.
>
>
>     Could you tell us what exactly you mean by supporting sms? I guess
>     the answer
>     is no, but sending an http request on events to a sms gateway
>     shouldn't be too
>     difficult.
>
>
>
> --
> Jeremie Chism
> Triton Communications
>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
>



From jchism2 at gmail.com  Fri Mar  1 20:50:57 2013
From: jchism2 at gmail.com (Jeremie Chism)
Date: Fri, 1 Mar 2013 19:50:57 -0600
Subject: [Spce-user] SMS
In-Reply-To: <513158C2.1000900@gmail.com>
References: <EBB6CE73-9366-45FD-8E9E-8585458FA2B9@gmail.com>
	<20130301224615.1255f743@quenya>
	<CAG-pSShJ3mf8LdKKfwYjTj3=NyZ0gm_NkOBBBvmSin08MEYFrQ@mail.gmail.com>
	<513158C2.1000900@gmail.com>
Message-ID: <9EF84F26-E19D-42E7-B77A-0FE3EB44BB21@gmail.com>

Here is a link to the wiki our carrier has

http://wiki.voipinnovations.com/SMS.ashx

Sent from my iPhone

On Mar 1, 2013, at 7:41 PM, Skyler <skchopperguy at gmail.com> wrote:

> In theory, if your SMSC can translate MESSAGE it should not be too hard to treat your SMSC in the same way peers are handled. ie: lookup local, if not found then go to smsc gateway. Mind you I'm sure it would be more involved than that.
> 
> what are you using as an smsc? I'd be into tinkering for fun to see if a stable solution could come of it.
> 
> Skyler
> 
> On 3/1/2013 3:11 PM, Jeremie Chism wrote:
>> Yes it is sent by sip message.
>> 
>> On Friday, March 1, 2013, Jon Bonilla wrote:
>> 
>>    El Fri, 1 Mar 2013 15:37:41 -0600
>>    Jeremie Chism <jchism2 at gmail.com <javascript:;>> escribi?:
>> 
>>     > Does sipwise support SMS and if so is there something that needs
>>    to be
>>     > changed in the config to make it work.
>> 
>> 
>>    Could you tell us what exactly you mean by supporting sms? I guess
>>    the answer
>>    is no, but sending an http request on events to a sms gateway
>>    shouldn't be too
>>    difficult.
>> 
>> 
>> 
>> --
>> Jeremie Chism
>> Triton Communications
>> 
>> 
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/listinfo/spce-user
> 
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130301/08aee657/attachment-0001.html>

From skchopperguy at gmail.com  Fri Mar  1 22:05:18 2013
From: skchopperguy at gmail.com (Skyler)
Date: Fri, 01 Mar 2013 19:05:18 -0800
Subject: [Spce-user] SMS
In-Reply-To: <9EF84F26-E19D-42E7-B77A-0FE3EB44BB21@gmail.com>
References: <EBB6CE73-9366-45FD-8E9E-8585458FA2B9@gmail.com>
	<20130301224615.1255f743@quenya>
	<CAG-pSShJ3mf8LdKKfwYjTj3=NyZ0gm_NkOBBBvmSin08MEYFrQ@mail.gmail.com>
	<513158C2.1000900@gmail.com>
	<9EF84F26-E19D-42E7-B77A-0FE3EB44BB21@gmail.com>
Message-ID: <51316C6E.2060201@gmail.com>

That's very similar to what I was about to start work on. IM between 
multiple SPCE boxes. What the link describes is basically an IM being 
passed along to another system if not found locally. Can definitely be 
done but will take more thought into the handling and testing though.

Skyler

On 3/1/2013 5:50 PM, Jeremie Chism wrote:
> Here is a link to the wiki our carrier has
>
> http://wiki.voipinnovations.com/SMS.ashx
>
> Sent from my iPhone
>
> On Mar 1, 2013, at 7:41 PM, Skyler <skchopperguy at gmail.com
> <mailto:skchopperguy at gmail.com>> wrote:
>
>> In theory, if your SMSC can translate MESSAGE it should not be too
>> hard to treat your SMSC in the same way peers are handled. ie: lookup
>> local, if not found then go to smsc gateway. Mind you I'm sure it
>> would be more involved than that.
>>
>> what are you using as an smsc? I'd be into tinkering for fun to see if
>> a stable solution could come of it.
>>
>> Skyler
>>
>> On 3/1/2013 3:11 PM, Jeremie Chism wrote:
>>> Yes it is sent by sip message.
>>>
>>> On Friday, March 1, 2013, Jon Bonilla wrote:
>>>
>>>    El Fri, 1 Mar 2013 15:37:41 -0600
>>>    Jeremie Chism <jchism2 at gmail.com <mailto:jchism2 at gmail.com>
>>> <javascript:;>> escribi?:
>>>
>>>     > Does sipwise support SMS and if so is there something that needs
>>>    to be
>>>     > changed in the config to make it work.
>>>
>>>
>>>    Could you tell us what exactly you mean by supporting sms? I guess
>>>    the answer
>>>    is no, but sending an http request on events to a sms gateway
>>>    shouldn't be too
>>>    difficult.
>>>
>>>
>>>
>>> --
>>> Jeremie Chism
>>> Triton Communications
>>>
>>>
>>> _______________________________________________
>>> Spce-user mailing list
>>> Spce-user at lists.sipwise.com <mailto:Spce-user at lists.sipwise.com>
>>> http://lists.sipwise.com/listinfo/spce-user
>>>
>>
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com <mailto:Spce-user at lists.sipwise.com>
>> http://lists.sipwise.com/listinfo/spce-user



From blagojaanusev at anutel.ca  Sun Mar  3 18:06:39 2013
From: blagojaanusev at anutel.ca (Blagoja)
Date: Sun, 03 Mar 2013 18:06:39 -0500
Subject: [Spce-user] non existing right pvar
In-Reply-To: <CEEE718799582B47A55D7897DCB264FC280153@bvusaex.bestvoipusa.local>
References: <512FD9E2.3060003@anutel.ca>
	<CEEE718799582B47A55D7897DCB264FC28014A@bvusaex.bestvoipusa.local>
	<7B51A6CD-1098-44DE-B67B-35688C683785@anutel.ca>
	<CEEE718799582B47A55D7897DCB264FC280153@bvusaex.bestvoipusa.local>
Message-ID: <5133D77F.6090402@anutel.ca>

I'm geting this error on incoming call in proxy.cfg



Mar  3 17:59:52 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[2254]: ERROR: 
<core> [lvalue.c:347]: non existing right pvar
Mar  3 17:59:52 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[2254]: ERROR: 
<core> [lvalue.c:407]: assignment failed at pos: (1920,9-1920,28)
Mar  3 17:59:52 x1-6-00-1f-c6-69-61-d9 /usr/sbin/kamailio[2254]: ERROR: 
avpops [avpops_impl.c:334]: incomplet uri <sip:x.x.x.x:5060>





From axessofficetheo at gmail.com  Mon Mar  4 01:59:03 2013
From: axessofficetheo at gmail.com (Theo)
Date: Mon, 4 Mar 2013 08:59:03 +0200
Subject: [Spce-user] Upper Registration to Asterisk - transfers not
	working
In-Reply-To: <CA+9+E=XRB-hPxK1ZTBU5ebhnaq-y0wS0=B=8ZaeVCheBC6+tWg@mail.gmail.com>
References: <CA+9+E=X_qiRReX=c_XubTD89HEV2HApxFw1yRDgDa2nW0o_SZg@mail.gmail.com>
	<20130215213405.58b3a93a@quenya>
	<CA+9+E=Wnubt_Wy+xHjk0C6kw0XQ8tc2-agbFvkevcTpYJEqg4Q@mail.gmail.com>
	<720785848-1361030221-cardhu_decombobulator_blackberry.rim.net-863913820-@b16.c11.bise7.blackberry>
	<CA+9+E=Xh_1QHALsUzOhnUOK7q4uD8ZyoLwHoEs2R-zHh=7Ox5g@mail.gmail.com>
	<CA+9+E=U+N0GJY7QOmvTYgW5fVsetEZ_e2rYTOtp_EWeH5U7HTg@mail.gmail.com>
	<CA+9+E=U215fg8=y_ff46U2pyUxB4imXi1-i1Sx3bMbtECZ+28A@mail.gmail.com>
	<51224FFD.4020408@sipwise.com> <51225420.5000607@sipwise.com>
	<CA+9+E=U-t1paf0KW+H-vHkOwGFBkbn1=8NFd5YUPdEwm_xyZsg@mail.gmail.com>
	<5122598B.1060703@sipwise.com>
	<CA+9+E=XeYZRHuBQkhsHBTMCarq2bOdX9c+93suqyCCb-Gj+3Ag@mail.gmail.com>
	<20130226152313.2d977e1d@quenya>
	<CA+9+E=WQUrn1HbeekF2fchdZ+7kX54Snphe0QY6W_x4oXN6yyA@mail.gmail.com>
	<CA+9+E=XRB-hPxK1ZTBU5ebhnaq-y0wS0=B=8ZaeVCheBC6+tWg@mail.gmail.com>
Message-ID: <CA+9+E=WU=Sa9NTZ=HJrzLXZ+VnTON_--+9fBaHtivs+Lfh7Vwg@mail.gmail.com>

Hi

Not sure if I am breaking protocol by bumping - if I did - apologies. I am
almost ready to put this in as our SBC upper registration solution (more to
follow after that I hope :-) - but need to resolve the attended transfer
problem. Any ideas on this? Sip Header filtering is off altogether (does
that have any caveats?). Transfer working fine now for all phones tested,
but attended transfer not.

Cheers

On Tue, Feb 26, 2013 at 4:43 PM, Theo <axessofficetheo at gmail.com> wrote:

> ah,
>
> It did not fix the issue of attended transfer, but it DID fix the other
> phone (fanvil) that didn't work on any type of transfer. So, yes, something
> to do with the headers.
>
> Now onto the attended transfer :-)
>
>
> On Tue, Feb 26, 2013 at 4:41 PM, Theo <axessofficetheo at gmail.com> wrote:
>
>> Hi Jon
>>
>> Assuming that commenting out like this:
>>
>> #header_filter=whitelist
>>
>> Is all i have to do to disable it, it doesn't work. Am I correct in
>> assuming that will disable header filtering?
>>
>>
>>
>> On Tue, Feb 26, 2013 at 4:23 PM, Jon Bonilla <jbonilla at sipwise.com>wrote:
>>
>>> El Tue, 26 Feb 2013 15:27:26 +0200
>>> Theo <axessofficetheo at gmail.com> escribi?:
>>>
>>> > Hi
>>> >
>>> > Real life took over for a little bit, back at this now. It's not
>>> turning
>>> > out to be as easy as I had hoped :-(
>>> >
>>> > I have 2 issues currently with this.
>>> >
>>> > 1. SNOM 300 - blind transfer works fine but attended transfer doesn't.
>>> It
>>> > does send the refer-to and using ngrep I don't see a difference
>>> between the
>>> > two, but then again I am usually wrong. The attended transfer just
>>> seems to
>>> > put me on hold and not get out of that.
>>> >
>>> > 2. I have another phone for testing - a Fanvil. Transferring connected
>>> > directly to asterisk works fine, but through SCPE it doesn't - error
>>> > message "refer-to missing"
>>> >
>>> > How do I troubleshoot this further?
>>> >
>>> >
>>>
>>> Try to check if the headers of are the same on both legs. Maybe sems is
>>> eating
>>> headers. Another option would be to disable header filtering
>>> in /etc/sems/etc/ngcp.sbcprofile.conf ans retry.
>>>
>>>
>>> _______________________________________________
>>> Spce-user mailing list
>>> Spce-user at lists.sipwise.com
>>> http://lists.sipwise.com/listinfo/spce-user
>>>
>>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130304/ffd08cdf/attachment-0001.html>

From joan at megatro.com  Mon Mar  4 04:24:29 2013
From: joan at megatro.com (Joan Jou)
Date: Mon, 04 Mar 2013 10:24:29 +0100
Subject: [Spce-user] Statistics
Message-ID: <5134684D.203@megatro.com>

Hi,
On a new intallation.
Everythink looks running but Information/statistics  stay "loading...".

Thank you,


-- 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130304/58f04b58/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signatura1.gif
Type: image/gif
Size: 13564 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130304/58f04b58/attachment-0001.gif>

From chessmancaryl at gmail.com  Mon Mar  4 06:48:51 2013
From: chessmancaryl at gmail.com (chess man)
Date: Mon, 4 Mar 2013 12:48:51 +0100
Subject: [Spce-user] class diagramm
Message-ID: <CAB9Pg_qH=nYDqmXF6UYdFcvW8Y8h-zfDsNZDdvA3Fp_MhavELQ@mail.gmail.com>

Good morning

Could you please provide me with the class UML diagramm of the sip wise
solution
and the schema of the database management.

Best Regards.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130304/d4915ae9/attachment-0001.html>

From kmasse at questblue.com  Mon Mar  4 07:15:34 2013
From: kmasse at questblue.com (Kevin Masse)
Date: Mon, 4 Mar 2013 07:15:34 -0500
Subject: [Spce-user] Statistics
References: <5134684D.203@megatro.com>
Message-ID: <CEEE718799582B47A55D7897DCB264FC280176@bvusaex.bestvoipusa.local>

Good morning Joan, if you are using Internet Explorer try using Chrome
or Firefox.

 

I have noticed that IE does have this issue with the stats page.

 

Kevin

 

 

 

From: spce-user-bounces at lists.sipwise.com
[mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of Joan Jou
Sent: Monday, March 04, 2013 4:24 AM
To: spce-user at lists.sipwise.com
Subject: [Spce-user] Statistics

 

Hi,
On a new intallation.
Everythink looks running but Information/statistics  stay  "loading...".

Thank you,



-- 
 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130304/9e1dfeb5/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.gif
Type: image/gif
Size: 13564 bytes
Desc: image001.gif
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130304/9e1dfeb5/attachment-0001.gif>

From kmasse at questblue.com  Mon Mar  4 07:41:22 2013
From: kmasse at questblue.com (Kevin Masse)
Date: Mon, 4 Mar 2013 07:41:22 -0500
Subject: [Spce-user] Statistics
References: <5134684D.203@megatro.com>
	<CEEE718799582B47A55D7897DCB264FC280176@bvusaex.bestvoipusa.local>
	<51349462.1080701@megatro.com>
Message-ID: <CEEE718799582B47A55D7897DCB264FC280177@bvusaex.bestvoipusa.local>

Ok, I will step away from this one.  I do not have additional
suggestions.

 

 

Kevin

 

 

 

From: Joan Jou [mailto:joan at megatro.com] 
Sent: Monday, March 04, 2013 7:33 AM
To: Kevin Masse
Subject: Re: [Spce-user] Statistics

 

Good morning,
I try it whith  Explorer, Chrome and Firefox.

Joan,

Al 04/03/2013 13:15, En/na Kevin Masse ha escrit:

	Good morning Joan, if you are using Internet Explorer try using
Chrome or Firefox.

	 

	I have noticed that IE does have this issue with the stats page.

	 

	Kevin

	 

	 

	 

	From: spce-user-bounces at lists.sipwise.com [
mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of Joan Jou
	Sent: Monday, March 04, 2013 4:24 AM
	To: spce-user at lists.sipwise.com
	Subject: [Spce-user] Statistics

	 

	Hi,
	On a new intallation.
	Everythink looks running but Information/statistics  stay
"loading...".
	
	Thank you,
	
	
	

	-- 
	 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130304/b6070cea/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.gif
Type: image/gif
Size: 13564 bytes
Desc: image001.gif
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130304/b6070cea/attachment-0001.gif>

From agranig at sipwise.com  Mon Mar  4 07:55:01 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Mon, 04 Mar 2013 13:55:01 +0100
Subject: [Spce-user] class diagramm
In-Reply-To: <CAB9Pg_qH=nYDqmXF6UYdFcvW8Y8h-zfDsNZDdvA3Fp_MhavELQ@mail.gmail.com>
References: <CAB9Pg_qH=nYDqmXF6UYdFcvW8Y8h-zfDsNZDdvA3Fp_MhavELQ@mail.gmail.com>
Message-ID: <513499A5.5000208@sipwise.com>

Hi,

On 03/04/2013 12:48 PM, chess man wrote:
> Could you please provide me with the class UML diagramm of the sip wise
> solution

There is http://sipwise.com/doc/2.7/spce/ar01s02.html, but there is no 
such thing as a class UML diagram, as the solution is not a single 
application, but consists of quite a lot of different sub-systems.

> and the schema of the database management.

You can create one using http://schemaspy.sourceforge.net/, which 
produces quite usable db schema descriptions.

Andreas



From agranig at sipwise.com  Mon Mar  4 07:56:47 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Mon, 04 Mar 2013 13:56:47 +0100
Subject: [Spce-user] Statistics
In-Reply-To: <CEEE718799582B47A55D7897DCB264FC280177@bvusaex.bestvoipusa.local>
References: <5134684D.203@megatro.com>
	<CEEE718799582B47A55D7897DCB264FC280176@bvusaex.bestvoipusa.local>
	<51349462.1080701@megatro.com>
	<CEEE718799582B47A55D7897DCB264FC280177@bvusaex.bestvoipusa.local>
Message-ID: <51349A0F.7050403@sipwise.com>

Hi,

Check /var/log/ngcp/oss.log to see if there are any errors. Maybe it 
doesn't find the rrd files in /var/lib/collectd/rrd, maybe because 
collectd is not running?

Andreas

On 03/04/2013 01:41 PM, Kevin Masse wrote:
> Ok, I will step away from this one.  I do not have additional suggestions.
>
> Kevin
>
> *From:*Joan Jou [mailto:joan at megatro.com]
> *Sent:* Monday, March 04, 2013 7:33 AM
> *To:* Kevin Masse
> *Subject:* Re: [Spce-user] Statistics
>
> Good morning,
> I try it whith  Explorer, Chrome and Firefox.
>
> Joan,
>
> Al 04/03/2013 13:15, En/na Kevin Masse ha escrit:
>
>     Good morning Joan, if you are using Internet Explorer try using
>     Chrome or Firefox.
>
>     I have noticed that IE does have this issue with the stats page.
>
>     Kevin
>
>     *From:*spce-user-bounces at lists.sipwise.com
>     <mailto:spce-user-bounces at lists.sipwise.com>
>     [mailto:spce-user-bounces at lists.sipwise.com] *On Behalf Of *Joan Jou
>     *Sent:* Monday, March 04, 2013 4:24 AM
>     *To:* spce-user at lists.sipwise.com <mailto:spce-user at lists.sipwise.com>
>     *Subject:* [Spce-user] Statistics
>
>     Hi,
>     On a new intallation.
>     Everythink looks running but Information/statistics  stay  "loading...".
>
>     Thank you,
>
>
>     --
>
>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
>



From jbonilla at sipwise.com  Mon Mar  4 14:55:27 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Mon, 4 Mar 2013 20:55:27 +0100
Subject: [Spce-user] non existing right pvar
In-Reply-To: <5133D77F.6090402@anutel.ca>
References: <512FD9E2.3060003@anutel.ca>
	<CEEE718799582B47A55D7897DCB264FC28014A@bvusaex.bestvoipusa.local>
	<7B51A6CD-1098-44DE-B67B-35688C683785@anutel.ca>
	<CEEE718799582B47A55D7897DCB264FC280153@bvusaex.bestvoipusa.local>
	<5133D77F.6090402@anutel.ca>
Message-ID: <20130304205527.5c8a411f@multivac>

El Sun, 03 Mar 2013 18:06:39 -0500
Blagoja <blagojaanusev at anutel.ca> escribi?:

> En-Respuesta-A:
> <CEEE718799582B47A55D7897DCB264FC280153 at bvusaex.bestvoipusa.local

> Asunto:
> [Spce-user] non existing right pvar



Please don't answer a mail to create a new thread as you break threading in
mail clients. use the "new mail" button.

Thanks


Jon




From csaroli at isysp.com  Mon Mar  4 16:25:12 2013
From: csaroli at isysp.com (Christopher Saroli)
Date: Mon, 4 Mar 2013 21:25:12 +0000
Subject: [Spce-user]  Billing Software
Message-ID: <EE7149FDD60B3045A4377F05AAFFF3F118333057@ISS-Server.ISS.local>

Anyone know of a decent 3rd party billing system that works well with SPCE.  Preferably open-source.

Any suggestions would be appreciated, thinking of writing my own if I cannot find anything.

Thanks

---------------------------------
Christopher Saroli
csaroli at isysp.com


From gavin.sweet at skyracktelecom.com  Mon Mar  4 16:35:27 2013
From: gavin.sweet at skyracktelecom.com (Gavin Sweet)
Date: Mon, 4 Mar 2013 21:35:27 -0000
Subject: [Spce-user] Billing Software
In-Reply-To: <EE7149FDD60B3045A4377F05AAFFF3F118333057@ISS-Server.ISS.local>
References: <EE7149FDD60B3045A4377F05AAFFF3F118333057@ISS-Server.ISS.local>
Message-ID: <015901ce1920$3072f090$9158d1b0$@sweet@skyracktelecom.com>

Hi Christopher - Try jbilling. I'm partisan as we happen to act as
professional services partners for jbilling, but we do also run our own
hosted voice operations from a combination of open source software that
includes both sipwise and jbilling. It is actually designed to deliver
telecoms billing with mediation and rating.

Cheers
Gavin



> -----Original Message-----
> From: spce-user-bounces at lists.sipwise.com [mailto:spce-user-
> bounces at lists.sipwise.com] On Behalf Of Christopher Saroli
> Sent: 04 March 2013 21:25
> To: spce-user at lists.sipwise.com
> Subject: [Spce-user] Billing Software
> 
> Anyone know of a decent 3rd party billing system that works well with
> SPCE.  Preferably open-source.
> 
> Any suggestions would be appreciated, thinking of writing my own if I
> cannot find anything.
> 
> Thanks
> 
> ---------------------------------
> Christopher Saroli
> csaroli at isysp.com
> 
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
> -----
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2013.0.2899 / Virus Database: 2641/6145 - Release Date:
> 03/03/13




From orenyny at gmail.com  Mon Mar  4 19:43:02 2013
From: orenyny at gmail.com (Oren Yehezkely)
Date: Mon, 4 Mar 2013 19:43:02 -0500
Subject: [Spce-user] Billing Software
In-Reply-To: <513513b2.41ae0e0a.5d09.0c12SMTPIN_ADDED_BROKEN@mx.google.com>
References: <EE7149FDD60B3045A4377F05AAFFF3F118333057@ISS-Server.ISS.local>
	<513513b2.41ae0e0a.5d09.0c12SMTPIN_ADDED_BROKEN@mx.google.com>
Message-ID: <CAL+WrhJpsp0pvUAYL-rgxt09zQ-SALqN9joEbY4DnQ18bObWkw@mail.gmail.com>

Christopher,

If you are writing your own, I may be interested to participate.

The market for open source does not look very good.
jBilling seems to be open, only by name. I think they found it to be a
good marketing tool.
You can download the code but what to do after that?

Here are a few projects I have found in my searches . I will be glad
to learn about others you have found.

1. http://www.agileco.com/ - Agilebill - This project seems to be
dead. There is a maintainer, but the founder seems to have abandoned
the project with no one else in his place.
2. www.opencdrrate.org - OpenCDR - They seem to have made some
significant progress in the last year.
It looks like they are active in adding more features and fixing bugs.
Even though small they look vibrant.
3. http://freeside.biz/freeside/ - Freeside - this is an active
company as far as I can tell. They also have some kind of CRM. Take a
look at the demo they have online.
They integrate with multiple platforms.

Regards,

Oren


On Mon, Mar 4, 2013 at 4:35 PM, Gavin Sweet
<gavin.sweet at skyracktelecom.com> wrote:
> Hi Christopher - Try jbilling. I'm partisan as we happen to act as
> professional services partners for jbilling, but we do also run our own
> hosted voice operations from a combination of open source software that
> includes both sipwise and jbilling. It is actually designed to deliver
> telecoms billing with mediation and rating.
>
> Cheers
> Gavin
>
>
>
>> -----Original Message-----
>> From: spce-user-bounces at lists.sipwise.com [mailto:spce-user-
>> bounces at lists.sipwise.com] On Behalf Of Christopher Saroli
>> Sent: 04 March 2013 21:25
>> To: spce-user at lists.sipwise.com
>> Subject: [Spce-user] Billing Software
>>
>> Anyone know of a decent 3rd party billing system that works well with
>> SPCE.  Preferably open-source.
>>
>> Any suggestions would be appreciated, thinking of writing my own if I
>> cannot find anything.
>>
>> Thanks
>>
>> ---------------------------------
>> Christopher Saroli
>> csaroli at isysp.com
>>
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/listinfo/spce-user
>> -----
>> No virus found in this message.
>> Checked by AVG - www.avg.com
>> Version: 2013.0.2899 / Virus Database: 2641/6145 - Release Date:
>> 03/03/13
>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user



From skchopperguy at gmail.com  Mon Mar  4 20:50:11 2013
From: skchopperguy at gmail.com (Skyler)
Date: Mon, 04 Mar 2013 17:50:11 -0800
Subject: [Spce-user] Billing Software
In-Reply-To: <CAL+WrhJpsp0pvUAYL-rgxt09zQ-SALqN9joEbY4DnQ18bObWkw@mail.gmail.com>
References: <EE7149FDD60B3045A4377F05AAFFF3F118333057@ISS-Server.ISS.local>
	<513513b2.41ae0e0a.5d09.0c12SMTPIN_ADDED_BROKEN@mx.google.com>
	<CAL+WrhJpsp0pvUAYL-rgxt09zQ-SALqN9joEbY4DnQ18bObWkw@mail.gmail.com>
Message-ID: <51354F53.9080707@gmail.com>

Blesta looks pretty good. Has an open back-end to create modules. ie: 
SOAP. They have a v3.0 coming out, not sure when though. I'd be into 
collaboration.

Skyler

On 3/4/2013 4:43 PM, Oren Yehezkely wrote:
> Christopher,
>
> If you are writing your own, I may be interested to participate.
>
> The market for open source does not look very good.
> jBilling seems to be open, only by name. I think they found it to be a
> good marketing tool.
> You can download the code but what to do after that?
>
> Here are a few projects I have found in my searches . I will be glad
> to learn about others you have found.
>
> 1. http://www.agileco.com/ - Agilebill - This project seems to be
> dead. There is a maintainer, but the founder seems to have abandoned
> the project with no one else in his place.
> 2. www.opencdrrate.org - OpenCDR - They seem to have made some
> significant progress in the last year.
> It looks like they are active in adding more features and fixing bugs.
> Even though small they look vibrant.
> 3. http://freeside.biz/freeside/ - Freeside - this is an active
> company as far as I can tell. They also have some kind of CRM. Take a
> look at the demo they have online.
> They integrate with multiple platforms.
>
> Regards,
>
> Oren
>
>
> On Mon, Mar 4, 2013 at 4:35 PM, Gavin Sweet
> <gavin.sweet at skyracktelecom.com> wrote:
>> Hi Christopher - Try jbilling. I'm partisan as we happen to act as
>> professional services partners for jbilling, but we do also run our own
>> hosted voice operations from a combination of open source software that
>> includes both sipwise and jbilling. It is actually designed to deliver
>> telecoms billing with mediation and rating.
>>
>> Cheers
>> Gavin
>>
>>
>>
>>> -----Original Message-----
>>> From: spce-user-bounces at lists.sipwise.com [mailto:spce-user-
>>> bounces at lists.sipwise.com] On Behalf Of Christopher Saroli
>>> Sent: 04 March 2013 21:25
>>> To: spce-user at lists.sipwise.com
>>> Subject: [Spce-user] Billing Software
>>>
>>> Anyone know of a decent 3rd party billing system that works well with
>>> SPCE.  Preferably open-source.
>>>
>>> Any suggestions would be appreciated, thinking of writing my own if I
>>> cannot find anything.
>>>
>>> Thanks
>>>
>>> ---------------------------------
>>> Christopher Saroli
>>> csaroli at isysp.com
>>>
>>> _______________________________________________
>>> Spce-user mailing list
>>> Spce-user at lists.sipwise.com
>>> http://lists.sipwise.com/listinfo/spce-user
>>> -----
>>> No virus found in this message.
>>> Checked by AVG - www.avg.com
>>> Version: 2013.0.2899 / Virus Database: 2641/6145 - Release Date:
>>> 03/03/13
>>
>>
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/listinfo/spce-user
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
>



From orenyny at gmail.com  Mon Mar  4 21:36:39 2013
From: orenyny at gmail.com (Oren Yehezkely)
Date: Mon, 4 Mar 2013 21:36:39 -0500
Subject: [Spce-user] Billing Software
In-Reply-To: <51354F53.9080707@gmail.com>
References: <EE7149FDD60B3045A4377F05AAFFF3F118333057@ISS-Server.ISS.local>
	<513513b2.41ae0e0a.5d09.0c12SMTPIN_ADDED_BROKEN@mx.google.com>
	<CAL+WrhJpsp0pvUAYL-rgxt09zQ-SALqN9joEbY4DnQ18bObWkw@mail.gmail.com>
	<51354F53.9080707@gmail.com>
Message-ID: <CAL+WrhL3_g3D4CNZ5DMix4XLYqTfOhbO6g5DEN0EiQ1nO622wA@mail.gmail.com>

Yes, it does look good.
But I think version 3.0 is a long time in the making.

Not sure if it was designed for VoIP.

Oren


On Mon, Mar 4, 2013 at 8:50 PM, Skyler <skchopperguy at gmail.com> wrote:
> Blesta looks pretty good. Has an open back-end to create modules. ie: SOAP.
> They have a v3.0 coming out, not sure when though. I'd be into
> collaboration.
>
> Skyler
>
>
> On 3/4/2013 4:43 PM, Oren Yehezkely wrote:
>>
>> Christopher,
>>
>> If you are writing your own, I may be interested to participate.
>>
>> The market for open source does not look very good.
>> jBilling seems to be open, only by name. I think they found it to be a
>> good marketing tool.
>> You can download the code but what to do after that?
>>
>> Here are a few projects I have found in my searches . I will be glad
>> to learn about others you have found.
>>
>> 1. http://www.agileco.com/ - Agilebill - This project seems to be
>> dead. There is a maintainer, but the founder seems to have abandoned
>> the project with no one else in his place.
>> 2. www.opencdrrate.org - OpenCDR - They seem to have made some
>> significant progress in the last year.
>> It looks like they are active in adding more features and fixing bugs.
>> Even though small they look vibrant.
>> 3. http://freeside.biz/freeside/ - Freeside - this is an active
>> company as far as I can tell. They also have some kind of CRM. Take a
>> look at the demo they have online.
>> They integrate with multiple platforms.
>>
>> Regards,
>>
>> Oren
>>
>>
>> On Mon, Mar 4, 2013 at 4:35 PM, Gavin Sweet
>> <gavin.sweet at skyracktelecom.com> wrote:
>>>
>>> Hi Christopher - Try jbilling. I'm partisan as we happen to act as
>>> professional services partners for jbilling, but we do also run our own
>>> hosted voice operations from a combination of open source software that
>>> includes both sipwise and jbilling. It is actually designed to deliver
>>> telecoms billing with mediation and rating.
>>>
>>> Cheers
>>> Gavin
>>>
>>>
>>>
>>>> -----Original Message-----
>>>> From: spce-user-bounces at lists.sipwise.com [mailto:spce-user-
>>>> bounces at lists.sipwise.com] On Behalf Of Christopher Saroli
>>>> Sent: 04 March 2013 21:25
>>>> To: spce-user at lists.sipwise.com
>>>> Subject: [Spce-user] Billing Software
>>>>
>>>> Anyone know of a decent 3rd party billing system that works well with
>>>> SPCE.  Preferably open-source.
>>>>
>>>> Any suggestions would be appreciated, thinking of writing my own if I
>>>> cannot find anything.
>>>>
>>>> Thanks
>>>>
>>>> ---------------------------------
>>>> Christopher Saroli
>>>> csaroli at isysp.com
>>>>
>>>> _______________________________________________
>>>> Spce-user mailing list
>>>> Spce-user at lists.sipwise.com
>>>> http://lists.sipwise.com/listinfo/spce-user
>>>> -----
>>>> No virus found in this message.
>>>> Checked by AVG - www.avg.com
>>>> Version: 2013.0.2899 / Virus Database: 2641/6145 - Release Date:
>>>> 03/03/13
>>>
>>>
>>>
>>> _______________________________________________
>>> Spce-user mailing list
>>> Spce-user at lists.sipwise.com
>>> http://lists.sipwise.com/listinfo/spce-user
>>
>>
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/listinfo/spce-user
>>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user



From skchopperguy at gmail.com  Mon Mar  4 22:23:09 2013
From: skchopperguy at gmail.com (Skyler)
Date: Mon, 04 Mar 2013 19:23:09 -0800
Subject: [Spce-user] Billing Software
In-Reply-To: <CAL+WrhL3_g3D4CNZ5DMix4XLYqTfOhbO6g5DEN0EiQ1nO622wA@mail.gmail.com>
References: <EE7149FDD60B3045A4377F05AAFFF3F118333057@ISS-Server.ISS.local>
	<513513b2.41ae0e0a.5d09.0c12SMTPIN_ADDED_BROKEN@mx.google.com>
	<CAL+WrhJpsp0pvUAYL-rgxt09zQ-SALqN9joEbY4DnQ18bObWkw@mail.gmail.com>
	<51354F53.9080707@gmail.com>
	<CAL+WrhL3_g3D4CNZ5DMix4XLYqTfOhbO6g5DEN0EiQ1nO622wA@mail.gmail.com>
Message-ID: <5135651D.5060501@gmail.com>

I think it was more designed for hosting. The trial looks like it can 
make service packages and those packages tie-into modules. I looked at 
the modules in there and should be straight forward to create an SPCE 
module. Looks like there is cron also which could be an idea for 
querying SPCE by external_id and building invoice(s). Maybe.

Skyler

On 3/4/2013 6:36 PM, Oren Yehezkely wrote:
> Yes, it does look good.
> But I think version 3.0 is a long time in the making.
>
> Not sure if it was designed for VoIP.
>
> Oren
>
>
> On Mon, Mar 4, 2013 at 8:50 PM, Skyler <skchopperguy at gmail.com> wrote:
>> Blesta looks pretty good. Has an open back-end to create modules. ie: SOAP.
>> They have a v3.0 coming out, not sure when though. I'd be into
>> collaboration.
>>
>> Skyler
>>
>>
>> On 3/4/2013 4:43 PM, Oren Yehezkely wrote:
>>>
>>> Christopher,
>>>
>>> If you are writing your own, I may be interested to participate.
>>>
>>> The market for open source does not look very good.
>>> jBilling seems to be open, only by name. I think they found it to be a
>>> good marketing tool.
>>> You can download the code but what to do after that?
>>>
>>> Here are a few projects I have found in my searches . I will be glad
>>> to learn about others you have found.
>>>
>>> 1. http://www.agileco.com/ - Agilebill - This project seems to be
>>> dead. There is a maintainer, but the founder seems to have abandoned
>>> the project with no one else in his place.
>>> 2. www.opencdrrate.org - OpenCDR - They seem to have made some
>>> significant progress in the last year.
>>> It looks like they are active in adding more features and fixing bugs.
>>> Even though small they look vibrant.
>>> 3. http://freeside.biz/freeside/ - Freeside - this is an active
>>> company as far as I can tell. They also have some kind of CRM. Take a
>>> look at the demo they have online.
>>> They integrate with multiple platforms.
>>>
>>> Regards,
>>>
>>> Oren
>>>
>>>
>>> On Mon, Mar 4, 2013 at 4:35 PM, Gavin Sweet
>>> <gavin.sweet at skyracktelecom.com> wrote:
>>>>
>>>> Hi Christopher - Try jbilling. I'm partisan as we happen to act as
>>>> professional services partners for jbilling, but we do also run our own
>>>> hosted voice operations from a combination of open source software that
>>>> includes both sipwise and jbilling. It is actually designed to deliver
>>>> telecoms billing with mediation and rating.
>>>>
>>>> Cheers
>>>> Gavin
>>>>
>>>>
>>>>
>>>>> -----Original Message-----
>>>>> From: spce-user-bounces at lists.sipwise.com [mailto:spce-user-
>>>>> bounces at lists.sipwise.com] On Behalf Of Christopher Saroli
>>>>> Sent: 04 March 2013 21:25
>>>>> To: spce-user at lists.sipwise.com
>>>>> Subject: [Spce-user] Billing Software
>>>>>
>>>>> Anyone know of a decent 3rd party billing system that works well with
>>>>> SPCE.  Preferably open-source.
>>>>>
>>>>> Any suggestions would be appreciated, thinking of writing my own if I
>>>>> cannot find anything.
>>>>>
>>>>> Thanks
>>>>>
>>>>> ---------------------------------
>>>>> Christopher Saroli
>>>>> csaroli at isysp.com
>>>>>
>>>>> _______________________________________________
>>>>> Spce-user mailing list
>>>>> Spce-user at lists.sipwise.com
>>>>> http://lists.sipwise.com/listinfo/spce-user
>>>>> -----
>>>>> No virus found in this message.
>>>>> Checked by AVG - www.avg.com
>>>>> Version: 2013.0.2899 / Virus Database: 2641/6145 - Release Date:
>>>>> 03/03/13
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Spce-user mailing list
>>>> Spce-user at lists.sipwise.com
>>>> http://lists.sipwise.com/listinfo/spce-user
>>>
>>>
>>> _______________________________________________
>>> Spce-user mailing list
>>> Spce-user at lists.sipwise.com
>>> http://lists.sipwise.com/listinfo/spce-user
>>>
>>
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/listinfo/spce-user



From jbonilla at sipwise.com  Tue Mar  5 01:59:53 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Tue, 5 Mar 2013 07:59:53 +0100
Subject: [Spce-user] Upper Registration to Asterisk - transfers not
 working
In-Reply-To: <CA+9+E=WU=Sa9NTZ=HJrzLXZ+VnTON_--+9fBaHtivs+Lfh7Vwg@mail.gmail.com>
References: <CA+9+E=X_qiRReX=c_XubTD89HEV2HApxFw1yRDgDa2nW0o_SZg@mail.gmail.com>
	<20130215213405.58b3a93a@quenya>
	<CA+9+E=Wnubt_Wy+xHjk0C6kw0XQ8tc2-agbFvkevcTpYJEqg4Q@mail.gmail.com>
	<720785848-1361030221-cardhu_decombobulator_blackberry.rim.net-863913820-@b16.c11.bise7.blackberry>
	<CA+9+E=Xh_1QHALsUzOhnUOK7q4uD8ZyoLwHoEs2R-zHh=7Ox5g@mail.gmail.com>
	<CA+9+E=U+N0GJY7QOmvTYgW5fVsetEZ_e2rYTOtp_EWeH5U7HTg@mail.gmail.com>
	<CA+9+E=U215fg8=y_ff46U2pyUxB4imXi1-i1Sx3bMbtECZ+28A@mail.gmail.com>
	<51224FFD.4020408@sipwise.com> <51225420.5000607@sipwise.com>
	<CA+9+E=U-t1paf0KW+H-vHkOwGFBkbn1=8NFd5YUPdEwm_xyZsg@mail.gmail.com>
	<5122598B.1060703@sipwise.com>
	<CA+9+E=XeYZRHuBQkhsHBTMCarq2bOdX9c+93suqyCCb-Gj+3Ag@mail.gmail.com>
	<20130226152313.2d977e1d@quenya>
	<CA+9+E=WQUrn1HbeekF2fchdZ+7kX54Snphe0QY6W_x4oXN6yyA@mail.gmail.com>
	<CA+9+E=XRB-hPxK1ZTBU5ebhnaq-y0wS0=B=8ZaeVCheBC6+tWg@mail.gmail.com>
	<CA+9+E=WU=Sa9NTZ=HJrzLXZ+VnTON_--+9fBaHtivs+Lfh7Vwg@mail.gmail.com>
Message-ID: <20130305075953.50b44f18@multivac>

El Mon, 4 Mar 2013 08:59:03 +0200
Theo <axessofficetheo at gmail.com> escribi?:

> Hi
> 
> Not sure if I am breaking protocol by bumping - if I did - apologies. I am
> almost ready to put this in as our SBC upper registration solution (more to
> follow after that I hope :-) - but need to resolve the attended transfer
> problem. Any ideas on this? Sip Header filtering is off altogether (does
> that have any caveats?). Transfer working fine now for all phones tested,
> but attended transfer not.
>

ngrep-sip?




From joan at megatro.com  Tue Mar  5 07:51:11 2013
From: joan at megatro.com (Joan Jou)
Date: Tue, 05 Mar 2013 13:51:11 +0100
Subject: [Spce-user] Statistics
In-Reply-To: <51349A0F.7050403@sipwise.com>
References: <5134684D.203@megatro.com>
	<CEEE718799582B47A55D7897DCB264FC280176@bvusaex.bestvoipusa.local>
	<51349462.1080701@megatro.com>
	<CEEE718799582B47A55D7897DCB264FC280177@bvusaex.bestvoipusa.local>
	<51349A0F.7050403@sipwise.com>
Message-ID: <5135EA3F.6040602@megatro.com>

Hi,
There are not files in rrd directori, only localhost and sip directories.

This is log file content:
ar  4 14:36:40 sip oss: 46.253.41.146 - - [04/Mar/2013:14:36:40 +0000] 
"GET / HTTP/1.1" 200 2803 "https://185.14.187.173:1443/dashboard/system" 
"Mozilla/5.0 (Windows NT 6.1; WOW64; rv:18.0) Gecko/20100101 
Firefox/18.0 FirePHP/0.7.1"
Mar  4 14:36:55 sip apache2: [info] [client 46.253.41.146] (70007)The 
timeout specified has expired: SSL input filter read failed.
Mar  4 14:36:55 sip apache2: [info] [client 46.253.41.146] Connection 
closed to child 7 with standard shutdown (server myserver:443)
Mar  4 14:37:03 sip apache2: [info] [client 46.253.41.146] Connection to 
child 5 established (server myserver:443)
Mar  4 14:37:03 sip apache2: [info] Seeding PRNG with 656 bytes of entropy
Mar  4 14:37:03 sip apache2: [info] Initial (No.1) HTTPS request 
received for child 5 (server myserver:443)
Mar  4 14:37:04 sip oss: 46.253.41.146 - - [04/Mar/2013:14:37:03 +0000] 
"GET /dashboard HTTP/1.1" 302 963 "https://185.14.187.173:1443/" 
"Mozilla/5.0 (Windows NT 6.1; WOW64; rv:18.0) Gecko/20100101 
Firefox/18.0 FirePHP/0.7.1"
Mar  4 14:37:04 sip apache2: [info] Subsequent (No.2) HTTPS request 
received for child 5 (server myserver:443)
Mar  4 14:37:04 sip oss: 46.253.41.146 - - [04/Mar/2013:14:37:04 +0000] 
"GET /dashboard/system HTTP/1.1" 200 3634 "https://185.14.187.173:1443/" 
"Mozilla/5.0 (Windows NT 6.1; WOW64; rv:18.0) Gecko/20100101 
Firefox/18.0 FirePHP/0.7.1"
Mar  4 14:37:19 sip apache2: [info] [client 46.253.41.146] (70007)The 
timeout specified has expired: SSL input filter read failed.
Mar  4 14:37:19 sip apache2: [info] [client 46.253.41.146] Connection 
closed to child 5 with standard shutdown (server myserver:443)

Joan,

Al 04/03/2013 13:56, En/na Andreas Granig ha escrit:
> Hi,
>
> Check /var/log/ngcp/oss.log to see if there are any errors. Maybe it 
> doesn't find the rrd files in /var/lib/collectd/rrd, maybe because 
> collectd is not running?
>
> Andreas
>
> On 03/04/2013 01:41 PM, Kevin Masse wrote:
>> Ok, I will step away from this one.  I do not have additional 
>> suggestions.
>>
>> Kevin
>>
>> *From:*Joan Jou [mailto:joan at megatro.com]
>> *Sent:* Monday, March 04, 2013 7:33 AM
>> *To:* Kevin Masse
>> *Subject:* Re: [Spce-user] Statistics
>>
>> Good morning,
>> I try it whith  Explorer, Chrome and Firefox.
>>
>> Joan,
>>
>> Al 04/03/2013 13:15, En/na Kevin Masse ha escrit:
>>
>>     Good morning Joan, if you are using Internet Explorer try using
>>     Chrome or Firefox.
>>
>>     I have noticed that IE does have this issue with the stats page.
>>
>>     Kevin
>>
>>     *From:*spce-user-bounces at lists.sipwise.com
>>     <mailto:spce-user-bounces at lists.sipwise.com>
>>     [mailto:spce-user-bounces at lists.sipwise.com] *On Behalf Of *Joan Jou
>>     *Sent:* Monday, March 04, 2013 4:24 AM
>>     *To:* spce-user at lists.sipwise.com 
>> <mailto:spce-user at lists.sipwise.com>
>>     *Subject:* [Spce-user] Statistics
>>
>>     Hi,
>>     On a new intallation.
>>     Everythink looks running but Information/statistics  stay 
>> "loading...".
>>
>>     Thank you,
>>
>>
>>     --
>>
>>
>>
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/listinfo/spce-user
>>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130305/f63dc772/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signatura1.gif
Type: image/gif
Size: 13564 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130305/f63dc772/attachment-0001.gif>

From matthew at tenacit.net  Tue Mar  5 08:34:18 2013
From: matthew at tenacit.net (Matthew Ogden)
Date: Tue, 5 Mar 2013 15:34:18 +0200
Subject: [Spce-user] rewrite rules
Message-ID: <8e982f2ac89360982fd9e03982d5c8d2@mail.gmail.com>

Hi



Its easy enough with a rewrite rule to replace something, but does it
support ?adding? one number to another



For example a replace for 555-555-55XX can make 3201 become 555-555-5501.
But what if you wanted 3201 to become 555-555-5506 and 3202 to become
555-555-5507?



Regards





*Matthew Ogden*

Management

TenacIT







*Strategic IT Consulting *?* Advanced Networking *? *Virtualisation* **

*Custom Development *?* **Hosting *?* **Syspro Support  *? *MS Licensing* **

National Tel: 041 10 10 100 | Cell: 084 205 4445 | Email:
matthew at tenacit.net

CT Tel: 021 201 0333 | Skype Name: matthew.ogden | Web:
http://www.tenacit.net
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130305/b171367d/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.jpg
Type: image/jpeg
Size: 6077 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130305/b171367d/attachment-0004.jpg>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image002.jpg
Type: image/jpeg
Size: 1240 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130305/b171367d/attachment-0005.jpg>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image003.png
Type: image/png
Size: 4650 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130305/b171367d/attachment-0001.png>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image004.jpg
Type: image/jpeg
Size: 1232 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130305/b171367d/attachment-0006.jpg>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image005.jpg
Type: image/jpeg
Size: 2254 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130305/b171367d/attachment-0007.jpg>

From sn at fest.dk  Tue Mar  5 08:47:19 2013
From: sn at fest.dk (Sylvester Nielsen)
Date: Tue, 5 Mar 2013 13:47:19 +0000
Subject: [Spce-user] rewrite rules
In-Reply-To: <8e982f2ac89360982fd9e03982d5c8d2@mail.gmail.com>
References: <8e982f2ac89360982fd9e03982d5c8d2@mail.gmail.com>
Message-ID: <E50575DD48D9B743925AC3BCE6310A4B2433E1AE@DB3PRD0710MB355.eurprd07.prod.outlook.com>

As far as I know you can't add or subtract in the replacement pattern or do any math for that matter.
However, you can create some clever match/replace to achieve a rewrite as you describe.
Personally I don't see the use of it though.

Med venlig hilsen

   Sylvester Nielsen

---------------------------------------------------------------

Fest Data A/S
Generatorvej 8D, 1.th
2730 Herlev

e-mail : sn at fest.dk<mailto:sn at fest.dk>

Tlf:     +45 4582 7389
Fax:   +45 4582 7387
Dir:    +45 6996 2011

http://www.festdata.dk

[Beskrivelse: Description: Description: Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse: cid:image003.jpg at 01CBCE06.BAABD4A0]




From: spce-user-bounces at lists.sipwise.com [mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of Matthew Ogden
Sent: 5. marts 2013 14:34
To: spce-user at lists.sipwise.com
Subject: [Spce-user] rewrite rules

Hi

Its easy enough with a rewrite rule to replace something, but does it support "adding" one number to another

For example a replace for 555-555-55XX can make 3201 become 555-555-5501. But what if you wanted 3201 to become 555-555-5506 and 3202 to become 555-555-5507?

Regards

[cid:image007.jpg at 01CE19B0.75ABB990]

Matthew Ogden
Management
TenacIT



Strategic IT Consulting * Advanced Networking * Virtualisation
Custom Development * Hosting * Syspro Support  * MS Licensing
National Tel: 041 10 10 100 | Cell: 084 205 4445 | Email: matthew at tenacit.net<mailto:matthew at tenacit.net>
CT Tel: 021 201 0333 | Skype Name: matthew.ogden | Web: http://www.tenacit.net<http://www.tenacit.net/>

[cid:image008.jpg at 01CE19B0.75ABB990] [cid:image009.png at 01CE19B0.75ABB990]  [cid:image010.jpg at 01CE19B0.75ABB990]  [cid:image011.jpg at 01CE19B0.75ABB990]


________________________________
Denne mail er blevet scannet af fest.dk ApS<http://www.fest.dk/> og indeholder ikke virus!
________________________________
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130305/c1578e8e/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image006.jpg
Type: image/jpeg
Size: 11834 bytes
Desc: image006.jpg
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130305/c1578e8e/attachment-0005.jpg>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image007.jpg
Type: image/jpeg
Size: 6077 bytes
Desc: image007.jpg
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130305/c1578e8e/attachment-0006.jpg>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image008.jpg
Type: image/jpeg
Size: 1240 bytes
Desc: image008.jpg
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130305/c1578e8e/attachment-0007.jpg>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image009.png
Type: image/png
Size: 4650 bytes
Desc: image009.png
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130305/c1578e8e/attachment-0001.png>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image010.jpg
Type: image/jpeg
Size: 1232 bytes
Desc: image010.jpg
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130305/c1578e8e/attachment-0008.jpg>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image011.jpg
Type: image/jpeg
Size: 2254 bytes
Desc: image011.jpg
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130305/c1578e8e/attachment-0009.jpg>

From dgrotti at sipwise.com  Tue Mar  5 10:14:17 2013
From: dgrotti at sipwise.com (Daniel Grotti)
Date: Tue, 05 Mar 2013 16:14:17 +0100
Subject: [Spce-user] rewrite rules
In-Reply-To: <8e982f2ac89360982fd9e03982d5c8d2@mail.gmail.com>
References: <8e982f2ac89360982fd9e03982d5c8d2@mail.gmail.com>
Message-ID: <51360BC9.9000401@sipwise.com>

Hi Matthew,

rewrite rules are based on Regular expression so you can do everything
you want.
Of course you can replace, remove or adding.

Daniel
 


On 03/05/2013 02:34 PM, Matthew Ogden wrote:
>
> Hi
>
>  
>
> Its easy enough with a rewrite rule to replace something, but does it
> support "adding" one number to another
>
>  
>
> For example a replace for 555-555-55XX can make 3201 become
> 555-555-5501. But what if you wanted 3201 to become 555-555-5506 and
> 3202 to become 555-555-5507?
>
>  
>
> Regards
>
>  
>
>  
>
> *Matthew Ogden*
>
> Management
>
> TenacIT
>
>  
>
>  
>
>  
>
> *Strategic IT Consulting *.*Advanced Networking *. *Virtualisation***
>
> *Custom Development *.***Hosting *.***Syspro Support  *. *MS Licensing***
>
> National Tel: 041 10 10 100 | Cell: 084 205 4445 | Email:
> matthew at tenacit.net <mailto:matthew at tenacit.net> 
>
> CT Tel: 021 201 0333 | Skype Name: matthew.ogden | Web:
> http://www.tenacit.net <http://www.tenacit.net/>
>
>  
>
>    
>
>  
>
>  
>
>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130305/5b1fe192/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: image/jpeg
Size: 6077 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130305/5b1fe192/attachment-0004.jpe>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: image/jpeg
Size: 1240 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130305/5b1fe192/attachment-0005.jpe>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: image/png
Size: 4650 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130305/5b1fe192/attachment-0001.png>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: image/jpeg
Size: 1232 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130305/5b1fe192/attachment-0006.jpe>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: image/jpeg
Size: 2254 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130305/5b1fe192/attachment-0007.jpe>

From axessofficetheo at gmail.com  Wed Mar  6 02:41:52 2013
From: axessofficetheo at gmail.com (Theo)
Date: Wed, 6 Mar 2013 09:41:52 +0200
Subject: [Spce-user] Upper Registration to Asterisk - transfers not
	working
In-Reply-To: <20130305075953.50b44f18@multivac>
References: <CA+9+E=X_qiRReX=c_XubTD89HEV2HApxFw1yRDgDa2nW0o_SZg@mail.gmail.com>
	<20130215213405.58b3a93a@quenya>
	<CA+9+E=Wnubt_Wy+xHjk0C6kw0XQ8tc2-agbFvkevcTpYJEqg4Q@mail.gmail.com>
	<720785848-1361030221-cardhu_decombobulator_blackberry.rim.net-863913820-@b16.c11.bise7.blackberry>
	<CA+9+E=Xh_1QHALsUzOhnUOK7q4uD8ZyoLwHoEs2R-zHh=7Ox5g@mail.gmail.com>
	<CA+9+E=U+N0GJY7QOmvTYgW5fVsetEZ_e2rYTOtp_EWeH5U7HTg@mail.gmail.com>
	<CA+9+E=U215fg8=y_ff46U2pyUxB4imXi1-i1Sx3bMbtECZ+28A@mail.gmail.com>
	<51224FFD.4020408@sipwise.com> <51225420.5000607@sipwise.com>
	<CA+9+E=U-t1paf0KW+H-vHkOwGFBkbn1=8NFd5YUPdEwm_xyZsg@mail.gmail.com>
	<5122598B.1060703@sipwise.com>
	<CA+9+E=XeYZRHuBQkhsHBTMCarq2bOdX9c+93suqyCCb-Gj+3Ag@mail.gmail.com>
	<20130226152313.2d977e1d@quenya>
	<CA+9+E=WQUrn1HbeekF2fchdZ+7kX54Snphe0QY6W_x4oXN6yyA@mail.gmail.com>
	<CA+9+E=XRB-hPxK1ZTBU5ebhnaq-y0wS0=B=8ZaeVCheBC6+tWg@mail.gmail.com>
	<CA+9+E=WU=Sa9NTZ=HJrzLXZ+VnTON_--+9fBaHtivs+Lfh7Vwg@mail.gmail.com>
	<20130305075953.50b44f18@multivac>
Message-ID: <CA+9+E=UKknSzZGHFp5qi2cLx=NdaZk879jQPQi_BvEY7ce-jAg@mail.gmail.com>

Hi

ngrep-sip gives nothing - Here's the process on a SNOM phone.

Call comes in. You put it on hold. You dial another extension and announce
the call. All that shows up with ngrep-sip. Once the call is announced you
press the transfer button. Nothing is recorded for that with ngrep sip
whatsoever, which is obviously why it doesn't work :-(.

Any ideas on that one?

On Tue, Mar 5, 2013 at 8:59 AM, Jon Bonilla <jbonilla at sipwise.com> wrote:

> El Mon, 4 Mar 2013 08:59:03 +0200
> Theo <axessofficetheo at gmail.com> escribi?:
>
> > Hi
> >
> > Not sure if I am breaking protocol by bumping - if I did - apologies. I
> am
> > almost ready to put this in as our SBC upper registration solution (more
> to
> > follow after that I hope :-) - but need to resolve the attended transfer
> > problem. Any ideas on this? Sip Header filtering is off altogether (does
> > that have any caveats?). Transfer working fine now for all phones tested,
> > but attended transfer not.
> >
>
> ngrep-sip?
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130306/e30f9de2/attachment-0001.html>

From axessofficetheo at gmail.com  Wed Mar  6 04:41:29 2013
From: axessofficetheo at gmail.com (Theo)
Date: Wed, 6 Mar 2013 11:41:29 +0200
Subject: [Spce-user] Upper Registration to Asterisk - transfers not
	working
In-Reply-To: <CA+9+E=UKknSzZGHFp5qi2cLx=NdaZk879jQPQi_BvEY7ce-jAg@mail.gmail.com>
References: <CA+9+E=X_qiRReX=c_XubTD89HEV2HApxFw1yRDgDa2nW0o_SZg@mail.gmail.com>
	<20130215213405.58b3a93a@quenya>
	<CA+9+E=Wnubt_Wy+xHjk0C6kw0XQ8tc2-agbFvkevcTpYJEqg4Q@mail.gmail.com>
	<720785848-1361030221-cardhu_decombobulator_blackberry.rim.net-863913820-@b16.c11.bise7.blackberry>
	<CA+9+E=Xh_1QHALsUzOhnUOK7q4uD8ZyoLwHoEs2R-zHh=7Ox5g@mail.gmail.com>
	<CA+9+E=U+N0GJY7QOmvTYgW5fVsetEZ_e2rYTOtp_EWeH5U7HTg@mail.gmail.com>
	<CA+9+E=U215fg8=y_ff46U2pyUxB4imXi1-i1Sx3bMbtECZ+28A@mail.gmail.com>
	<51224FFD.4020408@sipwise.com> <51225420.5000607@sipwise.com>
	<CA+9+E=U-t1paf0KW+H-vHkOwGFBkbn1=8NFd5YUPdEwm_xyZsg@mail.gmail.com>
	<5122598B.1060703@sipwise.com>
	<CA+9+E=XeYZRHuBQkhsHBTMCarq2bOdX9c+93suqyCCb-Gj+3Ag@mail.gmail.com>
	<20130226152313.2d977e1d@quenya>
	<CA+9+E=WQUrn1HbeekF2fchdZ+7kX54Snphe0QY6W_x4oXN6yyA@mail.gmail.com>
	<CA+9+E=XRB-hPxK1ZTBU5ebhnaq-y0wS0=B=8ZaeVCheBC6+tWg@mail.gmail.com>
	<CA+9+E=WU=Sa9NTZ=HJrzLXZ+VnTON_--+9fBaHtivs+Lfh7Vwg@mail.gmail.com>
	<20130305075953.50b44f18@multivac>
	<CA+9+E=UKknSzZGHFp5qi2cLx=NdaZk879jQPQi_BvEY7ce-jAg@mail.gmail.com>
Message-ID: <CA+9+E=UfxgtRPVjC_8BhcEm0kh2HSKnErhro7WrFJ-eLu8SzHA@mail.gmail.com>

A little more now - I upgraded the firmware of a SNOM phone. Following the
same process, upon pressing the transfer button I now get this:

U 2013/03/06 11:36:22.236542 41.177.66.116:22106 -> 196.41.123.113:5060
SIP/2.0 200 OK'
Via: SIP/2.0/UDP
196.41.123.113;branch=z9hG4bKaa2.bc57c1db3d0853f982c750b38e5926b2.0'
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK~IoIoamR;rport=5080'
Record-Route:
<sip:196.41.123.113;lr;r2=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
Record-Route:
<sip:127.0.0.1;r2=on;lr=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
Contact: <sip:user400 at 192.168.10.88:5060
;rinstance=7e064df24c43ccf0;transport=UDP>'
To: <sip:user400 at domain.domain.com>;tag=d37f6717'
From: <sip:2721403 at domain.domain.com
>;tag=304E3AA0-51370E09000C6324-2EE19700'
Call-ID: 51370e076928-0af4wphlmirc_b2b-1'
CSeq: 11 INVITE'
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
SUBSCRIBE'
Content-Type: application/sdp'
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri'
User-Agent: Zoiper rev.11619'
Allow-Events: presence, kpml'
Content-Length: 185'
'
v=0'
o=Z 0 3 IN IP4 192.168.10.88'
s=Z'
c=IN IP4 192.168.10.88'
t=0 0'
m=audio 8000 RTP/AVP 8 101'
a=rtpmap:8 PCMA/8000'
a=rtpmap:101 telephone-event/8000'
a=fmtp:101 0-15'
a=inactive'

#
U 2013/03/06 11:36:22.236896 127.0.0.1:5060 -> 127.0.0.1:5080
SIP/2.0 200 OK'
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK~IoIoamR;rport=5080'
Record-Route:
<sip:196.41.123.113;lr;r2=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
Record-Route:
<sip:127.0.0.1;r2=on;lr=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
Contact: <sip:user400 at 192.168.10.88:5060
;alias=41.177.66.116~22106~1;rinstance=7e064df24c43ccf0;transport=UDP>'
To: <sip:user400 at domain.domain.com>;tag=d37f6717'
From: <sip:2721403 at domain.domain.com
>;tag=304E3AA0-51370E09000C6324-2EE19700'
Call-ID: 51370e076928-0af4wphlmirc_b2b-1'
CSeq: 11 INVITE'
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
SUBSCRIBE'
Content-Type: application/sdp'
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri'
User-Agent: Zoiper rev.11619'
Allow-Events: presence, kpml'
Content-Length: 233'
P-NGCP-Src-Ip: 41.177.66.116'
P-NGCP-Src-Port: 22106'
P-NGCP-Src-Proto: udp'
P-NGCP-Src-Af: 4'
'
v=0'
o=Z 0 3 IN IP4 192.168.10.88'
s=Z'
c=IN IP4 41.177.66.116'
t=0 0'
m=audio 8000 RTP/AVP 8 101'
a=rtpmap:8 PCMA/8000'
a=rtpmap:101 telephone-event/8000'
a=fmtp:101 0-15'
a=inactive'
a=direction:active'
a=oldmediaip:192.168.10.88'

#
U 2013/03/06 11:36:22.236968 127.0.0.1:5080 -> 127.0.0.1:5060
ACK sip:user400 at 192.168.10.88:5060;alias=41.177.66.116~22106~1;rinstance=7e064df24c43ccf0;transport=UDP
SIP/2.0'
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bKTbHrca0H;rport'
From: <sip:2721403 at domain.domain.com
>;tag=304E3AA0-51370E09000C6324-2EE19700'
To: <sip:user400 at domain.domain.com>;tag=d37f6717'
CSeq: 11 ACK'
Call-ID: 51370e076928-0af4wphlmirc_b2b-1'
Contact: <sip:127.0.0.1:5080>'
Route:
<sip:127.0.0.1;r2=on;lr=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>,
<sip:196.41.123.113;lr;r2=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
Max-Forwards: 68'
Content-Length: 0'
'

##
U 2013/03/06 11:36:22.237048 196.41.123.113:5060 -> 41.177.66.116:22106
ACK sip:user400 at 192.168.10.88:5060;rinstance=7e064df24c43ccf0;transport=UDP
SIP/2.0'
Record-Route:
<sip:196.41.123.113;r2=on;lr=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
Record-Route:
<sip:127.0.0.1;r2=on;lr=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
Via: SIP/2.0/UDP
196.41.123.113;branch=z9hG4bKaa2.04ec964b43561d47cbdd4d22812c93a8.0'
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bKTbHrca0H;rport=5080'
From: <sip:2721403 at domain.domain.com
>;tag=304E3AA0-51370E09000C6324-2EE19700'
To: <sip:user400 at domain.domain.com>;tag=d37f6717'
CSeq: 11 ACK'
Call-ID: 51370e076928-0af4wphlmirc_b2b-1'
Max-Forwards: 67'
Content-Length: 0'
Contact: <sip:ngcp-lb at 196.41.123.113:5060;ngcpct='sip:127.0.0.1:5080'>'


On Wed, Mar 6, 2013 at 9:41 AM, Theo <axessofficetheo at gmail.com> wrote:

> Hi
>
> ngrep-sip gives nothing - Here's the process on a SNOM phone.
>
> Call comes in. You put it on hold. You dial another extension and announce
> the call. All that shows up with ngrep-sip. Once the call is announced you
> press the transfer button. Nothing is recorded for that with ngrep sip
> whatsoever, which is obviously why it doesn't work :-(.
>
> Any ideas on that one?
>
>
> On Tue, Mar 5, 2013 at 8:59 AM, Jon Bonilla <jbonilla at sipwise.com> wrote:
>
>> El Mon, 4 Mar 2013 08:59:03 +0200
>> Theo <axessofficetheo at gmail.com> escribi?:
>>
>> > Hi
>> >
>> > Not sure if I am breaking protocol by bumping - if I did - apologies. I
>> am
>> > almost ready to put this in as our SBC upper registration solution
>> (more to
>> > follow after that I hope :-) - but need to resolve the attended transfer
>> > problem. Any ideas on this? Sip Header filtering is off altogether (does
>> > that have any caveats?). Transfer working fine now for all phones
>> tested,
>> > but attended transfer not.
>> >
>>
>> ngrep-sip?
>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130306/b2ede2ad/attachment-0001.html>

From axessofficetheo at gmail.com  Wed Mar  6 05:45:35 2013
From: axessofficetheo at gmail.com (Theo)
Date: Wed, 6 Mar 2013 12:45:35 +0200
Subject: [Spce-user] Upper Registration to Asterisk - transfers not
	working
In-Reply-To: <CA+9+E=UfxgtRPVjC_8BhcEm0kh2HSKnErhro7WrFJ-eLu8SzHA@mail.gmail.com>
References: <CA+9+E=X_qiRReX=c_XubTD89HEV2HApxFw1yRDgDa2nW0o_SZg@mail.gmail.com>
	<20130215213405.58b3a93a@quenya>
	<CA+9+E=Wnubt_Wy+xHjk0C6kw0XQ8tc2-agbFvkevcTpYJEqg4Q@mail.gmail.com>
	<720785848-1361030221-cardhu_decombobulator_blackberry.rim.net-863913820-@b16.c11.bise7.blackberry>
	<CA+9+E=Xh_1QHALsUzOhnUOK7q4uD8ZyoLwHoEs2R-zHh=7Ox5g@mail.gmail.com>
	<CA+9+E=U+N0GJY7QOmvTYgW5fVsetEZ_e2rYTOtp_EWeH5U7HTg@mail.gmail.com>
	<CA+9+E=U215fg8=y_ff46U2pyUxB4imXi1-i1Sx3bMbtECZ+28A@mail.gmail.com>
	<51224FFD.4020408@sipwise.com> <51225420.5000607@sipwise.com>
	<CA+9+E=U-t1paf0KW+H-vHkOwGFBkbn1=8NFd5YUPdEwm_xyZsg@mail.gmail.com>
	<5122598B.1060703@sipwise.com>
	<CA+9+E=XeYZRHuBQkhsHBTMCarq2bOdX9c+93suqyCCb-Gj+3Ag@mail.gmail.com>
	<20130226152313.2d977e1d@quenya>
	<CA+9+E=WQUrn1HbeekF2fchdZ+7kX54Snphe0QY6W_x4oXN6yyA@mail.gmail.com>
	<CA+9+E=XRB-hPxK1ZTBU5ebhnaq-y0wS0=B=8ZaeVCheBC6+tWg@mail.gmail.com>
	<CA+9+E=WU=Sa9NTZ=HJrzLXZ+VnTON_--+9fBaHtivs+Lfh7Vwg@mail.gmail.com>
	<20130305075953.50b44f18@multivac>
	<CA+9+E=UKknSzZGHFp5qi2cLx=NdaZk879jQPQi_BvEY7ce-jAg@mail.gmail.com>
	<CA+9+E=UfxgtRPVjC_8BhcEm0kh2HSKnErhro7WrFJ-eLu8SzHA@mail.gmail.com>
Message-ID: <CA+9+E=X3bfBdtmTSy=TycC=veshUU8Wgb8H6HCyEzjvW1RpSJg@mail.gmail.com>

And the SIP trace from the SNOM logs. I don't see any refer-to being sent
from the phone....


Sent to udp:196.41.123.113:5060 at 6/3/2013 12:18:02:571 (1617 bytes):

INVITE sip:ngcp-lb at 196.41.123.113:5060;ngcpct='sip:127.0.0.1:5080' SIP/2.0
Via: SIP/2.0/UDP 192.168.10.62:2050;branch=z9hG4bK-ttlvkk77nm7u;rport
Route: <sip:196.41.123.113;r2=on;lr=on;ftag=v0z1vszklt;nat=yes;ngcplb=yes>
Route: <sip:127.0.0.1;r2=on;lr=on;ftag=v0z1vszklt;nat=yes;ngcplb=yes>
Route: <sip:127.0.0.1:5062
;lr=on;ftag=v0z1vszklt;did=da1.e171;mpd=ii;rtpprx=yes;vsf=V3V3ZwIILyJUOU1ORmtBYW50Z2JXdXdnRUx4YVQ5TU5mS2E->
From: "AA Office" <sip:user403 at domain.domain.com>;tag=v0z1vszklt
To: <sip:400 at domain.domain.com
;user=phone>;tag=6E486C78-513717CB00030190-2E40F700
Call-ID: 513717c8ba36-755kmq45j2i5
CSeq: 3 INVITE
Max-Forwards: 70
Contact: <sip:user403 at 192.168.10.62:2050;line=pxk7nlcg>;reg-id=1
X-Serialnumber: 0004132FAC49
P-Key-Flags: keys="3"
User-Agent: snom300/8.7.3.19
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 489

v=0
o=root 1854415615 1854415616 IN IP4 192.168.10.62
s=call
c=IN IP4 192.168.10.62
t=0 0
m=audio 64268 RTP/AVP 9 0 8 3 99 108 18 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:fhqyKsE9KbQmap0NxeGKIFRZ3Yj2so3mw/32wC5U
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:108 AAL2-G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendonly
Received from udp:196.41.123.113:5060 at 6/3/2013 12:18:02:719 (386 bytes):

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.62:2050
;received=41.177.66.139;branch=z9hG4bK-ttlvkk77nm7u;rport=14297
From: "AA Office" <sip:user403 at domain.domain.com>;tag=v0z1vszklt
To: <sip:400 at domain.domain.com
;user=phone>;tag=6E486C78-513717CB00030190-2E40F700
Call-ID: 513717c8ba36-755kmq45j2i5
CSeq: 3 INVITE
Server: Sipwise NGCP Proxy 2.X
Content-Length: 0

Received from udp:196.41.123.113:5060 at 6/3/2013 12:18:03:231 (1253 bytes):

SIP/2.0 200 OK
Record-Route: <sip:127.0.0.1:5062;lr=on;ftag=v0z1vszklt;rtpprx=yes;mpd=ii>
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=v0z1vszklt;nat=yes;ngcplb=yes>
Record-Route:
<sip:196.41.123.113;r2=on;lr=on;ftag=v0z1vszklt;nat=yes;ngcplb=yes>
Via: SIP/2.0/UDP 192.168.10.62:2050
;received=41.177.66.139;branch=z9hG4bK-ttlvkk77nm7u;rport=14297
From: "AA Office" <sip:user403 at domain.domain.com>;tag=v0z1vszklt
To: <sip:400 at domain.domain.com
;user=phone>;tag=6E486C78-513717CB00030190-2E40F700
Call-ID: 513717c8ba36-755kmq45j2i5
CSeq: 3 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
SUBSCRIBE
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper rev.11619
Allow-Events: presence, kpml
P-NGCP-Src-Ip: 41.177.66.116
P-NGCP-Src-Port: 22106
P-NGCP-Src-Proto: udp
P-NGCP-Src-Af: 4
Content-Type: application/sdp
Content-Length: 254
Contact: <sip:ngcp-lb at 196.41.123.113:5060;ngcpct='sip:127.0.0.1:5080'>

v=0
o=Z 0 3 IN IP4 196.41.123.113
s=Z
c=IN IP4 196.41.123.113
t=0 0
m=audio 37342 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=inactive
a=direction:active
a=oldmediaip:192.168.10.88
a=nortpproxy:yes
Sent to udp:196.41.123.113:5060 at 6/3/2013 12:18:03:255 (741 bytes):

ACK sip:ngcp-lb at 196.41.123.113:5060;ngcpct='sip:127.0.0.1:5080' SIP/2.0
Via: SIP/2.0/UDP 192.168.10.62:2050;branch=z9hG4bK-cntszxfllm77;rport
Route: <sip:196.41.123.113;r2=on;lr=on;ftag=v0z1vszklt;nat=yes;ngcplb=yes>
Route: <sip:127.0.0.1;r2=on;lr=on;ftag=v0z1vszklt;nat=yes;ngcplb=yes>
Route: <sip:127.0.0.1:5062
;lr=on;ftag=v0z1vszklt;did=da1.e171;mpd=ii;rtpprx=yes;vsf=V3V3ZwIILyJUOU1ORmtBYW50Z2JXdXdnRUx4YVQ5TU5mS2E->
From: "AA Office" <sip:user403 at domain.domain.com>;tag=v0z1vszklt
To: <sip:400 at domain.domain.com
;user=phone>;tag=6E486C78-513717CB00030190-2E40F700
Call-ID: 513717c8ba36-755kmq45j2i5
CSeq: 3 ACK
Max-Forwards: 70
Contact: <sip:user403 at 192.168.10.62:2050;line=pxk7nlcg>;reg-id=1
Content-Length: 0


On Wed, Mar 6, 2013 at 11:41 AM, Theo <axessofficetheo at gmail.com> wrote:

> A little more now - I upgraded the firmware of a SNOM phone. Following the
> same process, upon pressing the transfer button I now get this:
>
> U 2013/03/06 11:36:22.236542 41.177.66.116:22106 -> 196.41.123.113:5060
> SIP/2.0 200 OK'
> Via: SIP/2.0/UDP
> 196.41.123.113;branch=z9hG4bKaa2.bc57c1db3d0853f982c750b38e5926b2.0'
> Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK~IoIoamR;rport=5080'
> Record-Route:
> <sip:196.41.123.113;lr;r2=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
> Record-Route:
> <sip:127.0.0.1;r2=on;lr=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
> Contact: <sip:user400 at 192.168.10.88:5060
> ;rinstance=7e064df24c43ccf0;transport=UDP>'
> To: <sip:user400 at domain.domain.com>;tag=d37f6717'
> From: <sip:2721403 at domain.domain.com
> >;tag=304E3AA0-51370E09000C6324-2EE19700'
> Call-ID: 51370e076928-0af4wphlmirc_b2b-1'
> CSeq: 11 INVITE'
> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
> SUBSCRIBE'
> Content-Type: application/sdp'
> Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri'
> User-Agent: Zoiper rev.11619'
> Allow-Events: presence, kpml'
> Content-Length: 185'
> '
> v=0'
> o=Z 0 3 IN IP4 192.168.10.88'
> s=Z'
> c=IN IP4 192.168.10.88'
> t=0 0'
> m=audio 8000 RTP/AVP 8 101'
> a=rtpmap:8 PCMA/8000'
> a=rtpmap:101 telephone-event/8000'
> a=fmtp:101 0-15'
> a=inactive'
>
> #
> U 2013/03/06 11:36:22.236896 127.0.0.1:5060 -> 127.0.0.1:5080
> SIP/2.0 200 OK'
> Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK~IoIoamR;rport=5080'
> Record-Route:
> <sip:196.41.123.113;lr;r2=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
> Record-Route:
> <sip:127.0.0.1;r2=on;lr=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
> Contact: <sip:user400 at 192.168.10.88:5060
> ;alias=41.177.66.116~22106~1;rinstance=7e064df24c43ccf0;transport=UDP>'
> To: <sip:user400 at domain.domain.com>;tag=d37f6717'
> From: <sip:2721403 at domain.domain.com
> >;tag=304E3AA0-51370E09000C6324-2EE19700'
> Call-ID: 51370e076928-0af4wphlmirc_b2b-1'
> CSeq: 11 INVITE'
> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
> SUBSCRIBE'
> Content-Type: application/sdp'
> Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri'
> User-Agent: Zoiper rev.11619'
> Allow-Events: presence, kpml'
> Content-Length: 233'
> P-NGCP-Src-Ip: 41.177.66.116'
> P-NGCP-Src-Port: 22106'
> P-NGCP-Src-Proto: udp'
> P-NGCP-Src-Af: 4'
> '
> v=0'
> o=Z 0 3 IN IP4 192.168.10.88'
> s=Z'
> c=IN IP4 41.177.66.116'
> t=0 0'
> m=audio 8000 RTP/AVP 8 101'
> a=rtpmap:8 PCMA/8000'
> a=rtpmap:101 telephone-event/8000'
> a=fmtp:101 0-15'
> a=inactive'
> a=direction:active'
> a=oldmediaip:192.168.10.88'
>
> #
> U 2013/03/06 11:36:22.236968 127.0.0.1:5080 -> 127.0.0.1:5060
> ACK sip:user400 at 192.168.10.88:5060;alias=41.177.66.116~22106~1;rinstance=7e064df24c43ccf0;transport=UDP
> SIP/2.0'
> Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bKTbHrca0H;rport'
> From: <sip:2721403 at domain.domain.com
> >;tag=304E3AA0-51370E09000C6324-2EE19700'
> To: <sip:user400 at domain.domain.com>;tag=d37f6717'
> CSeq: 11 ACK'
> Call-ID: 51370e076928-0af4wphlmirc_b2b-1'
> Contact: <sip:127.0.0.1:5080>'
> Route:
> <sip:127.0.0.1;r2=on;lr=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>,
> <sip:196.41.123.113;lr;r2=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
> Max-Forwards: 68'
> Content-Length: 0'
> '
>
> ##
> U 2013/03/06 11:36:22.237048 196.41.123.113:5060 -> 41.177.66.116:22106
> ACK sip:user400 at 192.168.10.88:5060;rinstance=7e064df24c43ccf0;transport=UDP
> SIP/2.0'
> Record-Route:
> <sip:196.41.123.113;r2=on;lr=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
> Record-Route:
> <sip:127.0.0.1;r2=on;lr=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
> Via: SIP/2.0/UDP
> 196.41.123.113;branch=z9hG4bKaa2.04ec964b43561d47cbdd4d22812c93a8.0'
> Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bKTbHrca0H;rport=5080'
> From: <sip:2721403 at domain.domain.com
> >;tag=304E3AA0-51370E09000C6324-2EE19700'
> To: <sip:user400 at domain.domain.com>;tag=d37f6717'
> CSeq: 11 ACK'
> Call-ID: 51370e076928-0af4wphlmirc_b2b-1'
> Max-Forwards: 67'
> Content-Length: 0'
> Contact: <sip:ngcp-lb at 196.41.123.113:5060;ngcpct='sip:127.0.0.1:5080'>'
>
>
> On Wed, Mar 6, 2013 at 9:41 AM, Theo <axessofficetheo at gmail.com> wrote:
>
>> Hi
>>
>> ngrep-sip gives nothing - Here's the process on a SNOM phone.
>>
>> Call comes in. You put it on hold. You dial another extension and
>> announce the call. All that shows up with ngrep-sip. Once the call is
>> announced you press the transfer button. Nothing is recorded for that with
>> ngrep sip whatsoever, which is obviously why it doesn't work :-(.
>>
>> Any ideas on that one?
>>
>>
>> On Tue, Mar 5, 2013 at 8:59 AM, Jon Bonilla <jbonilla at sipwise.com> wrote:
>>
>>> El Mon, 4 Mar 2013 08:59:03 +0200
>>> Theo <axessofficetheo at gmail.com> escribi?:
>>>
>>> > Hi
>>> >
>>> > Not sure if I am breaking protocol by bumping - if I did - apologies.
>>> I am
>>> > almost ready to put this in as our SBC upper registration solution
>>> (more to
>>> > follow after that I hope :-) - but need to resolve the attended
>>> transfer
>>> > problem. Any ideas on this? Sip Header filtering is off altogether
>>> (does
>>> > that have any caveats?). Transfer working fine now for all phones
>>> tested,
>>> > but attended transfer not.
>>> >
>>>
>>> ngrep-sip?
>>>
>>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130306/7d4f7ea9/attachment-0001.html>

From admartin at nixus.es  Wed Mar  6 06:35:45 2013
From: admartin at nixus.es (Antonio David Martin)
Date: Wed, 6 Mar 2013 11:35:45 +0000
Subject: [Spce-user] live monitoring from shell
Message-ID: <70A7DFAD7F7A5A4EB0F34A47468371284E41C432@ACR01WIN004.grupoacr.com>

Is there any way to get the active calls from the command line? Something similar as in Asterisk (Asterisk 1.8.8.0)

#core show channels verbose
pbx*CLI> core show channels verbose
Channel              Context              Extension        Prio State   Application  Data                      CallerID        Duration Accountcode PeerAccount BridgedTo
SIP/nova105-00005 from-nova         102                 1 Up      Dial         SIP/nova102,,xXt       nova105      00:00:16                         SIP/nova102-00005
SIP/nova101-00005 from-nova         968777777           1 Up      Congestion   (Empty)                   34902000000     00:00:06                         (None)
SIP/oco103-000059c from-oco           1                   1 Up      AppQueue     (Outgoing Line)                           00:01:30                         SIP/asterisk-000059c
SIP/nova102-00005 from-nova                             1 Up      AppDial      (Outgoing Line)           102             00:00:15                         SIP/nova105-00005
SIP/asterisk-000059c incoming-oco       1                   2 Up      Queue        oco-atencionalcliente,t 607000000       00:01:40                         SIP/oco103-000059c
5 active channels
3 active calls
10785 calls processed

I have seen this command   "ngcp-kamctl proxy fifo get_statistics :dialog::all"  but It doesn't show the subscribers, source, destination, etc

Thanks in advance

David Mart?n



-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130306/b2f11a61/attachment-0001.html>

From apogrebennyk at sipwise.com  Wed Mar  6 11:09:31 2013
From: apogrebennyk at sipwise.com (Andrew Pogrebennyk)
Date: Wed, 06 Mar 2013 17:09:31 +0100
Subject: [Spce-user] live monitoring from shell
In-Reply-To: <70A7DFAD7F7A5A4EB0F34A47468371284E41C432@ACR01WIN004.grupoacr.com>
References: <70A7DFAD7F7A5A4EB0F34A47468371284E41C432@ACR01WIN004.grupoacr.com>
Message-ID: <51376A3B.6030301@sipwise.com>

Hi David,

just do:
ngcp-kamctl proxy fifo dlg_list

Also you may want to check the sems project for call monitoring tools.

HTH.
Andrew

On 03/06/2013 12:35 PM, Antonio David Martin wrote:
> Is there any way to get the active calls from the command line?
> Something similar as in Asterisk (Asterisk 1.8.8.0)
> 
>  
> 
> #core show channels verbose
> 
> pbx*CLI> core show channels verbose
> 
> Channel              Context              Extension        Prio State  
> Application  Data                      CallerID        Duration
> Accountcode PeerAccount BridgedTo          
> 
> SIP/nova105-00005 from-nova         102                 1 Up     
> Dial         SIP/nova102,,xXt       nova105     
> 00:00:16                         SIP/nova102-00005
> 
> SIP/nova101-00005 from-nova         968777777           1 Up     
> Congestion   (Empty)                   34902000000    
> 00:00:06                         (None)             
> 
> SIP/oco103-000059c from-oco           1                   1 Up     
> AppQueue     (Outgoing Line)                          
> 00:01:30                         SIP/asterisk-000059c
> 
> SIP/nova102-00005 from-nova                             1 Up     
> AppDial      (Outgoing Line)           102            
> 00:00:15                         SIP/nova105-00005
> 
> SIP/asterisk-000059c incoming-oco       1                   2 Up     
> Queue        oco-atencionalcliente,t 607000000      
> 00:01:40                         SIP/oco103-000059c
> 
> 5 active channels
> 
> 3 active calls
> 
> 10785 calls processed
> 
>  
> 
> I have seen this command   ?ngcp-kamctl proxy fifo get_statistics
> :dialog::all?  but It doesn?t show the subscribers, source, destination, etc
> 
>  
> 
> Thanks in advance
> 
>  
> 
> *David Mart?n **
> 
> 
> ***
> 
>  
> 
> 
> 
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
> 




From axessofficetheo at gmail.com  Wed Mar  6 11:28:59 2013
From: axessofficetheo at gmail.com (Theo)
Date: Wed, 6 Mar 2013 18:28:59 +0200
Subject: [Spce-user] Upper Registration to Asterisk - transfers not
	working
In-Reply-To: <CA+9+E=X3bfBdtmTSy=TycC=veshUU8Wgb8H6HCyEzjvW1RpSJg@mail.gmail.com>
References: <CA+9+E=X_qiRReX=c_XubTD89HEV2HApxFw1yRDgDa2nW0o_SZg@mail.gmail.com>
	<20130215213405.58b3a93a@quenya>
	<CA+9+E=Wnubt_Wy+xHjk0C6kw0XQ8tc2-agbFvkevcTpYJEqg4Q@mail.gmail.com>
	<720785848-1361030221-cardhu_decombobulator_blackberry.rim.net-863913820-@b16.c11.bise7.blackberry>
	<CA+9+E=Xh_1QHALsUzOhnUOK7q4uD8ZyoLwHoEs2R-zHh=7Ox5g@mail.gmail.com>
	<CA+9+E=U+N0GJY7QOmvTYgW5fVsetEZ_e2rYTOtp_EWeH5U7HTg@mail.gmail.com>
	<CA+9+E=U215fg8=y_ff46U2pyUxB4imXi1-i1Sx3bMbtECZ+28A@mail.gmail.com>
	<51224FFD.4020408@sipwise.com> <51225420.5000607@sipwise.com>
	<CA+9+E=U-t1paf0KW+H-vHkOwGFBkbn1=8NFd5YUPdEwm_xyZsg@mail.gmail.com>
	<5122598B.1060703@sipwise.com>
	<CA+9+E=XeYZRHuBQkhsHBTMCarq2bOdX9c+93suqyCCb-Gj+3Ag@mail.gmail.com>
	<20130226152313.2d977e1d@quenya>
	<CA+9+E=WQUrn1HbeekF2fchdZ+7kX54Snphe0QY6W_x4oXN6yyA@mail.gmail.com>
	<CA+9+E=XRB-hPxK1ZTBU5ebhnaq-y0wS0=B=8ZaeVCheBC6+tWg@mail.gmail.com>
	<CA+9+E=WU=Sa9NTZ=HJrzLXZ+VnTON_--+9fBaHtivs+Lfh7Vwg@mail.gmail.com>
	<20130305075953.50b44f18@multivac>
	<CA+9+E=UKknSzZGHFp5qi2cLx=NdaZk879jQPQi_BvEY7ce-jAg@mail.gmail.com>
	<CA+9+E=UfxgtRPVjC_8BhcEm0kh2HSKnErhro7WrFJ-eLu8SzHA@mail.gmail.com>
	<CA+9+E=X3bfBdtmTSy=TycC=veshUU8Wgb8H6HCyEzjvW1RpSJg@mail.gmail.com>
Message-ID: <CA+9+E=V-nzXP5CeQMCBKYkyp-oJ3xKiVmKgj5D9UpBoh_363pg@mail.gmail.com>

Tried with some other phones and it worked, so it seems specific to the
SNOM phones. Of course we have over a thousand SNOM phones out there so for
us to start using this we'd have to find a fix. My guess is that SNOM needs
to fix it, but don't have the required ammo nor knowledge to take them to
task on it (unless I am completely wrong to start off with). Aside from
that, I doubt whether they would fix that in a matter of days and I had
hoped to start implementing this as a solution.

Anything that can be done on the scpe to help with this? Any words of
wisdom I can share with SNOM?

On Wed, Mar 6, 2013 at 12:45 PM, Theo <axessofficetheo at gmail.com> wrote:

> And the SIP trace from the SNOM logs. I don't see any refer-to being sent
> from the phone....
>
>
> Sent to udp:196.41.123.113:5060 at 6/3/2013 12:18:02:571 (1617 bytes):
>
> INVITE sip:ngcp-lb at 196.41.123.113:5060;ngcpct='sip:127.0.0.1:5080' SIP/2.0
> Via: SIP/2.0/UDP 192.168.10.62:2050;branch=z9hG4bK-ttlvkk77nm7u;rport
> Route: <sip:196.41.123.113;r2=on;lr=on;ftag=v0z1vszklt;nat=yes;ngcplb=yes>
> Route: <sip:127.0.0.1;r2=on;lr=on;ftag=v0z1vszklt;nat=yes;ngcplb=yes>
> Route: <sip:127.0.0.1:5062
> ;lr=on;ftag=v0z1vszklt;did=da1.e171;mpd=ii;rtpprx=yes;vsf=V3V3ZwIILyJUOU1ORmtBYW50Z2JXdXdnRUx4YVQ5TU5mS2E->
> From: "AA Office" <sip:user403 at domain.domain.com>;tag=v0z1vszklt
> To: <sip:400 at domain.domain.com
> ;user=phone>;tag=6E486C78-513717CB00030190-2E40F700
> Call-ID: 513717c8ba36-755kmq45j2i5
> CSeq: 3 INVITE
> Max-Forwards: 70
> Contact: <sip:user403 at 192.168.10.62:2050;line=pxk7nlcg>;reg-id=1
> X-Serialnumber: 0004132FAC49
> P-Key-Flags: keys="3"
> User-Agent: snom300/8.7.3.19
> Accept: application/sdp
> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
> MESSAGE, INFO, UPDATE
> Allow-Events: talk, hold, refer, call-info
> Supported: timer, 100rel, replaces, from-change
> Session-Expires: 3600;refresher=uas
> Min-SE: 90
> Content-Type: application/sdp
> Content-Length: 489
>
> v=0
> o=root 1854415615 1854415616 IN IP4 192.168.10.62
> s=call
> c=IN IP4 192.168.10.62
> t=0 0
> m=audio 64268 RTP/AVP 9 0 8 3 99 108 18 101
> a=crypto:1 AES_CM_128_HMAC_SHA1_32
> inline:fhqyKsE9KbQmap0NxeGKIFRZ3Yj2so3mw/32wC5U
> a=rtpmap:9 G722/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:99 G726-32/8000
> a=rtpmap:108 AAL2-G726-32/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
> a=sendonly
> Received from udp:196.41.123.113:5060 at 6/3/2013 12:18:02:719 (386
> bytes):
>
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 192.168.10.62:2050
> ;received=41.177.66.139;branch=z9hG4bK-ttlvkk77nm7u;rport=14297
> From: "AA Office" <sip:user403 at domain.domain.com>;tag=v0z1vszklt
> To: <sip:400 at domain.domain.com
> ;user=phone>;tag=6E486C78-513717CB00030190-2E40F700
> Call-ID: 513717c8ba36-755kmq45j2i5
> CSeq: 3 INVITE
> Server: Sipwise NGCP Proxy 2.X
> Content-Length: 0
>
> Received from udp:196.41.123.113:5060 at 6/3/2013 12:18:03:231 (1253
> bytes):
>
> SIP/2.0 200 OK
> Record-Route: <sip:127.0.0.1:5062;lr=on;ftag=v0z1vszklt;rtpprx=yes;mpd=ii>
> Record-Route:
> <sip:127.0.0.1;r2=on;lr=on;ftag=v0z1vszklt;nat=yes;ngcplb=yes>
> Record-Route:
> <sip:196.41.123.113;r2=on;lr=on;ftag=v0z1vszklt;nat=yes;ngcplb=yes>
> Via: SIP/2.0/UDP 192.168.10.62:2050
> ;received=41.177.66.139;branch=z9hG4bK-ttlvkk77nm7u;rport=14297
> From: "AA Office" <sip:user403 at domain.domain.com>;tag=v0z1vszklt
> To: <sip:400 at domain.domain.com
> ;user=phone>;tag=6E486C78-513717CB00030190-2E40F700
> Call-ID: 513717c8ba36-755kmq45j2i5
> CSeq: 3 INVITE
> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
> SUBSCRIBE
> Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
> User-Agent: Zoiper rev.11619
> Allow-Events: presence, kpml
> P-NGCP-Src-Ip: 41.177.66.116
> P-NGCP-Src-Port: 22106
> P-NGCP-Src-Proto: udp
> P-NGCP-Src-Af: 4
> Content-Type: application/sdp
> Content-Length: 254
> Contact: <sip:ngcp-lb at 196.41.123.113:5060;ngcpct='sip:127.0.0.1:5080'>
>
> v=0
> o=Z 0 3 IN IP4 196.41.123.113
> s=Z
> c=IN IP4 196.41.123.113
> t=0 0
> m=audio 37342 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=inactive
> a=direction:active
> a=oldmediaip:192.168.10.88
> a=nortpproxy:yes
> Sent to udp:196.41.123.113:5060 at 6/3/2013 12:18:03:255 (741 bytes):
>
> ACK sip:ngcp-lb at 196.41.123.113:5060;ngcpct='sip:127.0.0.1:5080' SIP/2.0
> Via: SIP/2.0/UDP 192.168.10.62:2050;branch=z9hG4bK-cntszxfllm77;rport
> Route: <sip:196.41.123.113;r2=on;lr=on;ftag=v0z1vszklt;nat=yes;ngcplb=yes>
> Route: <sip:127.0.0.1;r2=on;lr=on;ftag=v0z1vszklt;nat=yes;ngcplb=yes>
> Route: <sip:127.0.0.1:5062
> ;lr=on;ftag=v0z1vszklt;did=da1.e171;mpd=ii;rtpprx=yes;vsf=V3V3ZwIILyJUOU1ORmtBYW50Z2JXdXdnRUx4YVQ5TU5mS2E->
> From: "AA Office" <sip:user403 at domain.domain.com>;tag=v0z1vszklt
> To: <sip:400 at domain.domain.com
> ;user=phone>;tag=6E486C78-513717CB00030190-2E40F700
> Call-ID: 513717c8ba36-755kmq45j2i5
> CSeq: 3 ACK
> Max-Forwards: 70
> Contact: <sip:user403 at 192.168.10.62:2050;line=pxk7nlcg>;reg-id=1
> Content-Length: 0
>
>
> On Wed, Mar 6, 2013 at 11:41 AM, Theo <axessofficetheo at gmail.com> wrote:
>
>> A little more now - I upgraded the firmware of a SNOM phone. Following
>> the same process, upon pressing the transfer button I now get this:
>>
>> U 2013/03/06 11:36:22.236542 41.177.66.116:22106 -> 196.41.123.113:5060
>> SIP/2.0 200 OK'
>> Via: SIP/2.0/UDP
>> 196.41.123.113;branch=z9hG4bKaa2.bc57c1db3d0853f982c750b38e5926b2.0'
>> Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK~IoIoamR;rport=5080'
>> Record-Route:
>> <sip:196.41.123.113;lr;r2=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
>> Record-Route:
>> <sip:127.0.0.1;r2=on;lr=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
>> Contact: <sip:user400 at 192.168.10.88:5060
>> ;rinstance=7e064df24c43ccf0;transport=UDP>'
>> To: <sip:user400 at domain.domain.com>;tag=d37f6717'
>> From: <sip:2721403 at domain.domain.com
>> >;tag=304E3AA0-51370E09000C6324-2EE19700'
>> Call-ID: 51370e076928-0af4wphlmirc_b2b-1'
>> CSeq: 11 INVITE'
>> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
>> SUBSCRIBE'
>> Content-Type: application/sdp'
>> Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri'
>> User-Agent: Zoiper rev.11619'
>> Allow-Events: presence, kpml'
>> Content-Length: 185'
>> '
>> v=0'
>> o=Z 0 3 IN IP4 192.168.10.88'
>> s=Z'
>> c=IN IP4 192.168.10.88'
>> t=0 0'
>> m=audio 8000 RTP/AVP 8 101'
>> a=rtpmap:8 PCMA/8000'
>> a=rtpmap:101 telephone-event/8000'
>> a=fmtp:101 0-15'
>> a=inactive'
>>
>> #
>> U 2013/03/06 11:36:22.236896 127.0.0.1:5060 -> 127.0.0.1:5080
>> SIP/2.0 200 OK'
>> Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK~IoIoamR;rport=5080'
>> Record-Route:
>> <sip:196.41.123.113;lr;r2=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
>> Record-Route:
>> <sip:127.0.0.1;r2=on;lr=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
>> Contact: <sip:user400 at 192.168.10.88:5060
>> ;alias=41.177.66.116~22106~1;rinstance=7e064df24c43ccf0;transport=UDP>'
>> To: <sip:user400 at domain.domain.com>;tag=d37f6717'
>> From: <sip:2721403 at domain.domain.com
>> >;tag=304E3AA0-51370E09000C6324-2EE19700'
>> Call-ID: 51370e076928-0af4wphlmirc_b2b-1'
>> CSeq: 11 INVITE'
>> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
>> SUBSCRIBE'
>> Content-Type: application/sdp'
>> Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri'
>> User-Agent: Zoiper rev.11619'
>> Allow-Events: presence, kpml'
>> Content-Length: 233'
>> P-NGCP-Src-Ip: 41.177.66.116'
>> P-NGCP-Src-Port: 22106'
>> P-NGCP-Src-Proto: udp'
>> P-NGCP-Src-Af: 4'
>> '
>> v=0'
>> o=Z 0 3 IN IP4 192.168.10.88'
>> s=Z'
>> c=IN IP4 41.177.66.116'
>> t=0 0'
>> m=audio 8000 RTP/AVP 8 101'
>> a=rtpmap:8 PCMA/8000'
>> a=rtpmap:101 telephone-event/8000'
>> a=fmtp:101 0-15'
>> a=inactive'
>> a=direction:active'
>> a=oldmediaip:192.168.10.88'
>>
>> #
>> U 2013/03/06 11:36:22.236968 127.0.0.1:5080 -> 127.0.0.1:5060
>>  ACK sip:user400 at 192.168.10.88:5060;alias=41.177.66.116~22106~1;rinstance=7e064df24c43ccf0;transport=UDP
>> SIP/2.0'
>> Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bKTbHrca0H;rport'
>> From: <sip:2721403 at domain.domain.com
>> >;tag=304E3AA0-51370E09000C6324-2EE19700'
>> To: <sip:user400 at domain.domain.com>;tag=d37f6717'
>> CSeq: 11 ACK'
>> Call-ID: 51370e076928-0af4wphlmirc_b2b-1'
>> Contact: <sip:127.0.0.1:5080>'
>> Route:
>> <sip:127.0.0.1;r2=on;lr=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>,
>> <sip:196.41.123.113;lr;r2=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
>> Max-Forwards: 68'
>> Content-Length: 0'
>> '
>>
>> ##
>> U 2013/03/06 11:36:22.237048 196.41.123.113:5060 -> 41.177.66.116:22106
>> ACK sip:user400 at 192.168.10.88:5060;rinstance=7e064df24c43ccf0;transport=UDP
>> SIP/2.0'
>> Record-Route:
>> <sip:196.41.123.113;r2=on;lr=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
>> Record-Route:
>> <sip:127.0.0.1;r2=on;lr=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
>> Via: SIP/2.0/UDP
>> 196.41.123.113;branch=z9hG4bKaa2.04ec964b43561d47cbdd4d22812c93a8.0'
>> Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bKTbHrca0H;rport=5080'
>> From: <sip:2721403 at domain.domain.com
>> >;tag=304E3AA0-51370E09000C6324-2EE19700'
>> To: <sip:user400 at domain.domain.com>;tag=d37f6717'
>> CSeq: 11 ACK'
>> Call-ID: 51370e076928-0af4wphlmirc_b2b-1'
>> Max-Forwards: 67'
>> Content-Length: 0'
>> Contact: <sip:ngcp-lb at 196.41.123.113:5060;ngcpct='sip:127.0.0.1:5080'>'
>>
>>
>> On Wed, Mar 6, 2013 at 9:41 AM, Theo <axessofficetheo at gmail.com> wrote:
>>
>>> Hi
>>>
>>> ngrep-sip gives nothing - Here's the process on a SNOM phone.
>>>
>>> Call comes in. You put it on hold. You dial another extension and
>>> announce the call. All that shows up with ngrep-sip. Once the call is
>>> announced you press the transfer button. Nothing is recorded for that with
>>> ngrep sip whatsoever, which is obviously why it doesn't work :-(.
>>>
>>> Any ideas on that one?
>>>
>>>
>>> On Tue, Mar 5, 2013 at 8:59 AM, Jon Bonilla <jbonilla at sipwise.com>wrote:
>>>
>>>> El Mon, 4 Mar 2013 08:59:03 +0200
>>>> Theo <axessofficetheo at gmail.com> escribi?:
>>>>
>>>> > Hi
>>>> >
>>>> > Not sure if I am breaking protocol by bumping - if I did - apologies.
>>>> I am
>>>> > almost ready to put this in as our SBC upper registration solution
>>>> (more to
>>>> > follow after that I hope :-) - but need to resolve the attended
>>>> transfer
>>>> > problem. Any ideas on this? Sip Header filtering is off altogether
>>>> (does
>>>> > that have any caveats?). Transfer working fine now for all phones
>>>> tested,
>>>> > but attended transfer not.
>>>> >
>>>>
>>>> ngrep-sip?
>>>>
>>>>
>>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130306/c9b4b465/attachment-0001.html>

From matthew at tenacit.net  Thu Mar  7 10:04:56 2013
From: matthew at tenacit.net (Matthew Ogden)
Date: Thu, 7 Mar 2013 17:04:56 +0200
Subject: [Spce-user] peering rules priority
In-Reply-To: <20121106141253.1b5b604c@multivac>
References: <354a10abf6bdbe7436c2943328ed1441@mail.gmail.com>
	<20121106141253.1b5b604c@multivac>
Message-ID: <CAB4Z0mKtmJZ6dvwOUGKqp_1OUHL_etOg4mXgWP0zhjJmoVJA0Q@mail.gmail.com>

Now,

How to make one get twice as much traffic as the other for a set numbers of
same prefix length?
Just add another group with the same peer IP and prefixes?



On Tue, Nov 6, 2012 at 3:12 PM, Jon Bonilla <jbonilla at sipwise.com> wrote:

> El Tue, 6 Nov 2012 12:13:49 +0200
> Matthew Ogden <matthew at tenacit.net> escribi?:
>
> > How would I achieve this:
> >
> >
> >
> > I have 3 peers, (currently as 3 separate peer groups)
> >
> >
> >
> > For certain number prefixes, I would like equal load balancing if
> possible.
> >
> >
> >
> > For everything else, I want them in a specific order of priority.
> >
>
> The gw selection priority is like this:
>
> - First match length
> - Second group priority
> - Third gw weight
>
> So for balancing between you 3 groupd you could have same match length,
> priority and weight in all three nodes.
>
> For others where you want strict priority you can add the preffixes with
> different match lenght if possible. An idea would be to use callee pattern.
> something like this (example spain):
>
> ^34
> ^34[0-9]
> ^34[0-9][0-9]
>
> There's another option if this is not possible but requires kamailio's lcr
> module patching. I've just tested it in my last week's training in Madrid
> and
> maybe could fit:
>
> - Make gws not unique between peer groups
> - Make gw weight behave as priority
> - Make group priority third option, after gw weight
>
> I could provide you with the patch if the default solution does not fit
> your
> needs.
>
>
> cheers,
>
> Jon
>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130307/4913bbaa/attachment-0001.html>

From john.murray at skyracktelecom.com  Thu Mar  7 17:00:44 2013
From: john.murray at skyracktelecom.com (John Murray)
Date: Thu, 7 Mar 2013 22:00:44 -0000
Subject: [Spce-user] inbound PSTN peer routing to local sip uri
In-Reply-To: <51308CD2.7040802@sipwise.com>
References: <013501ce15b3$90327c00$b0977400$@murray@skyracktelecom.com>
	<20130301103021.4a56afaf@quenya>
	<51307a6a.8665b40a.4f88.ffffa26e@mx.google.com>
	<51308CD2.7040802@sipwise.com>
Message-ID: <51390e0d.0a08b40a.33f2.ffff98a0@mx.google.com>

Andreas hi,

That worked great. I actually did:
	if ($si == "1.2.1.2"){
		$rd = $td;
        }

Which is perfect. Thanks very much for that answer.

Regards

John



-----Original Message-----
From: spce-user-bounces at lists.sipwise.com
[mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of Andreas Granig
Sent: 01 March 2013 11:11
To: spce-user at lists.sipwise.com
Subject: Re: [Spce-user] inbound PSTN peer routing to local sip uri

Hi,
On 03/01/2013 10:52 AM, John Murray wrote:
> SIP URI: a001002 at skyrack.net

Well, the problem here is that the call comes in as
sip:a001002 at 100.100.100.182, but the subscriber is sip:a001002 at skyrack.net.

There is currently no way to also change the domain part via rewrite rules,
so you'd have to do that hardcoded in the kamailio-lb config, like this:

# after the whole loose-route block:
if($rd == "100.100.100.182") {
   $rd = "skyrack.net";
}

You can use it more elegantly of course by using the template values of
network.yml, but that's basically the idea for now.

Andreas

_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com
http://lists.sipwise.com/listinfo/spce-user




From john.murray at skyracktelecom.com  Thu Mar  7 17:09:44 2013
From: john.murray at skyracktelecom.com (John Murray)
Date: Thu, 7 Mar 2013 22:09:44 -0000
Subject: [Spce-user] Send peer inbound call straight to voicemail
Message-ID: <021301ce1b80$79b93db0$6d2bb910$@murray@skyracktelecom.com>

Hi,

 

I want to send a specific call type to voicemail.

 

I send a call out to a PBX peer who has no voicemail. If the handset is not
answered I get a call from the peer with RURI 500 which is the voicemail
number, From is the original caller and a Divert: header with the subscriber
ddi.

 

I know how to take off the Divert header and get the subscriber ddi but what
would I put in the RURI to send straight to the subscribers voicemail?

I've tried prefixing 'vmu' to the ddi but that didn't work.

 

Thanks for your help.

 

Regards

 

John

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130307/db122a52/attachment-0001.html>

From jbonilla at sipwise.com  Thu Mar  7 19:22:02 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Fri, 8 Mar 2013 01:22:02 +0100
Subject: [Spce-user] peering rules priority
In-Reply-To: <CAB4Z0mKtmJZ6dvwOUGKqp_1OUHL_etOg4mXgWP0zhjJmoVJA0Q@mail.gmail.com>
References: <354a10abf6bdbe7436c2943328ed1441@mail.gmail.com>
	<20121106141253.1b5b604c@multivac>
	<CAB4Z0mKtmJZ6dvwOUGKqp_1OUHL_etOg4mXgWP0zhjJmoVJA0Q@mail.gmail.com>
Message-ID: <20130308012202.5b79c012@multivac>

El Thu, 7 Mar 2013 17:04:56 +0200
Matthew Ogden <matthew at tenacit.net> escribi?:

> Now,
> 
> How to make one get twice as much traffic as the other for a set numbers of
> same prefix length?
> Just add another group with the same peer IP and prefixes?
> 
> 

At the moment, due to the billing system, a gateway (same ip) can't be created
in two different peer groups.







From skchopperguy at gmail.com  Sat Mar  9 04:10:26 2013
From: skchopperguy at gmail.com (Skyler)
Date: Sat, 9 Mar 2013 01:10:26 -0800
Subject: [Spce-user] SMS
In-Reply-To: <51316C6E.2060201@gmail.com>
References: <EBB6CE73-9366-45FD-8E9E-8585458FA2B9@gmail.com>
	<20130301224615.1255f743@quenya>
	<CAG-pSShJ3mf8LdKKfwYjTj3=NyZ0gm_NkOBBBvmSin08MEYFrQ@mail.gmail.com>
	<513158C2.1000900@gmail.com>
	<9EF84F26-E19D-42E7-B77A-0FE3EB44BB21@gmail.com>
	<51316C6E.2060201@gmail.com>
Message-ID: <CAOUeCaQJA4bPkNfdH3k6EP5yxfahqPgAue929T3-tL0We9QQQg@mail.gmail.com>

Just for closure, the final response:yes SPCE does support SMS in this case.

On Fri, Mar 1, 2013 at 7:05 PM, Skyler <skchopperguy at gmail.com> wrote:

> That's very similar to what I was about to start work on. IM between
> multiple SPCE boxes. What the link describes is basically an IM being
> passed along to another system if not found locally. Can definitely be done
> but will take more thought into the handling and testing though.
>
> Skyler
>
>
> On 3/1/2013 5:50 PM, Jeremie Chism wrote:
>
>> Here is a link to the wiki our carrier has
>>
>> http://wiki.voipinnovations.**com/SMS.ashx<http://wiki.voipinnovations.com/SMS.ashx>
>>
>> Sent from my iPhone
>>
>> On Mar 1, 2013, at 7:41 PM, Skyler <skchopperguy at gmail.com
>> <mailto:skchopperguy at gmail.com**>> wrote:
>>
>>  In theory, if your SMSC can translate MESSAGE it should not be too
>>> hard to treat your SMSC in the same way peers are handled. ie: lookup
>>> local, if not found then go to smsc gateway. Mind you I'm sure it
>>> would be more involved than that.
>>>
>>> what are you using as an smsc? I'd be into tinkering for fun to see if
>>> a stable solution could come of it.
>>>
>>> Skyler
>>>
>>> On 3/1/2013 3:11 PM, Jeremie Chism wrote:
>>>
>>>> Yes it is sent by sip message.
>>>>
>>>> On Friday, March 1, 2013, Jon Bonilla wrote:
>>>>
>>>>    El Fri, 1 Mar 2013 15:37:41 -0600
>>>>    Jeremie Chism <jchism2 at gmail.com <mailto:jchism2 at gmail.com>
>>>>
>>>> <javascript:;>> escribi?:
>>>>
>>>>     > Does sipwise support SMS and if so is there something that needs
>>>>    to be
>>>>     > changed in the config to make it work.
>>>>
>>>>
>>>>    Could you tell us what exactly you mean by supporting sms? I guess
>>>>    the answer
>>>>    is no, but sending an http request on events to a sms gateway
>>>>    shouldn't be too
>>>>    difficult.
>>>>
>>>>
>>>>
>>>> --
>>>> Jeremie Chism
>>>> Triton Communications
>>>>
>>>>
>>>> ______________________________**_________________
>>>> Spce-user mailing list
>>>> Spce-user at lists.sipwise.com <mailto:Spce-user at lists.**sipwise.com<Spce-user at lists.sipwise.com>
>>>> >
>>>> http://lists.sipwise.com/**listinfo/spce-user<http://lists.sipwise.com/listinfo/spce-user>
>>>>
>>>>
>>> ______________________________**_________________
>>> Spce-user mailing list
>>> Spce-user at lists.sipwise.com <mailto:Spce-user at lists.**sipwise.com<Spce-user at lists.sipwise.com>
>>> >
>>> http://lists.sipwise.com/**listinfo/spce-user<http://lists.sipwise.com/listinfo/spce-user>
>>>
>>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130309/8a26345d/attachment-0001.html>

From jchism2 at gmail.com  Sun Mar 10 20:21:00 2013
From: jchism2 at gmail.com (Jeremie Chism)
Date: Sun, 10 Mar 2013 19:21:00 -0500
Subject: [Spce-user] Voicemail box limit
Message-ID: <9E6CB8F5-123E-408B-A1BD-975096191BD6@gmail.com>

Is there a limit to the number of voicemails or time limit for voicemail box. A customer notified me that their mailbox was full. If there is a limit can it be changed. 

Thanks

Sent from my iPhone


From jirka.jirout at mac.com  Mon Mar 11 04:31:23 2013
From: jirka.jirout at mac.com (Jirka Jirout)
Date: Mon, 11 Mar 2013 09:31:23 +0100
Subject: [Spce-user] Fwd: TLS problem
References: <DBDE31C2-0278-4CBC-920B-4DA2B18D54EC@me.com>
Message-ID: <3182F19C-DA78-4627-B891-4E430230C651@mac.com>

Hello,

I have now spent almost a week trying to figure out some very weird TLS behavior. I am putting together a new client app and I am experiencing strange things here. The new app can connect in about 10 % of all cases. Since SPCE is still working fine with the existing ones, I naturally lamed the new application any me code. 

HOWEVER:
Today I decided to start from scratch and wrote a simple commmand line application that does only three things: opens a TLS connection, writes the REGISTER SIP message to the socket and prints the response (which should be something like 407 Unauthorized or something, but that would be fine).

The first two steps are fine - ssl_connect() and ssl_write() return success. But then the strange thins start to happen:

- No answer ever comes back
- When I run ngrep-sip on the server, I do not see the message anywhere, although tcpdump shows it on the external interface

I first though this might be a certificate problem or something, but running my app against the spce administration interface on port 1433 returns the data just fine, although the interface uses the same SSL certificates.

The weirdest thing of course is that in about 10 % of cases, my new client connects just fine and works for hours without any problem. And that the older clients work fine.

my kamailio.cfg for lb looks like this in the tls section:

#!ifdef ENABLE_TLS
loadmodule "tls.so"
#!endif
....
#!ifdef ENABLE_TLS
modparam("tls", "certificate", "/crypto-keys/certificate.pem")
modparam("tls", "private_key", "/crypto-keys/private.key")
modparam("tls", "tls_method", "SSLv23")
#!endif

SPCE is v 2.7, openssl 0.9.8.o

Any help would be greatly appreciated as this is slowly starting to drive me nuts ;-)

regards, jj




From raul.alonso at tpartner.net  Mon Mar 11 04:57:42 2013
From: raul.alonso at tpartner.net (Raul Alonso)
Date: Mon, 11 Mar 2013 09:57:42 +0100
Subject: [Spce-user] Voicemail box limit
In-Reply-To: <9E6CB8F5-123E-408B-A1BD-975096191BD6@gmail.com>
References: <9E6CB8F5-123E-408B-A1BD-975096191BD6@gmail.com>
Message-ID: <006d01ce1e36$7ea36db0$7bea4910$@tpartner.net>

Hi,

You can change these settings in the config.yml.

    max_msg_length: 180
    maxgreet: 60
    maxmsg: 30
    min_msg_length: 3

Regards,
Raul.

-----Mensaje original-----
De: spce-user-bounces at lists.sipwise.com
[mailto:spce-user-bounces at lists.sipwise.com] En nombre de Jeremie Chism
Enviado el: lunes, 11 de marzo de 2013 1:21
Para: spce-user at lists.sipwise.com
Asunto: [Spce-user] Voicemail box limit

Is there a limit to the number of voicemails or time limit for voicemail
box. A customer notified me that their mailbox was full. If there is a limit
can it be changed. 

Thanks

Sent from my iPhone
_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com
http://lists.sipwise.com/listinfo/spce-user




From jirka.jirout at me.com  Sat Mar  9 16:45:02 2013
From: jirka.jirout at me.com (Jirka Jirout)
Date: Sat, 09 Mar 2013 22:45:02 +0100
Subject: [Spce-user] TLS problem
Message-ID: <DBDE31C2-0278-4CBC-920B-4DA2B18D54EC@me.com>

Hello,

I have now spent almost a week trying to figure out some very weird TLS behavior. I am putting together a new client app and I am experiencing strange things here. The new app can connect in about 10 % of all cases. Since SPCE is still working fine with the existing ones, I naturally lamed the new application any me code. 

HOWEVER:
 Today I decided to start from scratch and wrote a simple commmand line application that does only three things: opens a TLS connection, writes the REGISTER SIP message to the socket and prints the response (which should be something like 407 Unauthorized or something, but that would be fine).

The first two steps are fine - ssl_connect() and ssl_write() return success. But then the strange thins start to happen:

- No answer ever comes back
- When I run ngrep-sip on the server, I do not see the message anywhere, although tcpdump shows it on the external interface

I first though this might be a certificate problem or something, but running my app against the spce administration interface on port 1433 returns the data just fine, although the interface uses the same SSL certificates.

Next I tried openssl s_client -connect server:port - The connection is established fine. When I type some nonsense, i. e. GET\r\n\r\n, I end up with "read:errno=0" and a closed socket. But when I try to send the message (copy & paste), I end up with " RENEGOTIATING 11504:error:1409E0E5:SSL routines:SSL3_WRITE_BYTES:ssl handshake failure:s3_pkt.c:545:" which is the same error that my new client is getting. To be sure, I tried from several computers with openssl 0.9.8o (also used on the server) and 1.0.1c and got the same results.

The weirdest thing of course is that in about 10 % of cases, my new client connects just fine and works for hours without any problem. And that the older clients work fine.

my kamailio.cfg for lb looks like this in the tls section:

#!ifdef ENABLE_TLS
loadmodule "tls.so"
#!endif
....
#!ifdef ENABLE_TLS
modparam("tls", "certificate", "/crypto-keys/certificate.pem")
modparam("tls", "private_key", "/crypto-keys/private.key")
modparam("tls", "tls_method", "SSLv23")
#!endif

SPCE is v 2.7, openssl 0.9.8.o

Any help would be greatly appreciated as this is slowly starting to drive me nuts ;-)

regards, jj


From jbonilla at sipwise.com  Mon Mar 11 07:29:59 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Mon, 11 Mar 2013 12:29:59 +0100
Subject: [Spce-user] RTCP, Media Monitoring, Media Stats, MOS, etc..
In-Reply-To: <CAMXUi2TBTRXLeYm=x=rUOSdBVzvS2dbWDyymsiLrCWi6RbLcAA@mail.gmail.com>
References: <CAMXUi2TBTRXLeYm=x=rUOSdBVzvS2dbWDyymsiLrCWi6RbLcAA@mail.gmail.com>
Message-ID: <20130311122959.2e5f00e7@multivac>

El Fri, 1 Mar 2013 09:34:31 -0700
Gary Nieboer <gary at netdiverse.com> escribi?:

> This is a request for an update on media monitoring / performance stats,
> RTCP, etc...
> 
> Although it would be great to have a reporting engine to capture
> information on each call and provide summary stats and details per call, it
> will still be good if SPCE stores jitter packet loss, etc.. for each leg of
> a call that can be found in a log somewhere. If there is a way to see real
> time stats on multiple legs like asterisk >sip show channelstats, that
> would be great too.
> 
> What is available with SPCE?
> 
> Thanks,
> 

At this moment, all the kamailio-mediaproxy-ng protocol is being rewritten to
provide the next generation of our media proxy relay server. It will be able to
provide/gather more information than the current release so providing acurate
and more complete stats of the media traffic will be possible in the future.
When? I don't know as the first version will be released in 2.8 and we'll
continue working on the new branch for future versions.

You can keep a eye on it following our github account. Patches and suggestions
are also welcomed!

https://github.com/sipwise/


cheers,

Jon



From jbonilla at sipwise.com  Mon Mar 11 07:37:02 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Mon, 11 Mar 2013 12:37:02 +0100
Subject: [Spce-user] Fwd: TLS problem
In-Reply-To: <3182F19C-DA78-4627-B891-4E430230C651@mac.com>
References: <DBDE31C2-0278-4CBC-920B-4DA2B18D54EC@me.com>
	<3182F19C-DA78-4627-B891-4E430230C651@mac.com>
Message-ID: <20130311123702.213c7d7b@multivac>

El Mon, 11 Mar 2013 09:31:23 +0100
Jirka Jirout <jirka.jirout at mac.com> escribi?:

> 
> HOWEVER:
> Today I decided to start from scratch and wrote a simple commmand line
> application that does only three things: opens a TLS connection, writes the
> REGISTER SIP message to the socket and prints the response (which should be
> something like 407 Unauthorized or something, but that would be fine).
> 
> The first two steps are fine - ssl_connect() and ssl_write() return success.
> But then the strange thins start to happen:
> 
> - No answer ever comes back
> - When I run ngrep-sip on the server, I do not see the message anywhere,
> although tcpdump shows it on the external interface

are you listening tcp port 5061?
> 
> The weirdest thing of course is that in about 10 % of cases, my new client
> connects just fine and works for hours without any problem. And that the
> older clients work fine.
> 
> my kamailio.cfg for lb looks like this in the tls section:
> 
> #!ifdef ENABLE_TLS
> loadmodule "tls.so"
> #!endif
> ....
> #!ifdef ENABLE_TLS
> modparam("tls", "certificate", "/crypto-keys/certificate.pem")
> modparam("tls", "private_key", "/crypto-keys/private.key")
> modparam("tls", "tls_method", "SSLv23")
> #!endif
> 
> SPCE is v 2.7, openssl 0.9.8.o
> 
> Any help would be greatly appreciated as this is slowly starting to drive me
> nuts ;-)
> 

I'll try to take a look when I have some more time but meanwhile I'd suggest
you to crosspost this to the kamailio-users mailing list as this issue could
match there and you might also get an answer there.




From jbonilla at sipwise.com  Mon Mar 11 07:43:34 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Mon, 11 Mar 2013 12:43:34 +0100
Subject: [Spce-user] Upper Registration to Asterisk - transfers not
 working
In-Reply-To: <CA+9+E=V-nzXP5CeQMCBKYkyp-oJ3xKiVmKgj5D9UpBoh_363pg@mail.gmail.com>
References: <CA+9+E=X_qiRReX=c_XubTD89HEV2HApxFw1yRDgDa2nW0o_SZg@mail.gmail.com>
	<CA+9+E=U215fg8=y_ff46U2pyUxB4imXi1-i1Sx3bMbtECZ+28A@mail.gmail.com>
	<51224FFD.4020408@sipwise.com> <51225420.5000607@sipwise.com>
	<CA+9+E=U-t1paf0KW+H-vHkOwGFBkbn1=8NFd5YUPdEwm_xyZsg@mail.gmail.com>
	<5122598B.1060703@sipwise.com>
	<CA+9+E=XeYZRHuBQkhsHBTMCarq2bOdX9c+93suqyCCb-Gj+3Ag@mail.gmail.com>
	<20130226152313.2d977e1d@quenya>
	<CA+9+E=WQUrn1HbeekF2fchdZ+7kX54Snphe0QY6W_x4oXN6yyA@mail.gmail.com>
	<CA+9+E=XRB-hPxK1ZTBU5ebhnaq-y0wS0=B=8ZaeVCheBC6+tWg@mail.gmail.com>
	<CA+9+E=WU=Sa9NTZ=HJrzLXZ+VnTON_--+9fBaHtivs+Lfh7Vwg@mail.gmail.com>
	<20130305075953.50b44f18@multivac>
	<CA+9+E=UKknSzZGHFp5qi2cLx=NdaZk879jQPQi_BvEY7ce-jAg@mail.gmail.com>
	<CA+9+E=UfxgtRPVjC_8BhcEm0kh2HSKnErhro7WrFJ-eLu8SzHA@mail.gmail.com>
	<CA+9+E=X3bfBdtmTSy=TycC=veshUU8Wgb8H6HCyEzjvW1RpSJg@mail.gmail.com>
	<CA+9+E=V-nzXP5CeQMCBKYkyp-oJ3xKiVmKgj5D9UpBoh_363pg@mail.gmail.com>
Message-ID: <20130311124334.7c2c0c06@multivac>

El Wed, 6 Mar 2013 18:28:59 +0200
Theo <axessofficetheo at gmail.com> escribi?:

> Tried with some other phones and it worked, so it seems specific to the
> SNOM phones. Of course we have over a thousand SNOM phones out there so for
> us to start using this we'd have to find a fix. My guess is that SNOM needs
> to fix it, but don't have the required ammo nor knowledge to take them to
> task on it (unless I am completely wrong to start off with). Aside from
> that, I doubt whether they would fix that in a matter of days and I had
> hoped to start implementing this as a solution.
> 
> Anything that can be done on the scpe to help with this? Any words of
> wisdom I can share with SNOM?
> 


Sorry Theo I'm away form office and can't follow all threads. Please let me
know:

- Does the REFER sent from the snom phone to the spce and the REFER sent from
  the SPCE to the Asterisk look the same? Any headers missing?

- Is the REFER working fine in case you directly register the snom against the
  asterisk box? 

- If so could you capture both traces and put them attached (not inline as the
  mail clients break the lines)




From jbonilla at sipwise.com  Mon Mar 11 07:50:29 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Mon, 11 Mar 2013 12:50:29 +0100
Subject: [Spce-user] SPCE and NAT
In-Reply-To: <20130225101944.74670dc3@quenya>
References: <0A699BDB-8395-454D-8FFA-81A20B8B5C13@optionsdsl.ca>
	<511B9CA5.5010406@sipwise.com>
	<CAOUeCaSCXbR3fJo8nyy9nORdOtuB_7QcuqajyVKw3mn+mQJ+kQ@mail.gmail.com>
	<20130220231132.588a2f58@quenya>
	<CAOUeCaSS=Dqxk-Du0h_NXwKTYvvHwb9=m9Pov9zieiZYbJOSNw@mail.gmail.com>
	<20130225101944.74670dc3@quenya>
Message-ID: <20130311125029.70f1ca36@multivac>

El Mon, 25 Feb 2013 10:19:44 +0100
Jon Bonilla (Manwe) <jbonilla at sipwise.com> escribi?:


> We don't have NOTIFY keepalive support in the released versions. OPTIONS
> keepalive support isn't the best neither. I'd say that we could backport this
> to released versions and I'll discuss it internally. But what do you (our
> beloved community) say about this?
> 


Just as information, this patch was released for versions 2.5, 2.6 and 2.7.

Aditionally some small fixes were also backported. I'd suggest running the apt
+ ngcpcfg combo.

Those of you running 2.4 or lower versions of the spce should consider
upgrading to 2.7 (or 2.6 at least) as those older versions won't be
mantained much longer.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130311/6566afe3/attachment-0001.asc>

From jirka.jirout at mac.com  Mon Mar 11 12:27:03 2013
From: jirka.jirout at mac.com (Jirka Jirout)
Date: Mon, 11 Mar 2013 17:27:03 +0100
Subject: [Spce-user] TLS problem
Message-ID: <FB145375-D2D0-4DAC-AF84-090544413297@mac.com>

Hello,


>>  - No answer ever comes back
>>  - When I run ngrep-sip on the server, I do not see the message anywhere,
>>  although tcpdump shows it on the external interface
>> 
> 

> are you listening tcp port 5061?
I am not, we use a different port. But I do try to connect to the port on which TLS is enabled.

> I'll try to take a look when I have some more time but meanwhile I'd suggest
> you to crosspost this to the kamailio-users mailing list as this issue could
> match there and you might also get an answer there.
Thanks for suggestion, I will do that. If I do not manage to fix this soon, I will open a support ticket for this with you as well.

regards, jj


From mario.contreras at innovasur.es  Mon Mar 11 14:39:21 2013
From: mario.contreras at innovasur.es (Mario Contreras)
Date: Mon, 11 Mar 2013 19:39:21 +0100
Subject: [Spce-user] Avoid 302 response
Message-ID: <513E24D9.1060400@innovasur.es>

Hi all,

One of my subscribers has suffered an attack, and some calls have been 
accepted by sipwise because the subscriber sent an 302 message.

These are the messages:
SIP PEERING            SPCE


SPCE           SUBSCRIBER    SIP PEERING


Those are the two conversations, one between my sip peering and 
spce(it's an incoming call), and the other from spce to subscriber, but 
it redirects the call to another number with a 302.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130311/df4cfb9b/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: heabffjf.png
Type: image/png
Size: 3666 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130311/df4cfb9b/attachment-0002.png>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: cdgfdfae.png
Type: image/png
Size: 6168 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130311/df4cfb9b/attachment-0003.png>

From mario.contreras at innovasur.es  Mon Mar 11 14:45:53 2013
From: mario.contreras at innovasur.es (Mario Contreras)
Date: Mon, 11 Mar 2013 19:45:53 +0100
Subject: [Spce-user] Avoid 302 response
In-Reply-To: <513E24D9.1060400@innovasur.es>
References: <513E24D9.1060400@innovasur.es>
Message-ID: <513E2661.7080804@innovasur.es>

Sorry, I didn't want to send... I hope you can help me. I've been 
looking for in kamailio proxy files, but I haven't find anything 
helpful(I imagine it's there but I don't know where...).
Thanks for all!

El 11/03/2013 19:39, Mario Contreras escribi?:
> Hi all,
>
> One of my subscribers has suffered an attack, and some calls have been 
> accepted by sipwise because the subscriber sent an 302 message.
>
> These are the messages:
> SIP PEERING            SPCE
>
>
> SPCE           SUBSCRIBER    SIP PEERING
>
>
> Those are the two conversations, one between my sip peering and 
> spce(it's an incoming call), and the other from spce to subscriber, 
> but it redirects the call to another number with a 302.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130311/4d5c3923/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: image/png
Size: 3666 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130311/4d5c3923/attachment-0002.png>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: image/png
Size: 6168 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130311/4d5c3923/attachment-0003.png>

From mario.contreras at innovasur.es  Tue Mar 12 05:48:46 2013
From: mario.contreras at innovasur.es (Mario Contreras)
Date: Tue, 12 Mar 2013 10:48:46 +0100
Subject: [Spce-user] Avoid 302 response
In-Reply-To: <513E2F5B.4090707@sipwise.com>
References: <513E24D9.1060400@innovasur.es> <513E2661.7080804@innovasur.es>
	<513E2F5B.4090707@sipwise.com>
Message-ID: <513EF9FE.4060908@innovasur.es>

Thanks for your answer.

This is what I've done:

if(t_check_status("301|302"))
{
     xlog("L_ERROR", "Failed to fetch contact '$ct' from 301/302 - [% 
logreq -%]\n");
     acc_db_request("480", "acc");
     $var(announce_handle) = "callee_tmp_unavailable";
     $var(announce_set) = $avp(s:callee_sound_set);
     $(avp(s:announce_code)[*]) = 403;
     $(avp(s:announce_reason)[*]) = "Forbidden";
     route(ROUTE_EARLY_REJECT);
}

I think it will work. You think I can improve it?

Anyway, my spce treats the calls like they were incoming. Is that 
normal? Here I show an entry from cdr table:

*************************** 1. row ***************************
                                 id: 285697
                        update_time: 2013-03-10 05:09:27
                     source_user_id: 0
                 source_provider_id: 3
      source_external_subscriber_id:
        source_external_contract_id:
                  source_account_id: 0
                        source_user: myuser
                      source_domain: example.com
                         source_cli: CCCCCCCCCC
                        source_clir: 1
                destination_user_id: 0
            destination_provider_id: 3
destination_external_subscriber_id:
   destination_external_contract_id:
             destination_account_id: 0
                   destination_user: XXXXXXXXXX
                 destination_domain: 1.1.1.1
            destination_user_dialed: XXXXXXXXXX
                destination_user_in: XXXXXXXXXX
              destination_domain_in: example.com
                     peer_auth_user:
                    peer_auth_realm:
                          call_type: cfb
                        call_status: ok
                          call_code: 200
                          init_time: 1362888551.234
                         start_time: 1362888554.040
                           duration: 4.690
                            call_id: 
kif13x12kf0axb0kf1mc41ck102e3dkm at SoftX3000
                       carrier_cost: 0.00
                      reseller_cost: 0.00
                      customer_cost: 0.00
                  carrier_free_time: 0
                 reseller_free_time: 0
                 customer_free_time: 0
             carrier_billing_fee_id: NULL
            reseller_billing_fee_id: NULL
            customer_billing_fee_id: NULL
            carrier_billing_zone_id: NULL
           reseller_billing_zone_id: NULL
           customer_billing_zone_id: NULL
                frag_carrier_onpeak: NULL
               frag_reseller_onpeak: NULL
               frag_customer_onpeak: NULL
                      is_fragmented: NULL
                           rated_at: 2013-03-10 05:09:27
                      rating_status: ok
***************************************************************

Thank you!

El 11/03/2013 20:24, Andreas Granig escribi?:
> Hi,
>
> Search for "302" in your
> /etc/ngcp-config/templates/etc/kamailio/proxy/proxy.cfg and comment
> parts you don't want to be handled as is (preferably copy it to
> proxy.cfg.customtt.tt2 first and do your changes there to not get them
> overwritten with the next update).
>
> Andreas
>
> On 03/11/2013 07:45 PM, Mario Contreras wrote:
>> Sorry, I didn't want to send... I hope you can help me. I've been
>> looking for in kamailio proxy files, but I haven't find anything
>> helpful(I imagine it's there but I don't know where...).
>> Thanks for all!
>>
>> El 11/03/2013 19:39, Mario Contreras escribi?:
>>> Hi all,
>>>
>>> One of my subscribers has suffered an attack, and some calls have been
>>> accepted by sipwise because the subscriber sent an 302 message.
>>>
>>> These are the messages:
>>> SIP PEERING            SPCE
>>>
>>>
>>> SPCE           SUBSCRIBER    SIP PEERING
>>>
>>>
>>> Those are the two conversations, one between my sip peering and
>>> spce(it's an incoming call), and the other from spce to subscriber,
>>> but it redirects the call to another number with a 302.
>>
>>
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/listinfo/spce-user
>>




From agranig at sipwise.com  Tue Mar 12 06:28:55 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Tue, 12 Mar 2013 11:28:55 +0100
Subject: [Spce-user] Avoid 302 response
In-Reply-To: <513EF9FE.4060908@innovasur.es>
References: <513E24D9.1060400@innovasur.es> <513E2661.7080804@innovasur.es>
	<513E2F5B.4090707@sipwise.com> <513EF9FE.4060908@innovasur.es>
Message-ID: <513F0367.1060504@sipwise.com>

Hi,
On 03/12/2013 10:48 AM, Mario Contreras wrote:
> if(t_check_status("301|302"))
> {
>      xlog("L_ERROR", "Failed to fetch contact '$ct' from 301/302 - [%
> logreq -%]\n");
>      acc_db_request("480", "acc");
>      $var(announce_handle) = "callee_tmp_unavailable";
>      $var(announce_set) = $avp(s:callee_sound_set);
>      $(avp(s:announce_code)[*]) = 403;
>      $(avp(s:announce_reason)[*]) = "Forbidden";
>      route(ROUTE_EARLY_REJECT);
> }

Could work fine at first sight. Best is to try it out :)

> Anyway, my spce treats the calls like they were incoming. Is that
> normal? Here I show an entry from cdr table:
>
> *************************** 1. row ***************************
>                                  id: 285697
>                         update_time: 2013-03-10 05:09:27
>                      source_user_id: 0
>                  source_provider_id: 3
>       source_external_subscriber_id:
>         source_external_contract_id:
>                   source_account_id: 0
>                         source_user: myuser
>                       source_domain: example.com
>                          source_cli: CCCCCCCCCC
>                         source_clir: 1
>                 destination_user_id: 0
>             destination_provider_id: 3
> destination_external_subscriber_id:
>    destination_external_contract_id:
>              destination_account_id: 0
>                    destination_user: XXXXXXXXXX
>                  destination_domain: 1.1.1.1
>             destination_user_dialed: XXXXXXXXXX
>                 destination_user_in: XXXXXXXXXX
>               destination_domain_in: example.com
>                      peer_auth_user:
>                     peer_auth_realm:
>                           call_type: cfb

As you can see, the SPCE treats such call deflection as call-forward 
(call_type set to "cfb" instead of "call"). This is what you usually 
want, because if your subscriber configures his device to do a 
call-forward, you want this subscriber to pay for this call-forward, 
instead of charging it to the caller, because the caller doesn't know 
about this redirect.

It's the responsibility of the owner of the end device (if it's a 
bring-your-own-device service) or the platform operator (if you manage 
the devices of your customers) to keep it secure and not allow attackers 
to set it to arbitrary numbers.

Andreas



From mario.contreras at innovasur.es  Tue Mar 12 07:22:28 2013
From: mario.contreras at innovasur.es (Mario Contreras)
Date: Tue, 12 Mar 2013 12:22:28 +0100
Subject: [Spce-user] Avoid 302 response
In-Reply-To: <513F0367.1060504@sipwise.com>
References: <513E24D9.1060400@innovasur.es> <513E2661.7080804@innovasur.es>
	<513E2F5B.4090707@sipwise.com> <513EF9FE.4060908@innovasur.es>
	<513F0367.1060504@sipwise.com>
Message-ID: <513F0FF4.2080506@innovasur.es>

El 12/03/2013 11:28, Andreas Granig escribi?:
> As you can see, the SPCE treats such call deflection as call-forward
> (call_type set to "cfb" instead of "call"). This is what you usually
> want, because if your subscriber configures his device to do a
> call-forward, you want this subscriber to pay for this call-forward,
> instead of charging it to the caller, because the caller doesn't know
> about this redirect.
That's what I mean. Shouldn't the spce charge to my user? I think it 
should, but this call has almost five seconds of duration and the 
customer cost is zero.

> It's the responsibility of the owner of the end device (if it's a
> bring-your-own-device service) or the platform operator (if you manage
> the devices of your customers) to keep it secure and not allow attackers
> to set it to arbitrary numbers.
>
> Andreas

That's right of course. Thanks for your help! :)



From agranig at sipwise.com  Tue Mar 12 07:49:46 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Tue, 12 Mar 2013 12:49:46 +0100
Subject: [Spce-user] Avoid 302 response
In-Reply-To: <513F0FF4.2080506@innovasur.es>
References: <513E24D9.1060400@innovasur.es> <513E2661.7080804@innovasur.es>
	<513E2F5B.4090707@sipwise.com> <513EF9FE.4060908@innovasur.es>
	<513F0367.1060504@sipwise.com> <513F0FF4.2080506@innovasur.es>
Message-ID: <513F165A.8020204@sipwise.com>

Hi,

On 03/12/2013 12:22 PM, Mario Contreras wrote:
> El 12/03/2013 11:28, Andreas Granig escribi?:
>> As you can see, the SPCE treats such call deflection as call-forward
>> (call_type set to "cfb" instead of "call"). This is what you usually
>> want, because if your subscriber configures his device to do a
>> call-forward, you want this subscriber to pay for this call-forward,
>> instead of charging it to the caller, because the caller doesn't know
>> about this redirect.
> That's what I mean. Shouldn't the spce charge to my user? I think it
> should, but this call has almost five seconds of duration and the
> customer cost is zero.

You're right. Which version are you running? If 2.7, can you do an 
"apt-get update; apt-get upgrade; ngcpcfg apply" (might result in a 
short service down-time due to kamailio restart)? We've applied a fix 
for 2.7 for properly setting the account_id fields on 3xx around a month 
ago.

Andreas



From axessofficetheo at gmail.com  Tue Mar 12 10:30:23 2013
From: axessofficetheo at gmail.com (Theo)
Date: Tue, 12 Mar 2013 16:30:23 +0200
Subject: [Spce-user] Upper Registration to Asterisk - transfers not
	working
In-Reply-To: <20130311124334.7c2c0c06@multivac>
References: <CA+9+E=X_qiRReX=c_XubTD89HEV2HApxFw1yRDgDa2nW0o_SZg@mail.gmail.com>
	<CA+9+E=U215fg8=y_ff46U2pyUxB4imXi1-i1Sx3bMbtECZ+28A@mail.gmail.com>
	<51224FFD.4020408@sipwise.com> <51225420.5000607@sipwise.com>
	<CA+9+E=U-t1paf0KW+H-vHkOwGFBkbn1=8NFd5YUPdEwm_xyZsg@mail.gmail.com>
	<5122598B.1060703@sipwise.com>
	<CA+9+E=XeYZRHuBQkhsHBTMCarq2bOdX9c+93suqyCCb-Gj+3Ag@mail.gmail.com>
	<20130226152313.2d977e1d@quenya>
	<CA+9+E=WQUrn1HbeekF2fchdZ+7kX54Snphe0QY6W_x4oXN6yyA@mail.gmail.com>
	<CA+9+E=XRB-hPxK1ZTBU5ebhnaq-y0wS0=B=8ZaeVCheBC6+tWg@mail.gmail.com>
	<CA+9+E=WU=Sa9NTZ=HJrzLXZ+VnTON_--+9fBaHtivs+Lfh7Vwg@mail.gmail.com>
	<20130305075953.50b44f18@multivac>
	<CA+9+E=UKknSzZGHFp5qi2cLx=NdaZk879jQPQi_BvEY7ce-jAg@mail.gmail.com>
	<CA+9+E=UfxgtRPVjC_8BhcEm0kh2HSKnErhro7WrFJ-eLu8SzHA@mail.gmail.com>
	<CA+9+E=X3bfBdtmTSy=TycC=veshUU8Wgb8H6HCyEzjvW1RpSJg@mail.gmail.com>
	<CA+9+E=V-nzXP5CeQMCBKYkyp-oJ3xKiVmKgj5D9UpBoh_363pg@mail.gmail.com>
	<20130311124334.7c2c0c06@multivac>
Message-ID: <CA+9+E=XydUFsz0V85v4kGsZpcDZXs-asTzfetqKqmtUYegGCbQ@mail.gmail.com>

Hi Jon

I am away myself but as soon as I am back I will send the info.

thanks

Theo

On Mon, Mar 11, 2013 at 1:43 PM, Jon Bonilla <jbonilla at sipwise.com> wrote:

> El Wed, 6 Mar 2013 18:28:59 +0200
> Theo <axessofficetheo at gmail.com> escribi?:
>
> > Tried with some other phones and it worked, so it seems specific to the
> > SNOM phones. Of course we have over a thousand SNOM phones out there so
> for
> > us to start using this we'd have to find a fix. My guess is that SNOM
> needs
> > to fix it, but don't have the required ammo nor knowledge to take them to
> > task on it (unless I am completely wrong to start off with). Aside from
> > that, I doubt whether they would fix that in a matter of days and I had
> > hoped to start implementing this as a solution.
> >
> > Anything that can be done on the scpe to help with this? Any words of
> > wisdom I can share with SNOM?
> >
>
>
> Sorry Theo I'm away form office and can't follow all threads. Please let me
> know:
>
> - Does the REFER sent from the snom phone to the spce and the REFER sent
> from
>   the SPCE to the Asterisk look the same? Any headers missing?
>
> - Is the REFER working fine in case you directly register the snom against
> the
>   asterisk box?
>
> - If so could you capture both traces and put them attached (not inline as
> the
>   mail clients break the lines)
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130312/1c0045ee/attachment-0001.html>

From pwilbert at gmail.com  Tue Mar 12 15:24:20 2013
From: pwilbert at gmail.com (Penn Wilbert)
Date: Tue, 12 Mar 2013 15:24:20 -0400
Subject: [Spce-user] Voicemail not answering on fresh SIP provider CE
	installation
Message-ID: <CABSTHcoXc=cAGKrndSF4P+innKYZUdEDayp5gy9-XabtccO1JQ@mail.gmail.com>

I loaded the sip provider CE VirtualBox VM on my server. I went through the
motions to add the system. I have added a few SIP devices to the system and
can make calls back and forth. However I cannot get voicemail to work. I
set one number to forward to Voicebox all the time and when I call the
number I get a fast beep tone and a eventual call failed.

I can log in to the web interfaces with the users and see the missed calls.
I cannot find the PIN for the user when they dial 2000 to get access to the
voicemail system, even though the system answers and asks for a phone and
PIN.

The asterisk server is running and the config on the server does state that
voicemail is yes.

Thank you for your time!

-- 
Penn Wilbert
pwilbert at gmail.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130312/d0aa28d0/attachment-0001.html>

From agranig at sipwise.com  Tue Mar 12 15:40:22 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Tue, 12 Mar 2013 20:40:22 +0100
Subject: [Spce-user] WSDL without documentation tags
Message-ID: <513F84A6.4010901@sipwise.com>

Hi,

We've backported a change to the SOAP interface to 2.7 which allows you 
to fetch the WSDL without the inline documentation tags.

Some clients are known to choke on them (e.g. the stock SOAP client for 
PHP, where we recommended nusoap in the past, and also Visual Studio/C# 
and probably some more), so we added an URL parameter called "plain" to 
fetch the plain WSDL without a path to the stylesheet and without the 
documentation tags.

The previous URL

	https://<ip>:2443/SOAP/Provisioning.wsdl

works as usual and renders the online documentation in your browsers and 
can be used with most SOAP clients.

If this doesn't work, try

	https://<ip>:2443/SOAP/Provisioning.wsdl?plain

for the version without documentation in your SOAP client.

Do an "apt-get update && apt-get upgrade && ngcpcfg apply" to install 
this patch.

Andreas



From john.murray at skyracktelecom.com  Wed Mar 13 07:40:02 2013
From: john.murray at skyracktelecom.com (John Murray)
Date: Wed, 13 Mar 2013 11:40:02 -0000
Subject: [Spce-user] Voicemail config bug?
Message-ID: <026701ce1fdf$801e98e0$805bcaa0$@murray@skyracktelecom.com>

Hello,

 

If a subscriber is setup with diverts to voicemail and the subscribers DDI
is subsequently changed. This does not get reflected in the voicemail
setting and calls are forwarded to vmu-OLD-DDI- at voicebox.local which is
dropped:

 

asterisk[2961]: WARNING[18132]: app_voicemail.c:3972 in leave_voicemail: No
entry in voicemail config file for '-OLD-DDI-'

 

I can fix this by deleting the voicemail destination set and re-adding it.

 

Can this be fixed?

 

Best regards

 

John

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130313/0ce82ab9/attachment-0001.html>

From dgrotti at sipwise.com  Wed Mar 13 07:45:39 2013
From: dgrotti at sipwise.com (Daniel Grotti)
Date: Wed, 13 Mar 2013 12:45:39 +0100
Subject: [Spce-user] Voicemail not answering on fresh SIP provider CE
 installation
In-Reply-To: <CABSTHcoXc=cAGKrndSF4P+innKYZUdEDayp5gy9-XabtccO1JQ@mail.gmail.com>
References: <CABSTHcoXc=cAGKrndSF4P+innKYZUdEDayp5gy9-XabtccO1JQ@mail.gmail.com>
Message-ID: <514066E3.2050602@sipwise.com>

Hi Penn,
Can you see the call arrives in Asterisk ?
Try to connect to Asterisk (asterisk -vvvvvr) and then make a call and
see what happens.

Could you also provide us with some logs ?

- /var/log/ngcp/kamailio-proxy.log


Thanks,
Daniel


On 03/12/2013 08:24 PM, Penn Wilbert wrote:
> I loaded the sip provider CE VirtualBox VM on my server. I went
> through the motions to add the system. I have added a few SIP devices
> to the system and can make calls back and forth. However I cannot get
> voicemail to work. I set one number to forward to Voicebox all the
> time and when I call the number I get a fast beep tone and a eventual
> call failed.
>
> I can log in to the web interfaces with the users and see the missed
> calls. I cannot find the PIN for the user when they dial 2000 to get
> access to the voicemail system, even though the system answers and
> asks for a phone and PIN.
>
> The asterisk server is running and the config on the server does state
> that voicemail is yes.
>
> Thank you for your time!
>
> -- 
> Penn Wilbert
> pwilbert at gmail.com <mailto:pwilbert at gmail.com>
>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130313/9520f885/attachment-0001.html>

From pwilbert at gmail.com  Wed Mar 13 07:49:25 2013
From: pwilbert at gmail.com (Penn Wilbert)
Date: Wed, 13 Mar 2013 07:49:25 -0400
Subject: [Spce-user] Voicemail not answering on fresh SIP provider CE
	installation
In-Reply-To: <514066E3.2050602@sipwise.com>
References: <CABSTHcoXc=cAGKrndSF4P+innKYZUdEDayp5gy9-XabtccO1JQ@mail.gmail.com>
	<514066E3.2050602@sipwise.com>
Message-ID: <CABSTHcrRPFmk5PUh-OHjU8PmzvzFi+DeWgRiLOZAsT_hh0AASA@mail.gmail.com>

Daniel,

I'll update in a few hours about the status of this issue. I found the
archives after I had sent the email. In the archives I found a similar
issue happening to a user using a fresh install of 2.7 CE. We had not
updated the VM yet so I'll be testing shortly as to whether the issue still
happens or not after the update. If it doesn't work I'll send logs.


On Wed, Mar 13, 2013 at 7:45 AM, Daniel Grotti <dgrotti at sipwise.com> wrote:

>  Hi Penn,
> Can you see the call arrives in Asterisk ?
> Try to connect to Asterisk (asterisk -vvvvvr) and then make a call and see
> what happens.
>
> Could you also provide us with some logs ?
>
> - /var/log/ngcp/kamailio-proxy.log
>
>
> Thanks,
> Daniel
>
>
>
> On 03/12/2013 08:24 PM, Penn Wilbert wrote:
>
>  I loaded the sip provider CE VirtualBox VM on my server. I went through
> the motions to add the system. I have added a few SIP devices to the system
> and can make calls back and forth. However I cannot get voicemail to work.
> I set one number to forward to Voicebox all the time and when I call the
> number I get a fast beep tone and a eventual call failed.
>
> I can log in to the web interfaces with the users and see the missed
> calls. I cannot find the PIN for the user when they dial 2000 to get access
> to the voicemail system, even though the system answers and asks for a
> phone and PIN.
>
>  The asterisk server is running and the config on the server does state
> that voicemail is yes.
>
>  Thank you for your time!
>
> --
> Penn Wilbert
> pwilbert at gmail.com
>
>
> _______________________________________________
> Spce-user mailing listSpce-user at lists.sipwise.comhttp://lists.sipwise.com/listinfo/spce-user
>
>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
>
>


-- 
Penn Wilbert
pwilbert at gmail.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130313/6bfe2bc5/attachment-0001.html>

From john.murray at skyracktelecom.com  Wed Mar 13 07:54:04 2013
From: john.murray at skyracktelecom.com (John Murray)
Date: Wed, 13 Mar 2013 11:54:04 -0000
Subject: [Spce-user] Domain prefs for subscribers with 'Trusted Source'
Message-ID: <027e01ce1fe1$75bf5860$613e0920$@murray@skyracktelecom.com>

Hello,

 

Subscribers making calls from devices connected via 'Trusted Source' don't
seem to load the domain preferences of the subscriber but atempt to load
domain preferences of the incomming 'From: <domain part>' header.

Is this correct?

 

Regards

 

John

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130313/aad7b607/attachment-0001.html>

From apogrebennyk at sipwise.com  Wed Mar 13 08:00:58 2013
From: apogrebennyk at sipwise.com (Andrew Pogrebennyk)
Date: Wed, 13 Mar 2013 13:00:58 +0100
Subject: [Spce-user] Voicemail config bug?
In-Reply-To: <026701ce1fdf$801e98e0$805bcaa0$@murray@skyracktelecom.com>
References: <026701ce1fdf$801e98e0$805bcaa0$@murray@skyracktelecom.com>
Message-ID: <51406A7A.2090807@sipwise.com>

Hi John,

On 03/13/2013 12:40 PM, John Murray wrote:
> If a subscriber is setup with diverts to voicemail and the subscribers
> DDI is subsequently changed. This does not get reflected in the
> voicemail setting and calls are forwarded to vmu-OLD-DDI- at voicebox.local
> <mailto:vmu-OLD-DDI- at voicebox.local> which is dropped:

This is already in our bugtracker. I will notify you when this gets fixed.

Regards,
Andrew



From john.murray at skyracktelecom.com  Wed Mar 13 08:03:49 2013
From: john.murray at skyracktelecom.com (John Murray)
Date: Wed, 13 Mar 2013 12:03:49 -0000
Subject: [Spce-user] Voicemail config bug?
In-Reply-To: <51406A7A.2090807@sipwise.com>
References: <026701ce1fdf$801e98e0$805bcaa0$@murray@skyracktelecom.com>
	<51406A7A.2090807@sipwise.com>
Message-ID: <51406b26.8f4bb40a.2fee.ffffb73d@mx.google.com>

Many thanks.

-----Original Message-----
From: Andrew Pogrebennyk [mailto:apogrebennyk at sipwise.com] 
Sent: 13 March 2013 12:01
To: John Murray
Cc: spce-user at lists.sipwise.com
Subject: Re: [Spce-user] Voicemail config bug?

Hi John,

On 03/13/2013 12:40 PM, John Murray wrote:
> If a subscriber is setup with diverts to voicemail and the subscribers 
> DDI is subsequently changed. This does not get reflected in the 
> voicemail setting and calls are forwarded to 
> vmu-OLD-DDI- at voicebox.local <mailto:vmu-OLD-DDI- at voicebox.local> which is
dropped:

This is already in our bugtracker. I will notify you when this gets fixed.

Regards,
Andrew




From csaroli at isysp.com  Wed Mar 13 11:52:10 2013
From: csaroli at isysp.com (Christopher Saroli)
Date: Wed, 13 Mar 2013 15:52:10 +0000
Subject: [Spce-user]  Billing Profiles/Fees
Message-ID: <EE7149FDD60B3045A4377F05AAFFF3F118342A46@ISS-Server.ISS.local>

Little confused on this, every call I make/receive does not get a fee applied to it.  Just trying right now to apply a fee to all calls.  I might be totally wrong here, any info would be appreciated.

This is my config:

Billing Profile:
Interval Charge:  0.02
Freetime: 	  0
Free Cash: 	  0
Monthly Lim	  0
Lock		  none
Currency: 	 $


Bililng Fees
Destination:  	.*
Zone:      	All
Init Rate:	1
Init interval:	1
Follow Rate: 	1
Follow interval: 1


Also I have no outgoing calls recorded in the CDR under the admin interface but in the customer portal they are there. Is this by design or is something wrong.

Thanks




From Stephen at newbreakcommunications.com  Wed Mar 13 12:25:05 2013
From: Stephen at newbreakcommunications.com (Stephen Donovan)
Date: Wed, 13 Mar 2013 16:25:05 +0000
Subject: [Spce-user] random registration dropping
Message-ID: <2CDACF784B10A14EAAFAB6140DAE0D132409FC@EXCHANGE1.newbreakmanagement.local>

We have a few customers that are complaining of random dropped calls and no dialtone for short periods of time.  It has not happened for a long enough period of time to even nail it down.  Are there any known issues ?  I'm not certain these are the only ones it was happening to but the customers  that called were both using Linksys SPA8000 ATAs registered directly to spce.

I can provide any logs needed from the affected period of time.

Stephen

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130313/f43eaa3a/attachment-0001.html>

From jcifre at ib-red.com  Wed Mar 13 12:36:34 2013
From: jcifre at ib-red.com (Joan Cifre (ibred))
Date: Wed, 13 Mar 2013 17:36:34 +0100
Subject: [Spce-user] random registration dropping
In-Reply-To: <2CDACF784B10A14EAAFAB6140DAE0D132409FC@EXCHANGE1.newbreakmanagement.local>
References: <2CDACF784B10A14EAAFAB6140DAE0D132409FC@EXCHANGE1.newbreakmanagement.local>
Message-ID: <5140AB12.60301@ib-red.com>

In Linksys configuration try to disable keepalives for nat.


El 2013-03-13 17:25, Stephen Donovan escribi?:
>
> We have a few customers that are complaining of random dropped calls 
> and no dialtone for short periods of time.  It has not happened for a 
> long enough period of time to even nail it down.  Are there any known 
> issues ? I'm not certain these are the only ones it was happening to 
> but the customers  that called were both using Linksys SPA8000 ATAs 
> registered directly to spce.
>
> I can provide any logs needed from the affected period of time.
>
> Stephen
>
>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130313/6ce1d8d7/attachment-0001.html>

From Stephen at newbreakcommunications.com  Wed Mar 13 13:05:45 2013
From: Stephen at newbreakcommunications.com (Stephen Donovan)
Date: Wed, 13 Mar 2013 17:05:45 +0000
Subject: [Spce-user] random registration dropping
In-Reply-To: <5140AB12.60301@ib-red.com>
References: <2CDACF784B10A14EAAFAB6140DAE0D132409FC@EXCHANGE1.newbreakmanagement.local>
	<5140AB12.60301@ib-red.com>
Message-ID: <2CDACF784B10A14EAAFAB6140DAE0D13240B62@EXCHANGE1.newbreakmanagement.local>

Just verified no NAT keepalive enabled on any line.

Stephen


From: Joan Cifre (ibred) [mailto:jcifre at ib-red.com]
Sent: Wednesday, March 13, 2013 11:37 AM
To: Stephen Donovan
Cc: 'spce-user at lists.sipwise.com'
Subject: Re: [Spce-user] random registration dropping

In Linksys configuration try to disable keepalives for nat.

El 2013-03-13 17:25, Stephen Donovan escribi?:
We have a few customers that are complaining of random dropped calls and no dialtone for short periods of time.  It has not happened for a long enough period of time to even nail it down.  Are there any known issues ?  I'm not certain these are the only ones it was happening to but the customers  that called were both using Linksys SPA8000 ATAs registered directly to spce.

I can provide any logs needed from the affected period of time.

Stephen





_______________________________________________

Spce-user mailing list

Spce-user at lists.sipwise.com<mailto:Spce-user at lists.sipwise.com>

http://lists.sipwise.com/listinfo/spce-user

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130313/6ffee1e6/attachment-0001.html>

From yyicur4me at gmail.com  Wed Mar 13 10:41:34 2013
From: yyicur4me at gmail.com (V)
Date: Wed, 13 Mar 2013 10:41:34 -0400
Subject: [Spce-user] Initial registration fails W/malformed sip
Message-ID: <CALwLLOC0sDheu-5U-Kiv3gQzeTsp2yfjCScjgmtDMGL-WMt5_Q@mail.gmail.com>

Endpoint fails to register.

My logs show the domain coming through incorrect
I suspect something is setup incorrect at server

sip user ID: ryandavis
domain: telific.com
ngcp server ip: 98.103.105.117
ngcp server DNS name ash-butler
ngcp server FQDN: ash-butler.telific.com

It seems to be appending the DNS name to the user ID but I have no idea why

Any help will be greatly appreciated.
Please advise
Thanks
Vince


Mar 13 10:28:15 ash-bulter /usr/sbin/kamailio[1606]: INFO: <script>: New
request - M=OPTIONS R=sip:nagios at voip.sipwise.local F=
sip:sipsak at 98.103.105.117:6988 T=sip:nagios at voip.sipwise.local IP=udp:
127.0.0.1:6988 ID=239285313 at 98.103.105.117
Mar 13 10:28:15 ash-bulter /usr/sbin/kamailio[1606]: INFO: <script>: NAT
check disabled by administrator for source IP '127.0.0.1' - M=OPTIONS
R=sip:nagios at voip.sipwise.local
F=sip:sipsak at 98.103.105.117:6988T=sip:nagios at voip.sipwise.localIP=
127.0.0.1:6988 ID=239285313 at 98.103.105.117
Mar 13 10:28:15 ash-bulter /usr/sbin/kamailio[1606]: INFO: <script>:
Relaying request, du='sip:127.0.0.1:5062', fs='udp:127.0.0.1:5060' -
R=sip:nagios at voip.sipwise.local ID=239285313 at 98.103.105.117
Mar 13 10:28:15 ash-bulter /usr/sbin/kamailio[1604]: INFO: <script>: Reply
from Inbound - S=100 - Trying M=OPTIONS IP=udp:127.0.0.1:5062
ID=239285313 at 98.103.105.117
Mar 13 10:28:15 ash-bulter /usr/sbin/kamailio[1604]: INFO: <script>:
Sending reply, fs='udp:127.0.0.1:5060' - ID=239285313 at 98.103.105.117
Mar 13 10:28:15 ash-bulter /usr/sbin/kamailio[1602]: INFO: <script>: Reply
from Inbound - S=200 - Alive M=OPTIONS IP=udp:127.0.0.1:5062
ID=239285313 at 98.103.105.117
Mar 13 10:28:15 ash-bulter /usr/sbin/kamailio[1602]: INFO: <script>:
Sending reply, fs='udp:127.0.0.1:5060' - ID=239285313 at 98.103.105.117
Mar 13 10:28:21 ash-bulter /usr/sbin/kamailio[1593]: WARNING: sanity
[sanity.c:800]: sanity_check(): check_parse_uris(): failed to parse From uri
Mar 13 10:28:21 ash-bulter /usr/sbin/kamailio[1593]: WARNING: <script>:
Malformed SIP message detected - M=REGISTER
R=sip:ash-butler.telific.comF=sip:ryandavis at telific.com
@ash-butler.telific.com
T=sip:ryandavis at telific.com@ash-butler.telific.comIP=udp:
173.89.168.17:5060 ID=1136074965-5060-1 at BJC.BGI.A.C
Mar 13 10:28:25 ash-bulter /usr/sbin/kamailio[1601]: INFO: <script>: New
request - M=OPTIONS R=sip:nagios at voip.sipwise.local F=
sip:sipsak at 98.103.105.117:8393 T=sip:nagios at voip.sipwise.local IP=udp:
127.0.0.1:8393 ID=1964588741 at 98.103.105.117
Mar 13 10:28:25 ash-bulter /usr/sbin/kamailio[1601]: INFO: <script>: NAT
check disabled by administrator for source IP '127.0.0.1' - M=OPTIONS
R=sip:nagios at voip.sipwise.local
F=sip:sipsak at 98.103.105.117:8393T=sip:nagios at voip.sipwise.localIP=
127.0.0.1:8393 ID=1964588741 at 98.103.105.117
Mar 13 10:28:25 ash-bulter /usr/sbin/kamailio[1601]: INFO: <script>:
Relaying request, du='sip:127.0.0.1:5062', fs='udp:127.0.0.1:5060' -
R=sip:nagios at voip.sipwise.local ID=1964588741 at 98.103.105.117
Mar 13 10:28:25 ash-bulter /usr/sbin/kamailio[1599]: INFO: <script>: Reply
from Inbound - S=100 - Trying M=OPTIONS IP=udp:127.0.0.1:5062
ID=1964588741 at 98.103.105.117
Mar 13 10:28:25 ash-bulter /usr/sbin/kamailio[1599]: INFO: <script>:
Sending reply, fs='udp:127.0.0.1:5060' - ID=1964588741 at 98.103.105.117
Mar 13 10:28:25 ash-bulter /usr/sbin/kamailio[1608]: INFO: <script>: Reply
from Inbound - S=200 - Alive M=OPTIONS IP=udp:127.0.0.1:5062
ID=1964588741 at 98.103.105.117
Mar 13 10:28:25 ash-bulter /usr/sbin/kamailio[1608]: INFO: <script>:
Sending reply, fs='udp:127.0.0.1:5060' - ID=1964588741 at 98.103.105.117
Mar 13 10:28:35 ash-bulter /usr/sbin/kamailio[1600]: INFO: <script>: New
request - M=OPTIONS R=sip:nagios at voip.sipwise.local F=
sip:sipsak at 98.103.105.117:6584 T=sip:nagios at voip.sipwise.local IP=udp:
127.0.0.1:6584 ID=1724643046 at 98.103.105.117
Mar 13 10:28:35 ash-bulter /usr/sbin/kamailio[1600]: INFO: <script>: NAT
check disabled by administrator for source IP '127.0.0.1' - M=OPTIONS
R=sip:nagios at voip.sipwise.local
F=sip:sipsak at 98.103.105.117:6584T=sip:nagios at voip.sipwise.localIP=
127.0.0.1:6584 ID=1724643046 at 98.103.105.117
Mar 13 10:28:35 ash-bulter /usr/sbin/kamailio[1600]: INFO: <script>:
Relaying request, du='sip:127.0.0.1:5062', fs='udp:127.0.0.1:5060' -
R=sip:nagios at voip.sipwise.local ID=1724643046 at 98.103.105.117
Mar 13 10:28:35 ash-bulter /usr/sbin/kamailio[1603]: INFO: <script>: Reply
from Inbound - S=100 - Trying M=OPTIONS IP=udp:127.0.0.1:5062
ID=1724643046 at 98.103.105.117
Mar 13 10:28:35 ash-bulter /usr/sbin/kamailio[1603]: INFO: <script>:
Sending reply, fs='udp:127.0.0.1:5060' - ID=1724643046 at 98.103.105.117
Mar 13 10:28:35 ash-bulter /usr/sbin/kamailio[1603]: INFO: <script>: Reply
from Inbound - S=200 - Alive M=OPTIONS IP=udp:127.0.0.1:5062
ID=1724643046 at 98.103.105.117
Mar 13 10:28:35 ash-bulter /usr/sbin/kamailio[1603]: INFO: <script>:
Sending reply, fs='udp:127.0.0.1:5060' - ID=1724643046 at 98.103.105.117


-- 

*mY iRQs aRe cOnfLicTiNg ?*
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130313/001aa2f3/attachment-0001.html>

From agranig at sipwise.com  Wed Mar 13 14:59:05 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Wed, 13 Mar 2013 19:59:05 +0100
Subject: [Spce-user] Initial registration fails W/malformed sip
In-Reply-To: <CALwLLOC0sDheu-5U-Kiv3gQzeTsp2yfjCScjgmtDMGL-WMt5_Q@mail.gmail.com>
References: <CALwLLOC0sDheu-5U-Kiv3gQzeTsp2yfjCScjgmtDMGL-WMt5_Q@mail.gmail.com>
Message-ID: <5140CC79.9080309@sipwise.com>

Hi,

On 03/13/2013 03:41 PM, V wrote:
> Endpoint fails to register.
> My logs show the domain coming through incorrect
> I suspect something is setup incorrect at server

No, it's a misconfiguration of your client. Probably you're using a 
client with a username and domain field, and you put 
ryandavis at telific.com into the username, because your client is sending 
sip:ryandavis at telific.com@ash-butler.telific.com.

In any case, if you use telific.com as domain, but the server is 
reachable only via ash-butler.telific.com, you need to use ryndavis as 
username, telific.com as domain, and ash-butler.telific.com as 
outbound-proxy/registrar.

Andreas



From john.murray at skyracktelecom.com  Wed Mar 13 16:15:49 2013
From: john.murray at skyracktelecom.com (John Murray)
Date: Wed, 13 Mar 2013 20:15:49 -0000
Subject: [Spce-user] Delete subscriber does not delete 'trusted' entry
Message-ID: <035601ce2027$8e13aab0$aa3b0010$@murray@skyracktelecom.com>

Hello,

 

In v2.7 when I subscriber is deleted the entry in the trusted table is not
deleted therefore when adding the same pattern to a new subscriber the
original (deleted) subscriber is matched.

 

Regards

 

John

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130313/a0a6d65e/attachment-0001.html>

From pwilbert at gmail.com  Wed Mar 13 16:18:54 2013
From: pwilbert at gmail.com (Penn Wilbert)
Date: Wed, 13 Mar 2013 16:18:54 -0400
Subject: [Spce-user] Voicemail not answering on fresh SIP provider CE
	installation
In-Reply-To: <CABSTHcrRPFmk5PUh-OHjU8PmzvzFi+DeWgRiLOZAsT_hh0AASA@mail.gmail.com>
References: <CABSTHcoXc=cAGKrndSF4P+innKYZUdEDayp5gy9-XabtccO1JQ@mail.gmail.com>
	<514066E3.2050602@sipwise.com>
	<CABSTHcrRPFmk5PUh-OHjU8PmzvzFi+DeWgRiLOZAsT_hh0AASA@mail.gmail.com>
Message-ID: <CABSTHcpCp_LqdJfwnjZk2k6UPZxWBGbprLDy447FGg6GjEnMEA@mail.gmail.com>

Running updates against the server resolved the issue


On Wed, Mar 13, 2013 at 7:49 AM, Penn Wilbert <pwilbert at gmail.com> wrote:

> Daniel,
>
> I'll update in a few hours about the status of this issue. I found the
> archives after I had sent the email. In the archives I found a similar
> issue happening to a user using a fresh install of 2.7 CE. We had not
> updated the VM yet so I'll be testing shortly as to whether the issue still
> happens or not after the update. If it doesn't work I'll send logs.
>
>
> On Wed, Mar 13, 2013 at 7:45 AM, Daniel Grotti <dgrotti at sipwise.com>wrote:
>
>>  Hi Penn,
>> Can you see the call arrives in Asterisk ?
>> Try to connect to Asterisk (asterisk -vvvvvr) and then make a call and
>> see what happens.
>>
>> Could you also provide us with some logs ?
>>
>> - /var/log/ngcp/kamailio-proxy.log
>>
>>
>> Thanks,
>> Daniel
>>
>>
>>
>> On 03/12/2013 08:24 PM, Penn Wilbert wrote:
>>
>>  I loaded the sip provider CE VirtualBox VM on my server. I went through
>> the motions to add the system. I have added a few SIP devices to the system
>> and can make calls back and forth. However I cannot get voicemail to work.
>> I set one number to forward to Voicebox all the time and when I call the
>> number I get a fast beep tone and a eventual call failed.
>>
>> I can log in to the web interfaces with the users and see the missed
>> calls. I cannot find the PIN for the user when they dial 2000 to get access
>> to the voicemail system, even though the system answers and asks for a
>> phone and PIN.
>>
>>  The asterisk server is running and the config on the server does state
>> that voicemail is yes.
>>
>>  Thank you for your time!
>>
>> --
>> Penn Wilbert
>> pwilbert at gmail.com
>>
>>
>> _______________________________________________
>> Spce-user mailing listSpce-user at lists.sipwise.comhttp://lists.sipwise.com/listinfo/spce-user
>>
>>
>>
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/listinfo/spce-user
>>
>>
>
>
> --
> Penn Wilbert
> pwilbert at gmail.com
>



-- 
Penn Wilbert
pwilbert at gmail.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130313/f1ca7968/attachment-0001.html>

From jbonilla at sipwise.com  Wed Mar 13 18:02:43 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Wed, 13 Mar 2013 23:02:43 +0100
Subject: [Spce-user] Domain prefs for subscribers with 'Trusted Source'
In-Reply-To: <027e01ce1fe1$75bf5860$613e0920$@murray@skyracktelecom.com>
References: <027e01ce1fe1$75bf5860$613e0920$@murray@skyracktelecom.com>
Message-ID: <20130313230243.470b266c@quenya>

El Wed, 13 Mar 2013 11:54:04 -0000
"John Murray" <john.murray at skyracktelecom.com> escribi?:

> Hello,
> 
>  
> 
> Subscribers making calls from devices connected via 'Trusted Source' don't
> seem to load the domain preferences of the subscriber but atempt to load
> domain preferences of the incomming 'From: <domain part>' header.
> 
> Is this correct?
> 

I'd say yes. When a source is trusted you don't check which domain it belongs to
as you would usually do with a "normal" subscriber. You just trust what it
sends. 

Maybe this could be reviewed. Is this a desirable behaviour? It wouldn't be
difficult to check and load the real domain.



-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130313/eb2a0393/attachment-0001.asc>

From jbonilla at sipwise.com  Wed Mar 13 18:05:20 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Wed, 13 Mar 2013 23:05:20 +0100
Subject: [Spce-user] random registration dropping
In-Reply-To: <5140AB12.60301@ib-red.com>
References: <2CDACF784B10A14EAAFAB6140DAE0D132409FC@EXCHANGE1.newbreakmanagement.local>
	<5140AB12.60301@ib-red.com>
Message-ID: <20130313230520.5be292fb@quenya>

El Wed, 13 Mar 2013 17:36:34 +0100
"Joan Cifre (ibred)" <jcifre at ib-red.com> escribi?:

> In Linksys configuration try to disable keepalives for nat.
>

Nat keepalives sent by the clients are now handled by the spce without banning
the user. Unless they send too many and the ip is banned :)


-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130313/77fb5e2a/attachment-0001.asc>

From jbonilla at sipwise.com  Wed Mar 13 18:06:57 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Wed, 13 Mar 2013 23:06:57 +0100
Subject: [Spce-user] random registration dropping
In-Reply-To: <2CDACF784B10A14EAAFAB6140DAE0D132409FC@EXCHANGE1.newbreakmanagement.local>
References: <2CDACF784B10A14EAAFAB6140DAE0D132409FC@EXCHANGE1.newbreakmanagement.local>
Message-ID: <20130313230657.7eb033ac@quenya>

El Wed, 13 Mar 2013 16:25:05 +0000
Stephen Donovan <Stephen at newbreakcommunications.com> escribi?:

> We have a few customers that are complaining of random dropped calls and no
> dialtone for short periods of time.  It has not happened for a long enough
> period of time to even nail it down.  Are there any known issues ?  I'm not
> certain these are the only ones it was happening to but the customers  that
> called were both using Linksys SPA8000 ATAs registered directly to spce.
> 
> I can provide any logs needed from the affected period of time.
> 
> Stephen
> 

no known issue that I know. We would need to see a log matching such a call.
Kamailio-proxy as starting point as usual and then we could check how/why/where
the call was dropped.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130313/c179e508/attachment-0001.asc>

From gavin.sweet at skyracktelecom.com  Wed Mar 13 19:56:03 2013
From: gavin.sweet at skyracktelecom.com (Gavin Sweet)
Date: Wed, 13 Mar 2013 23:56:03 -0000
Subject: [Spce-user] Display of fee zone in subscriber CDR details GUI
Message-ID: <051801ce2046$52671550$f7353ff0$@sweet@skyracktelecom.com>

Hi all -

 

Trying to get description strings for zones/zone details rather than ID's in
the CDR details GUI..

 

Problem is that in the web admin interface, looking at an individual
subscriber, when I click the "Details" link next to a specific call, the
pop-over box displays carrier_billing_zone_id and customer_billing_zone_id
as integers, rather than pulling back the actual text. 

I can go look these up in the DB, but it makes the GUI much less useful for
doing quick admin or test checks.

Is this a bug, something fixable? Where can I find the code for this to see
what the issue might be?

 

Many thanks,

Gavin

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130313/f1838c15/attachment-0001.html>

From nh.developer at gmail.com  Thu Mar 14 07:34:17 2013
From: nh.developer at gmail.com (Developing Tomorrow)
Date: Thu, 14 Mar 2013 12:34:17 +0100
Subject: [Spce-user] media packets not travelling properly
Message-ID: <00ba01ce20a7$dd924120$98b6c360$@gmail.com>

Hi.,

 

This is my first mail to this mailing list, so please forgive my erroneous
procedure if it might be.

 

I have installed sipwise CE, with some pains but it does the job.

If I use a provider connected over normal line (any normal sip provider)

The signalling is done properly I can see the messages, but then the audio
is not following.

 

By that I mean I cannot hear the ringtone nor can I hear the other party
talking, nor they can hear me.

 

I have tried over a vpn, and I have one way audio.

 

The system is installed on a public server with the default iptables.

I have created only 1 domain that receive the external connections on port
5060

 

On the other hand, I have a vpn conection to a foreign network.

Client-----XXXXX (public ip) ----------provider********************* no
audio nor A nor B leg.

Client-----XXXXX(public ip)----------VPN_start
(10.8.0.1)-------------VPN_end(10.8.0.6)-----------provider (172.26.10.2
[not real])******** 1 way audio, they do not hear me

 

Anybody can shade a light here?

 

Thx

herve

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130314/ce2315ab/attachment-0001.html>

From nh.developer at gmail.com  Thu Mar 14 07:26:41 2013
From: nh.developer at gmail.com (Developing Tomorrow)
Date: Thu, 14 Mar 2013 12:26:41 +0100
Subject: [Spce-user] media packets not travelling properly
Message-ID: <00b501ce20a6$cdbdefc0$6939cf40$@gmail.com>

Hi.,

 

This is my first mail to this mailing list, so please forgive my erroneous
procedure if it might be.

 

I have installed sipwise CE, with some pains but it does the job.

If I use a provider connected over normal line (any normal sip provider)

The signalling is done properly I can see the messages, but then the audio
is not following.

 

By that I mean I cannot hear the ringtone nor can I hear the other party
talking, nor they can hear me.

 

I have tried over a vpn, and I have one way audio.

 

The system is installed on a public server with the default iptables.

I have created only 1 domain that receive the external connections on port
5060

 

On the other hand, I have a vpn conection to a foreign network.

Client-----XXXXX (public ip) ----------provider********************* no
audio nor A nor B leg.

Client-----XXXXX(public ip)----------VPN_start
(10.8.0.1)-------------VPN_end(10.8.0.6)-----------provider (172.26.10.2
[not real])******** 1 way audio, they do not hear me

 

Anybody can shade a light here?

 

Thx

herve

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130314/07c29be2/attachment-0001.html>

From dgrotti at sipwise.com  Thu Mar 14 07:49:01 2013
From: dgrotti at sipwise.com (Daniel Grotti)
Date: Thu, 14 Mar 2013 12:49:01 +0100
Subject: [Spce-user] media packets not travelling properly
In-Reply-To: <00ba01ce20a7$dd924120$98b6c360$@gmail.com>
References: <00ba01ce20a7$dd924120$98b6c360$@gmail.com>
Message-ID: <5141B92D.7010105@sipwise.com>

On 03/14/2013 12:34 PM, Developing Tomorrow wrote:
>
> Hi.,
>
>  
>
> This is my first mail to this mailing list, so please forgive my
> erroneous procedure if it might be.
>
>  
>
> I have installed sipwise CE, with some pains but it does the job.
>
> If I use a provider connected over normal line (any normal sip provider)
>
> The signalling is done properly I can see the messages, but then the
> audio is not following.
>
>  
>
> By that I mean I cannot hear the ringtone nor can I hear the other
> party talking, nor they can hear me.
>
>  
>
> I have tried over a vpn, and I have one way audio.
>
>  
>
> The system is installed on a public server with the default iptables.
>
> I have created only 1 domain that receive the external connections on
> port 5060
>
>  
>
> On the other hand, I have a vpn conection to a foreign network.
>
> Client-----XXXXX (public ip) ----------provider*********************
> no audio nor A nor B leg.
>
> Client-----XXXXX(public ip)----------VPN_start
> (10.8.0.1)-------------VPN_end(10.8.0.6)-----------provider
> (172.26.10.2 [not real])******** 1 way audio, they do not hear me
>
>  
>
> Anybody can shade a light here?
>
>  
>
> Thx
>
> herve
>
>  
>
>  
>
>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user


Hi Herve,
please check your mediaproxy process, is it running properly (ps -ef |
grep mediaproxy) and listening on your public IP ?
If the client is behind nat, please make sure it's not using any kind on
STUN server to discover its own public IP, or ICE or whatever, NAT
stuffs are handled by NGCP.

Try to run ngrep-sip and check the SIP messages. SDP part of INVITE to
provider and of 200 OK to the client should contain the mediaproxy's
public IP.


Daniel
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130314/be920fb1/attachment-0001.html>

From gavin.sweet at skyracktelecom.com  Thu Mar 14 08:51:15 2013
From: gavin.sweet at skyracktelecom.com (Gavin Sweet)
Date: Thu, 14 Mar 2013 12:51:15 -0000
Subject: [Spce-user] Problem (Bug?) in Billing Fee uploads
Message-ID: <05c801ce20b2$9d760cb0$d8622610$@sweet@skyracktelecom.com>

Hi - 
I have had some problems with destination regex support when uploading
Billing Fees using the batch Upload file function in the Admin GUI.
I have tested these problems with very small file uploads to ensure its not
a problem with file size/transfer etc.
I actually started off trying to ensure an exact match for the number
44118555, which should be simply ^44118555$ - however this doesn't upload.
I then investigated further and found some unexpected results as below. If
anyone can help, or let me know that this is a bug, that would be great.
Manual definition of these fees is not an option for workaround as we have
over 30000 records. An odd (inelegant) form of regex may force it to work,
but that's not what I'd prefer.

Other than case 7 and 8 which I tried for completeness, I think all the
below should work, but they don't.
----------------------------------------------------------------------------
---------------------
Case       Uploaded fee (csv record)                                  =>
Destination (in profile)
----------------------------------------------------------------------------
---------------------
1 (ok)     ^*.$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0                         =>
^*.$
2 (ok)     ^44118555.$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0                  =>
^44118555.$
3 (ok)     ^44118[1-9][0-9][0-9]$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0       =>
^44118[1-9][0-9][0-9]$
4 (ok)     ^4411855[5]$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0                 =>
^4411855[5]$
5 (fail)   ^44118555.*$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0                 =>
44118555
6 (fail)   ^44118555$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0                   =>
44118555
7 (fail)   '^44118555$',ABC,ABC,0,1,0.5,1,0,1,0.5,1,0                 =>
'^44118555$'
8 (fail)   "^44118555$",ABC,ABC,0,1,0.5,1,0,1,0.5,1,0                 =>
"^44118555$"
----------------------------------------------------------------------------
---------------------

Thanks,
Gavin




From gavin.sweet at skyracktelecom.com  Thu Mar 14 10:38:40 2013
From: gavin.sweet at skyracktelecom.com (Gavin Sweet)
Date: Thu, 14 Mar 2013 14:38:40 -0000
Subject: [Spce-user] Problem (Bug?) in Billing Fee uploads
In-Reply-To: <05c801ce20b2$9d760cb0$d8622610$@sweet@skyracktelecom.com>
References: <05c801ce20b2$9d760cb0$d8622610$@sweet@skyracktelecom.com>
Message-ID: <063801ce20c1$9f1da460$dd58ed20$@sweet@skyracktelecom.com>

My table of tests got a bit mangled.
Basically, this problem is when trying to upload fees from file. 
For example, for *exactly* 44118555, the destination should be: ^44118555$
But the ^ and $ get stripped for some reason during the upload from file.
Same with ^44118555.*$
But NOT with ^4411855[5]$ or  ^44118555.$

Cheers
Gavin

> -----Original Message-----
> From: spce-user-bounces at lists.sipwise.com [mailto:spce-user-
> bounces at lists.sipwise.com] On Behalf Of Gavin Sweet
> Sent: 14 March 2013 12:51
> To: spce-user at lists.sipwise.com
> Subject: [Spce-user] Problem (Bug?) in Billing Fee uploads
> 
> Hi -
> I have had some problems with destination regex support when uploading
> Billing Fees using the batch Upload file function in the Admin GUI.
> I have tested these problems with very small file uploads to ensure its
> not a problem with file size/transfer etc.
> I actually started off trying to ensure an exact match for the number
> 44118555, which should be simply ^44118555$ - however this doesn't
> upload.
> I then investigated further and found some unexpected results as below.
> If anyone can help, or let me know that this is a bug, that would be
> great.
> Manual definition of these fees is not an option for workaround as we
> have over 30000 records. An odd (inelegant) form of regex may force it
> to work, but that's not what I'd prefer.
> 
> Other than case 7 and 8 which I tried for completeness, I think all the
> below should work, but they don't.
> -----------------------------------------------------------------------
> -----
> ---------------------
> Case       Uploaded fee (csv record)
> =>
> Destination (in profile)
> -----------------------------------------------------------------------
> -----
> ---------------------
> 1 (ok)     ^*.$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> =>
> ^*.$
> 2 (ok)     ^44118555.$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> =>
> ^44118555.$
> 3 (ok)     ^44118[1-9][0-9][0-9]$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> =>
> ^44118[1-9][0-9][0-9]$
> 4 (ok)     ^4411855[5]$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> =>
> ^4411855[5]$
> 5 (fail)   ^44118555.*$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> =>
> 44118555
> 6 (fail)   ^44118555$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> =>
> 44118555
> 7 (fail)   '^44118555$',ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> =>
> '^44118555$'
> 8 (fail)   "^44118555$",ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> =>
> "^44118555$"
> -----------------------------------------------------------------------
> -----
> ---------------------
> 
> Thanks,
> Gavin
> 
> 
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
> -----
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2013.0.2904 / Virus Database: 2641/6166 - Release Date:
> 03/12/13




From jirka.jirout at mac.com  Thu Mar 14 11:16:01 2013
From: jirka.jirout at mac.com (Jirka Jirout)
Date: Thu, 14 Mar 2013 16:16:01 +0100
Subject: [Spce-user] BYE not correct when sent by called party
Message-ID: <672A65D6-6EDB-463D-8AE5-9EE631A9B076@mac.com>

Hello,

I have a problem with my client app. Both ends are connected via TLS on port 5889 of the LB. When the calling party terminates the call, it sends this and the call terminates correctly:

> BYE sip:127.0.0.1:5080 SIP/2.0'
> ...
> Record-Route: <sip:127.0.0.1:5888;ngcplb=yes;ftag=334359690;lr=on>'
> Record-Route: <sip:SERVER_PUBLIC_IP:5889;transport=tls;ftag=334359690;lr=on>'
> Via: SIP/2.0/UDP 127.0.0.1:5888;branch=z9hG4bKa6f1.654eced0f96edbfaf849c6c2ce831b08.0;i=d951'
> Via: SIP/2.0/TLS 192.168.178.23:54256;received=83.161.151.39;rport;branch=z9hG4bK685900316;rport=54256'
> From: <sip:611092 at OUR_DOMAIN>;tag=334359690'
> To: <sip:279208 at OUR_DOMAIN>;tag=46CD768A-5141E4A8000DAE1F-D32F2700'
> ...
> Route: <sip:127.0.0.1:5888;nat=yes;ngcplb=yes;ftag=334359690;lr=on>'
> Route: <sip:127.0.0.1:5062;lr=on;ftag=334359690;did=f08.4e81;mpd=ii;rtpprx=yes;vsf=dU5ZM1lleEZ6TkxNaUpYTlBkZk11TlkzWWV4Rno->'

When however the called party terminates the call, the packet looks like:

> BYE sip:127.0.0.1:5080 SIP/2.0'
> ...
> Record-Route: <sip:127.0.0.1:5888;ngcplb=yes;ftag=1501122437;lr=on>'
> Record-Route: <sip:SERVER_PUBLIC_IP:5889;transport=tls;ftag=1501122437;lr=on>'
> Via: SIP/2.0/UDP 127.0.0.1:5888;branch=z9hG4bKcc48.1dec6a8cec99aabfe552815890f9cfa7.0;i=8351'
> Via: SIP/2.0/TLS 192.168.178.46:60714;received=83.161.151.39;rport;branch=z9hG4bK1819339446;rport=60714'
> From: <sip:279208 at OUR_DOMAIN>;tag=1501122437'
> To: <sip:611092 at OUR_DOMAIN>;tag=0A95611E-5141E52C000C3E89-D81C3700'
> ...
> Route: <sip:127.0.0.1:5888;ftag=0A95611E-5141E52C000C3E89-D81C3700;lr=on>'

And the called party receives 403 Domain not served here

The kamailio log shows:

Mar 14 15:56:57 server /usr/sbin/kamailio[399]: WARNING: <core> [forward.c:268]: WARNING: get_send_socket: protocol/port mismatch (forced udp:127.0.0.1:5888, to tls:83.161.151.39:60714)

Now, I was under the impression, that the answers are always sent by using Via headers. Here it does not appear to be the case and . We will of course try to solve this in our client app's code, but I was wondering whether this could be somehow fixed on the server side only.

regards, jj


From nh.developer at gmail.com  Thu Mar 14 11:33:09 2013
From: nh.developer at gmail.com (Developing Tomorrow)
Date: Thu, 14 Mar 2013 16:33:09 +0100
Subject: [Spce-user] How to route the packet thru a VPN
Message-ID: <001401ce20c9$3be08bd0$b3a1a370$@gmail.com>

Hi

 

Sipwise routes the packets perfectly to standard operators

And to gateways that are connected directly thru a PPP connection.

 

How can we achieve the same when the VPN tunnel is created thru a tun+
device (ex: openvpn)

The tunnel works perfectly.

I can ping my gateway on the other side.

My routes are well, and can exchange files back and forth.

 

But sipwise can simply not find the gateway when put in sip peering..

When I dump the messages of the tun+ device, I absolutely see nothing
regarding sip messages, so it seems that it's not even trying.

 

Any idea would be appreciated.

 

thanks

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130314/a79c98d7/attachment-0001.html>

From nh.developer at gmail.com  Thu Mar 14 11:35:25 2013
From: nh.developer at gmail.com (Developing Tomorrow)
Date: Thu, 14 Mar 2013 16:35:25 +0100
Subject: [Spce-user] Spce-user Digest, Vol 28, Issue 25
In-Reply-To: <mailman.1164.1363271921.2547.spce-user@lists.sipwise.com>
References: <mailman.1164.1363271921.2547.spce-user@lists.sipwise.com>
Message-ID: <001901ce20c9$8d357810$a7a06830$@gmail.com>

Well, issue is "resolved"

Apparently on a fresh install, if you do not define the prefix properly, 
Sipwise will not let you call an unformatted number.
I have tried to dial my mobile number directly , but it did not work.
But putting 00 is would recognize the format and go thru...

So lesson is that all small details count.

Thank you for the assistance



_________________________________________________________________________
Please consider the environment before printing this email.
?
Notice: Please do not include customer identifying information (i.e.,
account number, social security, etc.) in email communications.
?
Privacy Statement: This message may contain confidential information that is
protected by law. If you received this message in error, please notify the
sender by fax or email and delete this message. If you properly received
this message, you may use its contents only in strictaccordance with our
instructions and privacy policy.
If any person makes a false or misleading representation to obtain customer
information, that person may have committed a federal crime, and we may
report that person to the proper authorities.

-----Original Message-----
From: spce-user-bounces at lists.sipwise.com
[mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of
spce-user-request at lists.sipwise.com
Sent: 14 March 2013 15:39
To: spce-user at lists.sipwise.com
Subject: Spce-user Digest, Vol 28, Issue 25

Send Spce-user mailing list submissions to
	spce-user at lists.sipwise.com

To subscribe or unsubscribe via the World Wide Web, visit
	http://lists.sipwise.com/listinfo/spce-user
or, via email, send a message with subject or body 'help' to
	spce-user-request at lists.sipwise.com

You can reach the person managing the list at
	spce-user-owner at lists.sipwise.com

When replying, please edit your Subject line so it is more specific than
"Re: Contents of Spce-user digest..."


Today's Topics:

   1. media packets not travelling properly (Developing Tomorrow)
   2. media packets not travelling properly (Developing Tomorrow)
   3. Re: media packets not travelling properly (Daniel Grotti)
   4. Problem (Bug?) in Billing Fee uploads (Gavin Sweet)
   5. Re: Problem (Bug?) in Billing Fee uploads (Gavin Sweet)


----------------------------------------------------------------------

Message: 1
Date: Thu, 14 Mar 2013 12:34:17 +0100
From: "Developing Tomorrow" <nh.developer at gmail.com>
Subject: [Spce-user] media packets not travelling properly
To: <spce-user at lists.sipwise.com>
Message-ID: <00ba01ce20a7$dd924120$98b6c360$@gmail.com>
Content-Type: text/plain; charset="us-ascii"

Hi.,

 

This is my first mail to this mailing list, so please forgive my erroneous
procedure if it might be.

 

I have installed sipwise CE, with some pains but it does the job.

If I use a provider connected over normal line (any normal sip provider)

The signalling is done properly I can see the messages, but then the audio
is not following.

 

By that I mean I cannot hear the ringtone nor can I hear the other party
talking, nor they can hear me.

 

I have tried over a vpn, and I have one way audio.

 

The system is installed on a public server with the default iptables.

I have created only 1 domain that receive the external connections on port
5060

 

On the other hand, I have a vpn conection to a foreign network.

Client-----XXXXX (public ip) ----------provider********************* no
audio nor A nor B leg.

Client-----XXXXX(public ip)----------VPN_start
(10.8.0.1)-------------VPN_end(10.8.0.6)-----------provider (172.26.10.2
[not real])******** 1 way audio, they do not hear me

 

Anybody can shade a light here?

 

Thx

herve

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130314/ce2315ab/
attachment-0001.htm>

------------------------------

Message: 2
Date: Thu, 14 Mar 2013 12:26:41 +0100
From: "Developing Tomorrow" <nh.developer at gmail.com>
Subject: [Spce-user] media packets not travelling properly
To: <spce-user at lists.sipwise.com>
Message-ID: <00b501ce20a6$cdbdefc0$6939cf40$@gmail.com>
Content-Type: text/plain; charset="us-ascii"

Hi.,

 

This is my first mail to this mailing list, so please forgive my erroneous
procedure if it might be.

 

I have installed sipwise CE, with some pains but it does the job.

If I use a provider connected over normal line (any normal sip provider)

The signalling is done properly I can see the messages, but then the audio
is not following.

 

By that I mean I cannot hear the ringtone nor can I hear the other party
talking, nor they can hear me.

 

I have tried over a vpn, and I have one way audio.

 

The system is installed on a public server with the default iptables.

I have created only 1 domain that receive the external connections on port
5060

 

On the other hand, I have a vpn conection to a foreign network.

Client-----XXXXX (public ip) ----------provider********************* no
audio nor A nor B leg.

Client-----XXXXX(public ip)----------VPN_start
(10.8.0.1)-------------VPN_end(10.8.0.6)-----------provider (172.26.10.2
[not real])******** 1 way audio, they do not hear me

 

Anybody can shade a light here?

 

Thx

herve

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130314/07c29be2/
attachment-0001.htm>

------------------------------

Message: 3
Date: Thu, 14 Mar 2013 12:49:01 +0100
From: Daniel Grotti <dgrotti at sipwise.com>
Subject: Re: [Spce-user] media packets not travelling properly
To: spce-user at lists.sipwise.com
Message-ID: <5141B92D.7010105 at sipwise.com>
Content-Type: text/plain; charset="iso-8859-1"

On 03/14/2013 12:34 PM, Developing Tomorrow wrote:
>
> Hi.,
>
>  
>
> This is my first mail to this mailing list, so please forgive my 
> erroneous procedure if it might be.
>
>  
>
> I have installed sipwise CE, with some pains but it does the job.
>
> If I use a provider connected over normal line (any normal sip 
> provider)
>
> The signalling is done properly I can see the messages, but then the 
> audio is not following.
>
>  
>
> By that I mean I cannot hear the ringtone nor can I hear the other 
> party talking, nor they can hear me.
>
>  
>
> I have tried over a vpn, and I have one way audio.
>
>  
>
> The system is installed on a public server with the default iptables.
>
> I have created only 1 domain that receive the external connections on 
> port 5060
>
>  
>
> On the other hand, I have a vpn conection to a foreign network.
>
> Client-----XXXXX (public ip) ----------provider*********************
> no audio nor A nor B leg.
>
> Client-----XXXXX(public ip)----------VPN_start 
> (10.8.0.1)-------------VPN_end(10.8.0.6)-----------provider
> (172.26.10.2 [not real])******** 1 way audio, they do not hear me
>
>  
>
> Anybody can shade a light here?
>
>  
>
> Thx
>
> herve
>
>  
>
>  
>
>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user


Hi Herve,
please check your mediaproxy process, is it running properly (ps -ef | grep
mediaproxy) and listening on your public IP ?
If the client is behind nat, please make sure it's not using any kind on
STUN server to discover its own public IP, or ICE or whatever, NAT stuffs
are handled by NGCP.

Try to run ngrep-sip and check the SIP messages. SDP part of INVITE to
provider and of 200 OK to the client should contain the mediaproxy's public
IP.


Daniel
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130314/be920fb1/
attachment-0001.htm>

------------------------------

Message: 4
Date: Thu, 14 Mar 2013 12:51:15 -0000
From: "Gavin Sweet" <gavin.sweet at skyracktelecom.com>
Subject: [Spce-user] Problem (Bug?) in Billing Fee uploads
To: <spce-user at lists.sipwise.com>
Message-ID: <05c801ce20b2$9d760cb0$d8622610$@sweet at skyracktelecom.com>
Content-Type: text/plain;	charset="us-ascii"

Hi -
I have had some problems with destination regex support when uploading
Billing Fees using the batch Upload file function in the Admin GUI.
I have tested these problems with very small file uploads to ensure its not
a problem with file size/transfer etc.
I actually started off trying to ensure an exact match for the number
44118555, which should be simply ^44118555$ - however this doesn't upload.
I then investigated further and found some unexpected results as below. If
anyone can help, or let me know that this is a bug, that would be great.
Manual definition of these fees is not an option for workaround as we have
over 30000 records. An odd (inelegant) form of regex may force it to work,
but that's not what I'd prefer.

Other than case 7 and 8 which I tried for completeness, I think all the
below should work, but they don't.
----------------------------------------------------------------------------
---------------------
Case       Uploaded fee (csv record)                                  =>
Destination (in profile)
----------------------------------------------------------------------------
---------------------
1 (ok)     ^*.$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0                         =>
^*.$
2 (ok)     ^44118555.$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0                  =>
^44118555.$
3 (ok)     ^44118[1-9][0-9][0-9]$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0       =>
^44118[1-9][0-9][0-9]$
4 (ok)     ^4411855[5]$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0                 =>
^4411855[5]$
5 (fail)   ^44118555.*$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0                 =>
44118555
6 (fail)   ^44118555$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0                   =>
44118555
7 (fail)   '^44118555$',ABC,ABC,0,1,0.5,1,0,1,0.5,1,0                 =>
'^44118555$'
8 (fail)   "^44118555$",ABC,ABC,0,1,0.5,1,0,1,0.5,1,0                 =>
"^44118555$"
----------------------------------------------------------------------------
---------------------

Thanks,
Gavin




------------------------------

Message: 5
Date: Thu, 14 Mar 2013 14:38:40 -0000
From: "Gavin Sweet" <gavin.sweet at skyracktelecom.com>
Subject: Re: [Spce-user] Problem (Bug?) in Billing Fee uploads
To: <spce-user at lists.sipwise.com>
Message-ID: <063801ce20c1$9f1da460$dd58ed20$@sweet at skyracktelecom.com>
Content-Type: text/plain;	charset="us-ascii"

My table of tests got a bit mangled.
Basically, this problem is when trying to upload fees from file. 
For example, for *exactly* 44118555, the destination should be: ^44118555$
But the ^ and $ get stripped for some reason during the upload from file.
Same with ^44118555.*$
But NOT with ^4411855[5]$ or  ^44118555.$

Cheers
Gavin

> -----Original Message-----
> From: spce-user-bounces at lists.sipwise.com [mailto:spce-user- 
> bounces at lists.sipwise.com] On Behalf Of Gavin Sweet
> Sent: 14 March 2013 12:51
> To: spce-user at lists.sipwise.com
> Subject: [Spce-user] Problem (Bug?) in Billing Fee uploads
> 
> Hi -
> I have had some problems with destination regex support when uploading 
> Billing Fees using the batch Upload file function in the Admin GUI.
> I have tested these problems with very small file uploads to ensure 
> its not a problem with file size/transfer etc.
> I actually started off trying to ensure an exact match for the number 
> 44118555, which should be simply ^44118555$ - however this doesn't 
> upload.
> I then investigated further and found some unexpected results as below.
> If anyone can help, or let me know that this is a bug, that would be 
> great.
> Manual definition of these fees is not an option for workaround as we 
> have over 30000 records. An odd (inelegant) form of regex may force it 
> to work, but that's not what I'd prefer.
> 
> Other than case 7 and 8 which I tried for completeness, I think all 
> the below should work, but they don't.
> ----------------------------------------------------------------------
> -
> -----
> ---------------------
> Case       Uploaded fee (csv record)
> =>
> Destination (in profile)
> ----------------------------------------------------------------------
> -
> -----
> ---------------------
> 1 (ok)     ^*.$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> =>
> ^*.$
> 2 (ok)     ^44118555.$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> =>
> ^44118555.$
> 3 (ok)     ^44118[1-9][0-9][0-9]$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> =>
> ^44118[1-9][0-9][0-9]$
> 4 (ok)     ^4411855[5]$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> =>
> ^4411855[5]$
> 5 (fail)   ^44118555.*$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> =>
> 44118555
> 6 (fail)   ^44118555$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> =>
> 44118555
> 7 (fail)   '^44118555$',ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> =>
> '^44118555$'
> 8 (fail)   "^44118555$",ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> =>
> "^44118555$"
> ----------------------------------------------------------------------
> -
> -----
> ---------------------
> 
> Thanks,
> Gavin
> 
> 
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
> -----
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2013.0.2904 / Virus Database: 2641/6166 - Release Date:
> 03/12/13




------------------------------

_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com
http://lists.sipwise.com/listinfo/spce-user


End of Spce-user Digest, Vol 28, Issue 25
*****************************************




From agranig at sipwise.com  Thu Mar 14 12:16:50 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Thu, 14 Mar 2013 17:16:50 +0100
Subject: [Spce-user] Delete subscriber does not delete 'trusted' entry
In-Reply-To: <035601ce2027$8e13aab0$aa3b0010$@murray@skyracktelecom.com>
References: <035601ce2027$8e13aab0$aa3b0010$@murray@skyracktelecom.com>
Message-ID: <5141F7F2.9020006@sipwise.com>

Hi John,

On 03/13/2013 09:15 PM, John Murray wrote:
> In v2.7 when I subscriber is deleted the entry in the trusted table is
> not deleted therefore when adding the same pattern to a new subscriber
> the original (deleted) subscriber is matched.

Thanks, we're going to have a look at that!

Andreas



From Stephen at newbreakcommunications.com  Thu Mar 14 12:39:29 2013
From: Stephen at newbreakcommunications.com (Stephen Donovan)
Date: Thu, 14 Mar 2013 16:39:29 +0000
Subject: [Spce-user] random registration dropping
In-Reply-To: <20130314150404.6daf9c05@multivac>
References: <2CDACF784B10A14EAAFAB6140DAE0D132409FC@EXCHANGE1.newbreakmanagement.local>
	<20130313230657.7eb033ac@quenya>
	<AF650DB9-A3FD-44B3-9E52-A38B53103CAB@newbreakcommunications.com>
	<20130314150404.6daf9c05@multivac>
Message-ID: <2CDACF784B10A14EAAFAB6140DAE0D1324262B@EXCHANGE1.newbreakmanagement.local>

We are seeing authentication failed messages in kamailio-proxy.log on registration and outbound calls.  Sometimes as many as 2-3 failed attempts before registration succeeds.  On outbound calls, usually just one failure, try again it goes through.

It is the user getting banned due to registration attempts.

Stephen



-----Original Message-----
From: Jon Bonilla [mailto:jbonilla at sipwise.com] 
Sent: Thursday, March 14, 2013 9:04 AM
To: Stephen Donovan
Subject: Re: [Spce-user] random registration dropping

El Thu, 14 Mar 2013 13:47:10 +0000
Stephen Donovan <Stephen at newbreakcommunications.com> escribi?:

> I'm going to sit back and wait, everything seems to have settled down 
> after update last night. If we see any more dropped calls I will 
> forward relevant logs.
> 
> We were seeing the ip temporarily show up banned sometimes and almost 
> immediately drop from the ban list.
> 
>

Is it set to 50 requests in 5 seconds? Is the ip banned or the user is? You could raise the limit or switch the protection off just for testing.

Please keep the list in cc


From gavin.sweet at skyracktelecom.com  Thu Mar 14 14:03:16 2013
From: gavin.sweet at skyracktelecom.com (Gavin Sweet)
Date: Thu, 14 Mar 2013 18:03:16 -0000
Subject: [Spce-user] Problem (Bug?) in Billing Fee uploads
In-Reply-To: <063801ce20c1$9f1da460$dd58ed20$@sweet@skyracktelecom.com>
References: <05c801ce20b2$9d760cb0$d8622610$@sweet@skyracktelecom.com>
	<063801ce20c1$9f1da460$dd58ed20$@sweet@skyracktelecom.com>
Message-ID: <06e601ce20de$34361070$9ca23150$@sweet@skyracktelecom.com>

I re-read the handbook regarding this point, but the behaviour is not
consistent.
According to the handbook, the web interface removes the regex, but this
isn't the case - only for some formats of regex.
I'd like really to be able to see the full code (including regex) that's
being matched, because that's the only way you can be sure you have a fee
that's correct.
At the moment it's pretty tricky to manage because you're not quite sure
what you're looking at.
Thanks for any light anyone can shed...




> -----Original Message-----
> From: spce-user-bounces at lists.sipwise.com [mailto:spce-user-
> bounces at lists.sipwise.com] On Behalf Of Gavin Sweet
> Sent: 14 March 2013 14:39
> To: spce-user at lists.sipwise.com
> Subject: Re: [Spce-user] Problem (Bug?) in Billing Fee uploads
> 
> My table of tests got a bit mangled.
> Basically, this problem is when trying to upload fees from file.
> For example, for *exactly* 44118555, the destination should be:
> ^44118555$ But the ^ and $ get stripped for some reason during the
> upload from file.
> Same with ^44118555.*$
> But NOT with ^4411855[5]$ or  ^44118555.$
> 
> Cheers
> Gavin
> 
> > -----Original Message-----
> > From: spce-user-bounces at lists.sipwise.com [mailto:spce-user-
> > bounces at lists.sipwise.com] On Behalf Of Gavin Sweet
> > Sent: 14 March 2013 12:51
> > To: spce-user at lists.sipwise.com
> > Subject: [Spce-user] Problem (Bug?) in Billing Fee uploads
> >
> > Hi -
> > I have had some problems with destination regex support when
> uploading
> > Billing Fees using the batch Upload file function in the Admin GUI.
> > I have tested these problems with very small file uploads to ensure
> > its not a problem with file size/transfer etc.
> > I actually started off trying to ensure an exact match for the number
> > 44118555, which should be simply ^44118555$ - however this doesn't
> > upload.
> > I then investigated further and found some unexpected results as
> below.
> > If anyone can help, or let me know that this is a bug, that would be
> > great.
> > Manual definition of these fees is not an option for workaround as we
> > have over 30000 records. An odd (inelegant) form of regex may force
> it
> > to work, but that's not what I'd prefer.
> >
> > Other than case 7 and 8 which I tried for completeness, I think all
> > the below should work, but they don't.
> > ---------------------------------------------------------------------
> -
> > -
> > -----
> > ---------------------
> > Case       Uploaded fee (csv record)
> > =>
> > Destination (in profile)
> > ---------------------------------------------------------------------
> -
> > -
> > -----
> > ---------------------
> > 1 (ok)     ^*.$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> > =>
> > ^*.$
> > 2 (ok)     ^44118555.$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> > =>
> > ^44118555.$
> > 3 (ok)     ^44118[1-9][0-9][0-9]$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> > =>
> > ^44118[1-9][0-9][0-9]$
> > 4 (ok)     ^4411855[5]$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> > =>
> > ^4411855[5]$
> > 5 (fail)   ^44118555.*$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> > =>
> > 44118555
> > 6 (fail)   ^44118555$,ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> > =>
> > 44118555
> > 7 (fail)   '^44118555$',ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> > =>
> > '^44118555$'
> > 8 (fail)   "^44118555$",ABC,ABC,0,1,0.5,1,0,1,0.5,1,0
> > =>
> > "^44118555$"
> > ---------------------------------------------------------------------
> -
> > -
> > -----
> > ---------------------
> >
> > Thanks,
> > Gavin
> >
> >
> > _______________________________________________
> > Spce-user mailing list
> > Spce-user at lists.sipwise.com
> > http://lists.sipwise.com/listinfo/spce-user
> > -----
> > No virus found in this message.
> > Checked by AVG - www.avg.com
> > Version: 2013.0.2904 / Virus Database: 2641/6166 - Release Date:
> > 03/12/13
> 
> 
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
> -----
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2013.0.2904 / Virus Database: 2641/6166 - Release Date:
> 03/12/13




From jbonilla at sipwise.com  Thu Mar 14 14:09:26 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Thu, 14 Mar 2013 19:09:26 +0100
Subject: [Spce-user] random registration dropping
In-Reply-To: <2CDACF784B10A14EAAFAB6140DAE0D1324262B@EXCHANGE1.newbreakmanagement.local>
References: <2CDACF784B10A14EAAFAB6140DAE0D132409FC@EXCHANGE1.newbreakmanagement.local>
	<20130313230657.7eb033ac@quenya>
	<AF650DB9-A3FD-44B3-9E52-A38B53103CAB@newbreakcommunications.com>
	<20130314150404.6daf9c05@multivac>
	<2CDACF784B10A14EAAFAB6140DAE0D1324262B@EXCHANGE1.newbreakmanagement.local>
Message-ID: <20130314190926.3dac1292@quenya>

El Thu, 14 Mar 2013 16:39:29 +0000
Stephen Donovan <Stephen at newbreakcommunications.com> escribi?:

> We are seeing authentication failed messages in kamailio-proxy.log on
> registration and outbound calls.  Sometimes as many as 2-3 failed attempts
> before registration succeeds.  On outbound calls, usually just one failure,
> try again it goes through.
> 
> It is the user getting banned due to registration attempts.
> 

If you're not provisioning the devices and the users might have wrong
credentials in their systems maybe you should disable the ddos protection in
config.yml and let the dos protection enabled.

btw, are you running the latest versions of the templates?

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130314/4122821f/attachment-0001.asc>

From agranig at sipwise.com  Thu Mar 14 15:19:48 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Thu, 14 Mar 2013 20:19:48 +0100
Subject: [Spce-user] Problem (Bug?) in Billing Fee uploads
In-Reply-To: <063801ce20c1$9f1da460$dd58ed20$@sweet@skyracktelecom.com>
References: <05c801ce20b2$9d760cb0$d8622610$@sweet@skyracktelecom.com>
	<063801ce20c1$9f1da460$dd58ed20$@sweet@skyracktelecom.com>
Message-ID: <514222D4.1070308@sipwise.com>

Hi,

On 03/14/2013 03:38 PM, Gavin Sweet wrote:
> My table of tests got a bit mangled.
> Basically, this problem is when trying to upload fees from file.
> For example, for *exactly* 44118555, the destination should be: ^44118555$
> But the ^ and $ get stripped for some reason during the upload from file.
> Same with ^44118555.*$
> But NOT with ^4411855[5]$ or  ^44118555.$

It's actually a bit of a consistency problem for historical reasons, 
mixed with an issue when displaying it, and it will be changed in the 
near future.

Currently it works like this:

If you put in a plain number like "441234", it internally gets extended 
to "^441234.*$". This is what people expected us to do in the past, as 
they only wanted to do prefix-based destination matching. When the value 
is displayed again on the admin panel, it is displayed as "441234" 
again. Although, if you check the billing.billing_fees table in the DB, 
you'll the the full regex there.

The same applies to IP addresses, where "1.2.3.4" gets 
"^.*@1\.2\.3\.4$", but are displayed in the admin panel as "1.2.3.4" 
again, and the same for SIP URIs like "user at example.org", which gets 
"^user at example\.org$".

Everything else will be passed to the DB as is. However, something like 
"^441234$" will be saved as is, but displayed as "441234", which is an 
inconsistency and is confusing, as I admit (I'd consider it as a bug).

Now since regex are already used quite heavily in rewrite rules, we have 
already decided to entirely drop this implicit and confusing "magic" in 
the next release, so people will be required to enter exactly the regex 
they want to match. It will break the logic of people who rely on this 
prefix-based thing, but properly documented (also on the admin panel 
itself) it should be manageable. For everyone else, it will be more obvious.

Andreas




From Stephen at newbreakcommunications.com  Thu Mar 14 15:57:28 2013
From: Stephen at newbreakcommunications.com (Stephen Donovan)
Date: Thu, 14 Mar 2013 19:57:28 +0000
Subject: [Spce-user] random registration dropping
In-Reply-To: <20130314190926.3dac1292@quenya>
References: <2CDACF784B10A14EAAFAB6140DAE0D132409FC@EXCHANGE1.newbreakmanagement.local>
	<20130313230657.7eb033ac@quenya>
	<AF650DB9-A3FD-44B3-9E52-A38B53103CAB@newbreakcommunications.com>
	<20130314150404.6daf9c05@multivac>
	<2CDACF784B10A14EAAFAB6140DAE0D1324262B@EXCHANGE1.newbreakmanagement.local>
	<20130314190926.3dac1292@quenya>
Message-ID: <2CDACF784B10A14EAAFAB6140DAE0D13242C28@EXCHANGE1.newbreakmanagement.local>

If I'm understanding correctly, the templates are for provisioning ?  I'm not currently using any provisioning, actually, we haven't figured out how to make it work.  Each ATA is manually configured and managed by us, we do not let the end user touch anything.  Credentials are correct, we will see one failed attempt and a couple seconds later a successful registration with no action on our part.

Stephen


-----Original Message-----
From: Jon Bonilla [mailto:jbonilla at sipwise.com] 
Sent: Thursday, March 14, 2013 1:09 PM
To: spce-user@
Cc: Stephen Donovan
Subject: Re: [Spce-user] random registration dropping

El Thu, 14 Mar 2013 16:39:29 +0000
Stephen Donovan <Stephen at newbreakcommunications.com> escribi?:

> We are seeing authentication failed messages in kamailio-proxy.log on 
> registration and outbound calls.  Sometimes as many as 2-3 failed 
> attempts before registration succeeds.  On outbound calls, usually 
> just one failure, try again it goes through.
> 
> It is the user getting banned due to registration attempts.
> 

If you're not provisioning the devices and the users might have wrong credentials in their systems maybe you should disable the ddos protection in config.yml and let the dos protection enabled.

btw, are you running the latest versions of the templates?


From lorenzo.mangani at gmail.com  Thu Mar 14 16:20:58 2013
From: lorenzo.mangani at gmail.com (Lorenzo Mangani)
Date: Thu, 14 Mar 2013 21:20:58 +0100
Subject: [Spce-user] random registration dropping
In-Reply-To: <2CDACF784B10A14EAAFAB6140DAE0D13242C28@EXCHANGE1.newbreakmanagement.local>
References: <2CDACF784B10A14EAAFAB6140DAE0D132409FC@EXCHANGE1.newbreakmanagement.local>
	<20130313230657.7eb033ac@quenya>
	<AF650DB9-A3FD-44B3-9E52-A38B53103CAB@newbreakcommunications.com>
	<20130314150404.6daf9c05@multivac>
	<2CDACF784B10A14EAAFAB6140DAE0D1324262B@EXCHANGE1.newbreakmanagement.local>
	<20130314190926.3dac1292@quenya>
	<2CDACF784B10A14EAAFAB6140DAE0D13242C28@EXCHANGE1.newbreakmanagement.local>
Message-ID: <CAKN+pHa9Z6Z3RXdGQ6ezNhCkHRHmcyg1JTVWw77pQYxyygLEsg@mail.gmail.com>

Stephen,

Can you share a signaling session for one such devices/events to see if the
device is sending anything unusual? (including the sequential failures
until the success)


Best,

Lorenzo Mangani

HOMER DEV TEAM
QXIP - Network Engineering


On Thu, Mar 14, 2013 at 8:57 PM, Stephen Donovan <
Stephen at newbreakcommunications.com> wrote:

> If I'm understanding correctly, the templates are for provisioning ?  I'm
> not currently using any provisioning, actually, we haven't figured out how
> to make it work.  Each ATA is manually configured and managed by us, we do
> not let the end user touch anything.  Credentials are correct, we will see
> one failed attempt and a couple seconds later a successful registration
> with no action on our part.
>
> Stephen
>
>
> -----Original Message-----
> From: Jon Bonilla [mailto:jbonilla at sipwise.com]
> Sent: Thursday, March 14, 2013 1:09 PM
> To: spce-user@
> Cc: Stephen Donovan
> Subject: Re: [Spce-user] random registration dropping
>
> El Thu, 14 Mar 2013 16:39:29 +0000
> Stephen Donovan <Stephen at newbreakcommunications.com> escribi?:
>
> > We are seeing authentication failed messages in kamailio-proxy.log on
> > registration and outbound calls.  Sometimes as many as 2-3 failed
> > attempts before registration succeeds.  On outbound calls, usually
> > just one failure, try again it goes through.
> >
> > It is the user getting banned due to registration attempts.
> >
>
> If you're not provisioning the devices and the users might have wrong
> credentials in their systems maybe you should disable the ddos protection
> in config.yml and let the dos protection enabled.
>
> btw, are you running the latest versions of the templates?
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
>



--
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130314/ad5816ea/attachment-0001.html>

From jbonilla at sipwise.com  Thu Mar 14 16:49:41 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Thu, 14 Mar 2013 21:49:41 +0100
Subject: [Spce-user] random registration dropping
In-Reply-To: <2CDACF784B10A14EAAFAB6140DAE0D13242C28@EXCHANGE1.newbreakmanagement.local>
References: <2CDACF784B10A14EAAFAB6140DAE0D132409FC@EXCHANGE1.newbreakmanagement.local>
	<20130313230657.7eb033ac@quenya>
	<AF650DB9-A3FD-44B3-9E52-A38B53103CAB@newbreakcommunications.com>
	<20130314150404.6daf9c05@multivac>
	<2CDACF784B10A14EAAFAB6140DAE0D1324262B@EXCHANGE1.newbreakmanagement.local>
	<20130314190926.3dac1292@quenya>
	<2CDACF784B10A14EAAFAB6140DAE0D13242C28@EXCHANGE1.newbreakmanagement.local>
Message-ID: <20130314214941.236f9d6f@quenya>

El Thu, 14 Mar 2013 19:57:28 +0000
Stephen Donovan <Stephen at newbreakcommunications.com> escribi?:

> If I'm understanding correctly, the templates are for provisioning ?

No. Templates are related to the configuration files. Just to be sure that
you're running latest versions. You can run "apt-get update && apt-get upgrade
&& ngcpcfg apply"

>  I'm not
> currently using any provisioning, actually, we haven't figured out how to
> make it work.  Each ATA is manually configured and managed by us, we do not
> let the end user touch anything.  Credentials are correct, we will see one
> failed attempt and a couple seconds later a successful registration with no
> action on our part.
> 

This is correct as the first attempt will receive a 401 and the second one will
receive the 200ok. That way the user is not banned as you need 3 attempts to be
banned with the default settings.


-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130314/68cf23b0/attachment-0001.asc>

From thilo.bangert at gmail.com  Thu Mar 14 17:32:21 2013
From: thilo.bangert at gmail.com (Thilo Bangert)
Date: Thu, 14 Mar 2013 22:32:21 +0100
Subject: [Spce-user] Private IP space without NAT
Message-ID: <7749345.FK6hup84n9@marsupilami>

Hi,

we have spce running in private rfc1918 IP space - all clients and peers are 
in either on the same subnet or in another, directly routed subnet, which is 
also in private IP space.

spce detects the clients being on private IP space and enables NAT support.

However, as outlined above, there is no NAT in our setup. I havent experienced 
any problem yet, but should i be careful? i already have set 
natping_interval: 0
is there anything else i should/could do?

ideally i would like to disable NAT processing (nathelper et. al)

thanks

kind regards
Thilo



From jbonilla at sipwise.com  Fri Mar 15 05:14:31 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Fri, 15 Mar 2013 10:14:31 +0100
Subject: [Spce-user] Private IP space without NAT
In-Reply-To: <7749345.FK6hup84n9@marsupilami>
References: <7749345.FK6hup84n9@marsupilami>
Message-ID: <20130315101431.2d153fe4@quenya>

El Thu, 14 Mar 2013 22:32:21 +0100
Thilo Bangert <thilo.bangert at gmail.com> escribi?:

> Hi,
> 
> we have spce running in private rfc1918 IP space - all clients and peers are 
> in either on the same subnet or in another, directly routed subnet, which is 
> also in private IP space.
> 
> spce detects the clients being on private IP space and enables NAT support.
> 
> However, as outlined above, there is no NAT in our setup. I havent
> experienced any problem yet, but should i be careful? i already have set 
> natping_interval: 0
> is there anything else i should/could do?
> 
> ideally i would like to disable NAT processing (nathelper et. al)
> 

There's no way that I know to easily disable NAT support in the spce. Usually,
there shouldn't be any problem in your nonat scenario. The only reason I could
imagine for disabling NAT is for performance reasons and it should be a quite
big install if one server doesn't handle all your traffic (you would need a
sip:carrier :P)

regarding performance, the NAT keepalive traffic, which you already disabled
and the rtp traffic, which can be disabled via the "never_use_rtpproxy" in the
admin interface are the options to disable.

If the reason is for disabling ip mangling in SIP, you have an array in
config.yml called  "kamailio.lb.nattest_exception_ips" you could use as guide
to check where you can create a custom template to disable the contact mangling.

Maybe someone of my colleagues at sipwise has a cleaner solution, but you don't
have a problem IMHO.


cheers,

Jon

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130315/f118f532/attachment-0001.asc>

From jbonilla at sipwise.com  Fri Mar 15 05:16:18 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Fri, 15 Mar 2013 10:16:18 +0100
Subject: [Spce-user] Spce-user Digest, Vol 28, Issue 25
In-Reply-To: <001901ce20c9$8d357810$a7a06830$@gmail.com>
References: <mailman.1164.1363271921.2547.spce-user@lists.sipwise.com>
	<001901ce20c9$8d357810$a7a06830$@gmail.com>
Message-ID: <20130315101618.20f05c26@quenya>

El Thu, 14 Mar 2013 16:35:25 +0100
"Developing Tomorrow" <nh.developer at gmail.com> escribi?:

> Well, issue is "resolved"
> 
> Apparently on a fresh install, if you do not define the prefix properly, 
> Sipwise will not let you call an unformatted number.
> I have tried to dial my mobile number directly , but it did not work.
> But putting 00 is would recognize the format and go thru...
> 
> So lesson is that all small details count.
> 
> Thank you for the assistance
> 
>

Hi

Sorry, but with this topic (daily Digest) and without any quotation I can't see
what issue you are talking about. 

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130315/b055e2d5/attachment-0001.asc>

From jbonilla at sipwise.com  Fri Mar 15 05:18:15 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Fri, 15 Mar 2013 10:18:15 +0100
Subject: [Spce-user] How to route the packet thru a VPN
In-Reply-To: <001401ce20c9$3be08bd0$b3a1a370$@gmail.com>
References: <001401ce20c9$3be08bd0$b3a1a370$@gmail.com>
Message-ID: <20130315101815.49382ef7@quenya>

El Thu, 14 Mar 2013 16:33:09 +0100
"Developing Tomorrow" <nh.developer at gmail.com> escribi?:

> Hi
> 
>  
> 
> Sipwise routes the packets perfectly to standard operators
> 
> And to gateways that are connected directly thru a PPP connection.
> 
>  
> 
> How can we achieve the same when the VPN tunnel is created thru a tun+
> device (ex: openvpn)
> 
> The tunnel works perfectly.
> 
> I can ping my gateway on the other side.
> 
> My routes are well, and can exchange files back and forth.
> 
>  
> 
> But sipwise can simply not find the gateway when put in sip peering..
> 
> When I dump the messages of the tun+ device, I absolutely see nothing
> regarding sip messages, so it seems that it's not even trying.
> 
>  
> 
> Any idea would be appreciated.
> 
>  
> 
> thanks
> 


What does a network capture say? And a ngrep-sip capture? There's another
thread for this issue where a colleague asks for a network capture AFAIKS

The thread is called ""[Spce-user] media packets not travelling properly""
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130315/c4fa27f7/attachment-0001.asc>

From jbonilla at sipwise.com  Fri Mar 15 05:21:12 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Fri, 15 Mar 2013 10:21:12 +0100
Subject: [Spce-user] Billing Profiles/Fees
In-Reply-To: <EE7149FDD60B3045A4377F05AAFFF3F118342A46@ISS-Server.ISS.local>
References: <EE7149FDD60B3045A4377F05AAFFF3F118342A46@ISS-Server.ISS.local>
Message-ID: <20130315102112.37d2c298@quenya>

El Wed, 13 Mar 2013 15:52:10 +0000
Christopher Saroli <csaroli at isysp.com> escribi?:

> Little confused on this, every call I make/receive does not get a fee applied
> to it.  Just trying right now to apply a fee to all calls.  I might be
> totally wrong here, any info would be appreciated.
> 
> This is my config:
> 
> Billing Profile:
> Interval Charge:  0.02
> Freetime: 	  0
> Free Cash: 	  0
> Monthly Lim	  0
> Lock		  none
> Currency: 	 $
> 
> 
> Bililng Fees
> Destination:  	.*
> Zone:      	All
> Init Rate:	1
> Init interval:	1
> Follow Rate: 	1
> Follow interval: 1
> 
> 
> Also I have no outgoing calls recorded in the CDR under the admin interface
> but in the customer portal they are there. Is this by design or is something
> wrong.
>

Well, he cdrs should be shown in both interfaces. We can go there later on. But
anyways, to check if your calls are rated fine you should check the exported
cdrs (if any).

Is rate-o-mat running in your system? If not, your calls won't be rated and you
won't get exported cdrs.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130315/a3e945f0/attachment-0001.asc>

From jbonilla at sipwise.com  Fri Mar 15 05:22:07 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Fri, 15 Mar 2013 10:22:07 +0100
Subject: [Spce-user] TLS problem
In-Reply-To: <FB145375-D2D0-4DAC-AF84-090544413297@mac.com>
References: <FB145375-D2D0-4DAC-AF84-090544413297@mac.com>
Message-ID: <20130315102207.7bb93f0a@quenya>

El Mon, 11 Mar 2013 17:27:03 +0100
Jirka Jirout <jirka.jirout at mac.com> escribi?:

> Hello,
> 
> 
> >>  - No answer ever comes back
> >>  - When I run ngrep-sip on the server, I do not see the message anywhere,
> >>  although tcpdump shows it on the external interface
> >> 
> > 
> 
> > are you listening tcp port 5061?
> I am not, we use a different port. But I do try to connect to the port on
> which TLS is enabled.
> 
> > I'll try to take a look when I have some more time but meanwhile I'd suggest
> > you to crosspost this to the kamailio-users mailing list as this issue could
> > match there and you might also get an answer there.
> Thanks for suggestion, I will do that. If I do not manage to fix this soon, I
> will open a support ticket for this with you as well.
> 
> 

I'm following this issue in the kamailio mailing list. Looks like it fits
better out there. Please let us know any updates.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130315/01a1ba6b/attachment-0001.asc>

From jbonilla at sipwise.com  Fri Mar 15 05:26:30 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Fri, 15 Mar 2013 10:26:30 +0100
Subject: [Spce-user] Send peer inbound call straight to voicemail
In-Reply-To: <021301ce1b80$79b93db0$6d2bb910$@murray@skyracktelecom.com>
References: <021301ce1b80$79b93db0$6d2bb910$@murray@skyracktelecom.com>
Message-ID: <20130315102630.32c7a024@quenya>

El Thu, 7 Mar 2013 22:09:44 -0000
"John Murray" <john.murray at skyracktelecom.com> escribi?:

> Hi,
> 
>  
> 
> I want to send a specific call type to voicemail.
> 
>  
> 
> I send a call out to a PBX peer who has no voicemail. If the handset is not
> answered I get a call from the peer with RURI 500 which is the voicemail
> number, From is the original caller and a Divert: header with the subscriber
> ddi.
> 
>  
> 
> I know how to take off the Divert header and get the subscriber ddi but what
> would I put in the RURI to send straight to the subscribers voicemail?
> 
> I've tried prefixing 'vmu' to the ddi but that didn't work.
> 
>  

what was the ngrep-sip (including port 5062), the kamailio-proxy log and the
asterisk console output of that call? 

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130315/14f6a6d5/attachment-0001.asc>

From jirka.jirout at mac.com  Fri Mar 15 06:00:54 2013
From: jirka.jirout at mac.com (Jirka Jirout)
Date: Fri, 15 Mar 2013 11:00:54 +0100
Subject: [Spce-user] BYE not correct when sent by called party
In-Reply-To: <mailman.1167.1363279166.2547.spce-user@lists.sipwise.com>
References: <mailman.1167.1363279166.2547.spce-user@lists.sipwise.com>
Message-ID: <E56D2EA8-47FE-4504-9A1C-90013B633FC4@mac.com>

Hello,

I have tried to "fix" this by using session timers, but apparently without much of success. My understanding is that the B2BUA will send UPDATE packets to both parties and if one of them fails to answer, it will send BYE to the other party and drop the call on the proxy.

I want the timeout to be 5 seconds, so I set the following in the subscriber preferences (using the admini interface):
	sst_expires: 10 
	sst_min_timer: 9
	sst_max_timer: 90
	sst_refresh_method: tried all three :-)

I have restarted sems, but the UPDATE or re-INVITE messages are still not being sent to the client. I looked into sems.log and I can see:

[#7f725e02c720] [readFromConfiguration, SBCCallProfile.cpp:229] INFO: SBC:      SST disabled

Why is the B2BUA ignoring my config? Are the timeouts perhaps too short?

regards, jj


From jbonilla at sipwise.com  Fri Mar 15 06:21:38 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Fri, 15 Mar 2013 11:21:38 +0100
Subject: [Spce-user] BYE not correct when sent by called party
In-Reply-To: <E56D2EA8-47FE-4504-9A1C-90013B633FC4@mac.com>
References: <mailman.1167.1363279166.2547.spce-user@lists.sipwise.com>
	<E56D2EA8-47FE-4504-9A1C-90013B633FC4@mac.com>
Message-ID: <20130315112138.3c7aec3f@quenya>

El Fri, 15 Mar 2013 11:00:54 +0100
Jirka Jirout <jirka.jirout at mac.com> escribi?:

> Hello,
> 
> I have tried to "fix" this by using session timers, but apparently without
> much of success. My understanding is that the B2BUA will send UPDATE packets
> to both parties and if one of them fails to answer, it will send BYE to the
> other party and drop the call on the proxy.
> 
> I want the timeout to be 5 seconds, so I set the following in the subscriber
> preferences (using the admini interface): sst_expires: 10 
> 	sst_min_timer: 9
> 	sst_max_timer: 90
> 	sst_refresh_method: tried all three :-)
> 
> I have restarted sems, but the UPDATE or re-INVITE messages are still not
> being sent to the client. I looked into sems.log and I can see:
> 
> [#7f725e02c720] [readFromConfiguration, SBCCallProfile.cpp:229] INFO:
> SBC:      SST disabled
> 
> Why is the B2BUA ignoring my config? Are the timeouts perhaps too short?
> 


Do you have the sst_enable option enabled? By default it's no.

If so, you'll need to set the sems debug to 3 to check the logs. You can do it
via "sems-stats -c "set_loglevel 3""
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130315/2d895f1e/attachment-0001.asc>

From thilo.bangert at gmail.com  Fri Mar 15 07:36:52 2013
From: thilo.bangert at gmail.com (Thilo Bangert)
Date: Fri, 15 Mar 2013 12:36:52 +0100
Subject: [Spce-user] Private IP space without NAT
In-Reply-To: <20130315101431.2d153fe4@quenya>
References: <7749345.FK6hup84n9@marsupilami> <20130315101431.2d153fe4@quenya>
Message-ID: <3202687.qLYUjX0zSM@marsupilami>

On Friday 15 March 2013 10:14:31 Jon Bonilla wrote:
> El Thu, 14 Mar 2013 22:32:21 +0100
> 
> Thilo Bangert <thilo.bangert at gmail.com> escribi?:
> > Hi,
> > 
> > we have spce running in private rfc1918 IP space - all clients and peers
> > are in either on the same subnet or in another, directly routed subnet,
> > which is also in private IP space.
> > 
> > spce detects the clients being on private IP space and enables NAT
> > support.
> > 
> > However, as outlined above, there is no NAT in our setup. I havent
> > experienced any problem yet, but should i be careful? i already have set
> > natping_interval: 0
> > is there anything else i should/could do?
> > 
> > ideally i would like to disable NAT processing (nathelper et. al)
> 
> There's no way that I know to easily disable NAT support in the spce.
> Usually, there shouldn't be any problem in your nonat scenario. The only
> reason I could imagine for disabling NAT is for performance reasons and it
> should be a quite big install if one server doesn't handle all your traffic
> (you would need a sip:carrier :P)

ok - we're not there (yet :)

> 
> regarding performance, the NAT keepalive traffic, which you already disabled
> and the rtp traffic, which can be disabled via the "never_use_rtpproxy" in
> the admin interface are the options to disable.

ah - yes. that one i also already have disabled.

> 
> If the reason is for disabling ip mangling in SIP, you have an array in
> config.yml called  "kamailio.lb.nattest_exception_ips" you could use as
> guide to check where you can create a custom template to disable the
> contact mangling.

i dont have that option in my 2.7 config - 2.8 feature? however, there is no 
mangling going on as far as i can tell, and currently i dont mind spending a 
few cpu cycles on doing nothing...

> 
> Maybe someone of my colleagues at sipwise has a cleaner solution, but you
> don't have a problem IMHO.

thanks for confirming this.

kind regards
Thilo

> 
> 
> cheers,
> 
> Jon



From jirka.jirout at mac.com  Fri Mar 15 08:56:23 2013
From: jirka.jirout at mac.com (Jirka Jirout)
Date: Fri, 15 Mar 2013 13:56:23 +0100
Subject: [Spce-user] BYE not correct when sent by called party
References: <1D5937E1-61DE-487E-B8DB-A3281360619A@me.com>
Message-ID: <CBAC996A-4FD4-4F79-B43F-0765063DAA01@mac.com>

Hello,

> Do you have the sst_enable option enabled? By default it's no.

I did enable it for the domain using administration interface. Does it need to be enabled globally somehow?

I have also tried to enable RTP timeout by setting dead_rtp_time, but that does not terminate the connection either.

Just for the record: I have restarted SEMS and both lb and proxy kamailios.

regards, jj




From nh.developer at gmail.com  Fri Mar 15 09:22:54 2013
From: nh.developer at gmail.com (Developing Tomorrow)
Date: Fri, 15 Mar 2013 14:22:54 +0100
Subject: [Spce-user]  media packets not travelling properly
Message-ID: <004301ce2180$34912120$9db36360$@gmail.com>

Hi.,

 

Spce is a powerful platform. My original problems seems to be with the
router used on the other side of the VPN. A TPLINK 1043RN using DD-WRT
firmware.

After a few reboots, it seems that the modem finally let the info pass thru.

I have added some new machines on the same VPN tunnel but with a different
router (same make and firware) but am having the strange effect. Not
working.

 

Am still analysing the traffic thru the vpn and will provide some captures
if somebody can help.

The spce server is running three different VPN tunnels (10.8.0.1, 10.9.0.1,
10.10.0.1.)

10.8.0.1 is working perfectly..

 

 

 

        [|sip]

13:15:48.621427 IP (tos 0x0, ttl 63, id 0, offset 0, flags [DF], proto UDP
(17), length 31)

    10.8.0.12.sip > 10.8.0.1.sip: SIP, length: 3

 

        [|sip]

13:15:48.621476 IP (tos 0xc0, ttl 64, id 54459, offset 0, flags [none],
proto ICMP (1), length 59)

    10.8.0.1 > 10.8.0.12: ICMP 10.8.0.1 udp port sip unreachable, length 39

        IP (tos 0x0, ttl 63, id 0, offset 0, flags [DF], proto UDP (17),
length 31)

    10.8.0.12.sip > 10.8.0.1.sip: SIP, length: 3

 

        [|sip]

13:15:58.603201 IP (tos 0x0, ttl 63, id 0, offset 0, flags [DF], proto UDP
(17), length 31)

    10.8.0.12.sip > 10.8.0.1.sip: SIP, length: 3

 

        [|sip]

13:15:58.603235 IP (tos 0xc0, ttl 64, id 54460, offset 0, flags [none],
proto ICMP (1), length 59)

    10.8.0.1 > 10.8.0.12: ICMP 10.8.0.1 udp port sip unreachable, length 39

        IP (tos 0x0, ttl 63, id 0, offset 0, flags [DF], proto UDP (17),
length 31)

    10.8.0.12.sip > 10.8.0.1.sip: SIP, length: 3

 

        [|sip]

13:16:08.606366 IP (tos 0x0, ttl 63, id 0, offset 0, flags [DF], proto UDP
(17), length 31)

    10.8.0.12.sip > 10.8.0.1.sip: SIP, length: 3

 

        [|sip]

13:16:08.606420 IP (tos 0xc0, ttl 64, id 54461, offset 0, flags [none],
proto ICMP (1), length 59)

    10.8.0.1 > 10.8.0.12: ICMP 10.8.0.1 udp port sip unreachable, length 39

        IP (tos 0x0, ttl 63, id 0, offset 0, flags [DF], proto UDP (17),
length 31)

    10.8.0.12.sip > 10.8.0.1.sip: SIP, length: 3

 

This is the view of the tunnel

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130315/da98090b/attachment-0001.html>

From gavin.sweet at skyracktelecom.com  Fri Mar 15 10:20:43 2013
From: gavin.sweet at skyracktelecom.com (Gavin Sweet)
Date: Fri, 15 Mar 2013 14:20:43 -0000
Subject: [Spce-user] Problem (Bug?) in Billing Fee uploads
In-Reply-To: <514222D4.1070308@sipwise.com>
References: <05c801ce20b2$9d760cb0$d8622610$@sweet@skyracktelecom.com>	<063801ce20c1$9f1da460$dd58ed20$@sweet@skyracktelecom.com>
	<514222D4.1070308@sipwise.com>
Message-ID: <07d901ce2188$475450e0$d5fcf2a0$@sweet@skyracktelecom.com>

Thanks Andreas, very clear.

In order to preserve regex consistency, we propose until the new release to
use this format for a full number (e.g.):
^4411855[5]$   (=UK dir enquiries)
which is functionally the same as ^4411855$ for but allows us to have the
same in UI and DB.

Or for a prefix (e.g.):
^3360[5]    (=France mobile)
Which, again, should match ^33605.*$

If you think this will fail or break something else, please shout.

Kind regards,
Gavin


> -----Original Message-----
> From: spce-user-bounces at lists.sipwise.com [mailto:spce-user-
> bounces at lists.sipwise.com] On Behalf Of Andreas Granig
> Sent: 14 March 2013 19:20
> To: spce-user at lists.sipwise.com
> Subject: Re: [Spce-user] Problem (Bug?) in Billing Fee uploads
> 
> Hi,
> 
> On 03/14/2013 03:38 PM, Gavin Sweet wrote:
> > My table of tests got a bit mangled.
> > Basically, this problem is when trying to upload fees from file.
> > For example, for *exactly* 44118555, the destination should be:
> > ^44118555$ But the ^ and $ get stripped for some reason during the
> upload from file.
> > Same with ^44118555.*$
> > But NOT with ^4411855[5]$ or  ^44118555.$
> 
> It's actually a bit of a consistency problem for historical reasons,
> mixed with an issue when displaying it, and it will be changed in the
> near future.
> 
> Currently it works like this:
> 
> If you put in a plain number like "441234", it internally gets extended
> to "^441234.*$". This is what people expected us to do in the past, as
> they only wanted to do prefix-based destination matching. When the
> value is displayed again on the admin panel, it is displayed as
> "441234"
> again. Although, if you check the billing.billing_fees table in the DB,
> you'll the the full regex there.
> 
> The same applies to IP addresses, where "1.2.3.4" gets
> "^.*@1\.2\.3\.4$", but are displayed in the admin panel as "1.2.3.4"
> again, and the same for SIP URIs like "user at example.org", which gets
> "^user at example\.org$".
> 
> Everything else will be passed to the DB as is. However, something like
> "^441234$" will be saved as is, but displayed as "441234", which is an
> inconsistency and is confusing, as I admit (I'd consider it as a bug).
> 
> Now since regex are already used quite heavily in rewrite rules, we
> have already decided to entirely drop this implicit and confusing
> "magic" in the next release, so people will be required to enter
> exactly the regex they want to match. It will break the logic of people
> who rely on this prefix-based thing, but properly documented (also on
> the admin panel
> itself) it should be manageable. For everyone else, it will be more
> obvious.
> 
> Andreas
> 
> 
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
> -----
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2013.0.2904 / Virus Database: 2641/6166 - Release Date:
> 03/12/13




From jbonilla at sipwise.com  Fri Mar 15 11:03:23 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Fri, 15 Mar 2013 16:03:23 +0100
Subject: [Spce-user] BYE not correct when sent by called party
In-Reply-To: <CBAC996A-4FD4-4F79-B43F-0765063DAA01@mac.com>
References: <1D5937E1-61DE-487E-B8DB-A3281360619A@me.com>
	<CBAC996A-4FD4-4F79-B43F-0765063DAA01@mac.com>
Message-ID: <20130315160323.7eca65ec@multivac>

El Fri, 15 Mar 2013 13:56:23 +0100
Jirka Jirout <jirka.jirout at mac.com> escribi?:

> Hello,
> 
> > Do you have the sst_enable option enabled? By default it's no.
> 
> I did enable it for the domain using administration interface. Does it need
> to be enabled globally somehow?
> 

No. If you have enabled for the domain it should be enabled for all
subscribers. But as you were not listing the option...

	sst_expires: 10 
	sst_min_timer: 9
	sst_max_timer: 90

	*** sst_enable: yes  ***

> I have also tried to enable RTP timeout by setting dead_rtp_time, but that
> does not terminate the connection either.

That's for the rtp going to sems. In a normal call scenario the rtp doesn't
traverse sems. You have the same feature in our mediaproxy-ng.

> 
> Just for the record: I have restarted SEMS and both lb and proxy kamailios.
> 
> 


If you have it enabled, please send us the invite sent by the proxy to sems.
sems is controlled by custom headers sent by the proxy (be careful as they
contain sensitive info). 

If the headers are right, enable sems debug and let's see what it says.




From Stephen at newbreakcommunications.com  Fri Mar 15 11:21:28 2013
From: Stephen at newbreakcommunications.com (Stephen Donovan)
Date: Fri, 15 Mar 2013 15:21:28 +0000
Subject: [Spce-user] random registration dropping
In-Reply-To: <20130314214941.236f9d6f@quenya>
References: <2CDACF784B10A14EAAFAB6140DAE0D132409FC@EXCHANGE1.newbreakmanagement.local>
	<20130313230657.7eb033ac@quenya>
	<AF650DB9-A3FD-44B3-9E52-A38B53103CAB@newbreakcommunications.com>
	<20130314150404.6daf9c05@multivac>
	<2CDACF784B10A14EAAFAB6140DAE0D1324262B@EXCHANGE1.newbreakmanagement.local>
	<20130314190926.3dac1292@quenya>
	<2CDACF784B10A14EAAFAB6140DAE0D13242C28@EXCHANGE1.newbreakmanagement.local>,
	<20130314214941.236f9d6f@quenya>
Message-ID: <2CDACF784B10A14EAAFAB6140DAE0D13243D59@EXCHANGE1.newbreakmanagement.local>

I have done the update, 2 days ago.

I'm finding this issue only with Linksys SPA8000 ATAs.  I swapped a client yesterday to a grandstream gxw4024 I had on the shelf, all problems are gone.  Other Linksys models are functioning well.  Jeremie Chism may be able to chime in with his experiences with the Linksys units as well, his customers are on our server as well.  I see the 401 on first attempt, I understand that to be the case, what I'm seeing is multiple attempts at every registration renewal.  I have the limit increased to 10 before banning so they do not get banned now but this still causes a 20 sec to 1 min interruption in service at most registration renewals but not all.

Again, only seeing this on SPA8000 units, nothing else.  I'm deploying all new grandstream units going forward and have had no issues with them.

Stephen



________________________________________
From: Jon Bonilla [jbonilla at sipwise.com]
Sent: Thursday, March 14, 2013 3:49 PM
To: spce-user@
Cc: Stephen Donovan
Subject: Re: [Spce-user] random registration dropping

El Thu, 14 Mar 2013 19:57:28 +0000
Stephen Donovan <Stephen at newbreakcommunications.com> escribi?:

> If I'm understanding correctly, the templates are for provisioning ?

No. Templates are related to the configuration files. Just to be sure that
you're running latest versions. You can run "apt-get update && apt-get upgrade
&& ngcpcfg apply"

>  I'm not
> currently using any provisioning, actually, we haven't figured out how to
> make it work.  Each ATA is manually configured and managed by us, we do not
> let the end user touch anything.  Credentials are correct, we will see one
> failed attempt and a couple seconds later a successful registration with no
> action on our part.
>

This is correct as the first attempt will receive a 401 and the second one will
receive the 200ok. That way the user is not banned as you need 3 attempts to be
banned with the default settings.





From jbonilla at sipwise.com  Fri Mar 15 11:57:06 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Fri, 15 Mar 2013 16:57:06 +0100
Subject: [Spce-user] random registration dropping
In-Reply-To: <2CDACF784B10A14EAAFAB6140DAE0D13243D59@EXCHANGE1.newbreakmanagement.local>
References: <2CDACF784B10A14EAAFAB6140DAE0D132409FC@EXCHANGE1.newbreakmanagement.local>
	<20130313230657.7eb033ac@quenya>
	<AF650DB9-A3FD-44B3-9E52-A38B53103CAB@newbreakcommunications.com>
	<20130314150404.6daf9c05@multivac>
	<2CDACF784B10A14EAAFAB6140DAE0D1324262B@EXCHANGE1.newbreakmanagement.local>
	<20130314190926.3dac1292@quenya>
	<2CDACF784B10A14EAAFAB6140DAE0D13242C28@EXCHANGE1.newbreakmanagement.local>
	<20130314214941.236f9d6f@quenya>
	<2CDACF784B10A14EAAFAB6140DAE0D13243D59@EXCHANGE1.newbreakmanagement.local>
Message-ID: <20130315165706.4854673f@multivac>

El Fri, 15 Mar 2013 15:21:28 +0000
Stephen Donovan <Stephen at newbreakcommunications.com> escribi?:

> I have done the update, 2 days ago.
> 
> I'm finding this issue only with Linksys SPA8000 ATAs.  I swapped a client
> yesterday to a grandstream gxw4024 I had on the shelf, all problems are
> gone.  Other Linksys models are functioning well.  Jeremie Chism may be able
> to chime in with his experiences with the Linksys units as well, his
> customers are on our server as well.  I see the 401 on first attempt, I
> understand that to be the case, what I'm seeing is multiple attempts at every
> registration renewal.  I have the limit increased to 10 before banning so
> they do not get banned now but this still causes a 20 sec to 1 min
> interruption in service at most registration renewals but not all.
> 
> Again, only seeing this on SPA8000 units, nothing else.  I'm deploying all
> new grandstream units going forward and have had no issues with them.
>

Thanks Stephen. Please, keep us updated. In case you don't find anything maybe
you could try to register one of your 8000 to one of my test servers so I could
investigate by myself. Let's wait/see how other devices behave.




From jirka.jirout at mac.com  Fri Mar 15 12:30:51 2013
From: jirka.jirout at mac.com (Jirka Jirout)
Date: Fri, 15 Mar 2013 17:30:51 +0100
Subject: [Spce-user] BYE not correct when sent by called party
In-Reply-To: <mailman.1176.1363363037.2547.spce-user@lists.sipwise.com>
References: <mailman.1176.1363363037.2547.spce-user@lists.sipwise.com>
Message-ID: <7BA44D46-0ECA-4012-8B14-406B5295A4E9@mac.com>

Hello,

> If you have it enabled, please send us the invite sent by the proxy to sems.
> sems is controlled by custom headers sent by the proxy (be careful as they
> contain sensitive info). 
> 
> If the headers are right, enable sems debug and let's see what it says.

here is the INVITE. As far as I can tell, the sst headers are there in P-App-Param:

U 2013/03/15 17:19:37.828094 127.0.0.1:5062 -> 127.0.0.1:5080
INVITE sip:611092 at 10.66.80.74:51522;transport=tls SIP/2.0'
Record-Route: <sip:127.0.0.1:5062;lr=on;ftag=827177332;did=243.d691;mpd=ii;rtpprx=yes;vsf=dU5ZM1lleEZ6TkxNaUpYTlBkZk11TlkzWWV4Rno->'
Record-Route: <sip:127.0.0.1:5888;nat=yes;ngcplb=yes;ftag=827177332;lr=on>'
Record-Route: <sip:SERVER_PUBLIC_IP:5888;nat=yes;ftag=827177332;lr=on>'
Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bK1ec4.a83e4ec5cd343f7fecf75bdf773646e4.0'
Route: <sip:lb at 127.0.0.1:5888;lr;received='sip:62.140.132.74:16292;transport=tls'>'
Via: SIP/2.0/UDP 127.0.0.1:5888;branch=z9hG4bK1ec4.76b4567dd1a238b177841633dc8db86d.0;i=d1'
Via: SIP/2.0/TCP 192.168.178.54:50796;received=CLIENT_PUBLIC_IP;branch=z9hG4bK64699119;rport=50796'
From: <sip:800169 at server.ourdomain.cz>;tag=827177332'
To: <sip:611092 at server.ourdomain.cz>'
Contact: <sip:800169 at 192.168.178.54:50796;alias=CLIENT_PUBLIC_IP~50796~2;transport=tcp>;+g.oma.sip-im;language="en,fr";+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"'
Call-ID: 26257500-8dda-7445-4475-daaf33515075'
CSeq: 847134761 INVITE'
Content-Type: application/sdp'
Content-Length: 734'
Max-Forwards: 31'
Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"'
P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel'
Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER'
P-Access-Network-Info: ADSL;utran-cell-id-3gpp=00000000'
User-Agent: CLIENT_APPLICATION
Supported: 100rel'
P-Called-Party-ID: <sip:611092 at server.ourdomain.cz>'
P-App-Param: ;aleg_sst_enable=yes;aleg_sst_expires=10;aleg_sst_min_timer=9;aleg_sst_max_timer=90;aleg_sst_refresh_method=UPDATE_FALLBACK_INVITE;sst_enable=yes;sst_expires=10;sst_min_timer=9;sst_max_timer=90;sst_refresh_method=UPDATE_FALLBACK_INVITE'
P-App-Name: sbc'
P-D-Uri: sip:lb at 127.0.0.1:5888;lr;received='sip:62.140.132.74:16292;transport=tls'
P-Caller-UUID: 0d7be19d-0565-4c80-995e-4aabca6f93b4'
P-Callee-UUID: 1358f8cc-c4dc-4f59-83dd-caa71267bfb7'

I do not see anything in the log. As a matter of fact, the logging seems to have stopped since i set the log level to 3 like you advised me :-)

regards, jj


From adriavidal at gmail.com  Sun Mar 17 04:18:50 2013
From: adriavidal at gmail.com (=?UTF-8?B?QWRyacOgIFZpZGFs?=)
Date: Sun, 17 Mar 2013 09:18:50 +0100
Subject: [Spce-user] login error Sipwise NGCP Administration Interface
Message-ID: <CAFYw3-duvK2yiPn_WaLOGzQoE58fNO65wu04LO8D3hcZd1yEOA@mail.gmail.com>

I've just installed NGCP Ce on a clean debian server, after install when i
try to login at https://myip:1443
administrator/administrator login is not accepted.

Can i reset or modify it from the console, config file or something else?

-- 
--
Adri? Vidal
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130317/d625a655/attachment-0001.html>

From agranig at sipwise.com  Sun Mar 17 04:42:55 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Sun, 17 Mar 2013 09:42:55 +0100
Subject: [Spce-user] login error Sipwise NGCP Administration Interface
In-Reply-To: <CAFYw3-duvK2yiPn_WaLOGzQoE58fNO65wu04LO8D3hcZd1yEOA@mail.gmail.com>
References: <CAFYw3-duvK2yiPn_WaLOGzQoE58fNO65wu04LO8D3hcZd1yEOA@mail.gmail.com>
Message-ID: <5145820F.9010700@sipwise.com>

Hi,

On 03/17/2013 09:18 AM, Adri? Vidal wrote:
> I've just installed NGCP Ce on a clean debian server, after install when
> i try to login at https://myip:1443
> administrator/administrator login is not accepted.
>
> Can i reset or modify it from the console, config file or something else?

Check /var/log/ngcp/oss.log for any error messages or something obvious. 
Probably MySQL is not running.

Andreas



From lorenzo.mangani at gmail.com  Sun Mar 17 04:52:00 2013
From: lorenzo.mangani at gmail.com (Lorenzo Mangani)
Date: Sun, 17 Mar 2013 09:52:00 +0100
Subject: [Spce-user] login error Sipwise NGCP Administration Interface
In-Reply-To: <CAFYw3-duvK2yiPn_WaLOGzQoE58fNO65wu04LO8D3hcZd1yEOA@mail.gmail.com>
References: <CAFYw3-duvK2yiPn_WaLOGzQoE58fNO65wu04LO8D3hcZd1yEOA@mail.gmail.com>
Message-ID: <CAKN+pHZ2p4a3upNKOJrckrM=7yUzsisz=-zKbcy2eaKLe66-GA@mail.gmail.com>

Adria,

The below query can be used to quickly reset the administrator account
password:

mysql> use billing;
mysql> UPDATE admins SET md5pass = MD5('test') WHERE login =
'administrator';


Best,

Lorenzo Mangani

HOMER DEV TEAM
QXIP - Network Engineering


On Sun, Mar 17, 2013 at 9:18 AM, Adri? Vidal <adriavidal at gmail.com> wrote:

> I've just installed NGCP Ce on a clean debian server, after install when i
> try to login at https://myip:1443
> administrator/administrator login is not accepted.
>
> Can i reset or modify it from the console, config file or something else?
>
> --
> --
> Adri? Vidal
>
>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
>
>


--
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130317/e64c3f57/attachment-0001.html>

From rea1ity at kaist.ac.kr  Sun Mar 17 05:18:11 2013
From: rea1ity at kaist.ac.kr (=?UTF-8?B?7ZeI7Jyk?=)
Date: Sun, 17 Mar 2013 18:18:11 +0900 (KST)
Subject: [Spce-user] =?utf-8?q?Could_you_let_me_know_how_to_registering_PS?=
	=?utf-8?q?TN_gateway_that_I_have=3F?=
Message-ID: <51458af03fe9_@_imoxion.com>

Hi all,
As I refer SIPWISE document it describe how to registering at a peer server same like following;
================================
6.4.4.2. Registering at a Peering Server
Unfortunately, the credentials configured above are not yet automatically used to register the SPCE at your peer hosts. There is however an easy manual way to do so, until this is addressed.

Configure your peering servers with the corresponding credentials in /etc/ngcp-config/templates/etc/sems/etc/reg_agent.conf.tt2, then execute ngcpcfg apply.
=================================
So that I opened the reg_agent.conf.tt2 file and found some could modify part it could register a peer server same like following;
=================================
1. Realm server name,
2. Realm username,
3. Password
=================================
And there have more information that I could modify same like proxy ... Etc.
But I don't use proxy server so that I didn't modified it.
Anyway, I don't really knows what information could I input to there.
It seems it have input about the PSTN gateway server information to there so I modified it as I think.
Also I don't exactly knows what is username and password information of PSTN gateway required the SIPWISE.
So I'm put on the username and password information as web management account of PSTN gateway.
But PSTN gateway register status still 'unregistered'.
Am I missing information about it?
Please let me know how to register with my PSTN gateway.
Is the file reg_agent.conf.tt2 needed to modify if I want to register with my PSTN gateway, right?
If not, please advise how it works.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130317/eafe9881/attachment-0001.html>

From rea1ity at kaist.ac.kr  Sun Mar 17 05:20:39 2013
From: rea1ity at kaist.ac.kr (=?UTF-8?B?7ZeI7Jyk?=)
Date: Sun, 17 Mar 2013 18:20:39 +0900 (KST)
Subject: [Spce-user] =?utf-8?q?Could_you_let_me_know_how_to_registering_PS?=
	=?utf-8?q?TN_gateway_that_I_have=3F?=
Message-ID: <51458b903fe8_@_imoxion.com>

Hi all,
As I refer SIPWISE document it describe how to registering at a peer server same like following;
================================
6.4.4.2. Registering at a Peering Server
Unfortunately, the credentials configured above are not yet automatically used to register the SPCE at your peer hosts. There is however an easy manual way to do so, until this is addressed.

Configure your peering servers with the corresponding credentials in /etc/ngcp-config/templates/etc/sems/etc/reg_agent.conf.tt2, then execute ngcpcfg apply.
=================================
So that I opened the reg_agent.conf.tt2 file and found some could modify part it could register a peer server same like following;
=================================
1. Realm server name,
2. Realm username,
3. Password
=================================
And there have more information that I could modify same like proxy ... Etc.
But I don't use proxy server so that I didn't modified it.
Anyway, I don't really knows what information could I input to there.
It seems it have input about the PSTN gateway server information to there so I modified it as I think.
Also I don't exactly knows what is username and password information of PSTN gateway required the SIPWISE.
So I'm put on the username and password information as web management account of PSTN gateway.
But PSTN gateway register status still 'unregistered'.
Am I missing information about it?
Please let me know how to register with my PSTN gateway.
Is the file reg_agent.conf.tt2 needed to modify if I want to register with my PSTN gateway, right?
If not, please advise how it works.

Sorry, missing my information.

Regards,

David Ho.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130317/7d551322/attachment-0001.html>

From adriavidal at gmail.com  Sun Mar 17 05:24:20 2013
From: adriavidal at gmail.com (=?UTF-8?B?QWRyacOgIFZpZGFs?=)
Date: Sun, 17 Mar 2013 10:24:20 +0100
Subject: [Spce-user] login error Sipwise NGCP Administration Interface
In-Reply-To: <CAKN+pHZ2p4a3upNKOJrckrM=7yUzsisz=-zKbcy2eaKLe66-GA@mail.gmail.com>
References: <CAFYw3-duvK2yiPn_WaLOGzQoE58fNO65wu04LO8D3hcZd1yEOA@mail.gmail.com>
	<CAKN+pHZ2p4a3upNKOJrckrM=7yUzsisz=-zKbcy2eaKLe66-GA@mail.gmail.com>
Message-ID: <CAFYw3-eazoF7HQP+Cz=H_NyOj55z5kRDq5fE7i=s3M9NS_webw@mail.gmail.com>

2013/3/17 Lorenzo Mangani <lorenzo.mangani at gmail.com>

> Adria,
>
> The below query can be used to quickly reset the administrator account
> password:
>
> mysql> use billing;
> mysql> UPDATE admins SET md5pass = MD5('test') WHERE login =
> 'administrator';
>
>
Thanks seeems InnoDB is not running on my Mysql server, will check that.

-- 
--
Adri? Vidal
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130317/e166a94d/attachment-0001.html>

From jbonilla at sipwise.com  Sun Mar 17 05:30:53 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Sun, 17 Mar 2013 10:30:53 +0100
Subject: [Spce-user] login error Sipwise NGCP Administration Interface
In-Reply-To: <CAFYw3-eazoF7HQP+Cz=H_NyOj55z5kRDq5fE7i=s3M9NS_webw@mail.gmail.com>
References: <CAFYw3-duvK2yiPn_WaLOGzQoE58fNO65wu04LO8D3hcZd1yEOA@mail.gmail.com>
	<CAKN+pHZ2p4a3upNKOJrckrM=7yUzsisz=-zKbcy2eaKLe66-GA@mail.gmail.com>
	<CAFYw3-eazoF7HQP+Cz=H_NyOj55z5kRDq5fE7i=s3M9NS_webw@mail.gmail.com>
Message-ID: <20130317103053.296ad77b@multivac>

El Sun, 17 Mar 2013 10:24:20 +0100
Adri? Vidal <adriavidal at gmail.com> escribi?:

> 2013/3/17 Lorenzo Mangani <lorenzo.mangani at gmail.com>
> 
> > Adria,
> >
> > The below query can be used to quickly reset the administrator account
> > password:
> >
> > mysql> use billing;
> > mysql> UPDATE admins SET md5pass = MD5('test') WHERE login =
> > 'administrator';
> >
> >
> Thanks seeems InnoDB is not running on my Mysql server, will check that.
> 

How much mem do you have in your system? 





From adriavidal at gmail.com  Sun Mar 17 06:26:54 2013
From: adriavidal at gmail.com (=?UTF-8?B?QWRyacOgIFZpZGFs?=)
Date: Sun, 17 Mar 2013 11:26:54 +0100
Subject: [Spce-user] login error Sipwise NGCP Administration Interface
In-Reply-To: <20130317103053.296ad77b@multivac>
References: <CAFYw3-duvK2yiPn_WaLOGzQoE58fNO65wu04LO8D3hcZd1yEOA@mail.gmail.com>
	<CAKN+pHZ2p4a3upNKOJrckrM=7yUzsisz=-zKbcy2eaKLe66-GA@mail.gmail.com>
	<CAFYw3-eazoF7HQP+Cz=H_NyOj55z5kRDq5fE7i=s3M9NS_webw@mail.gmail.com>
	<20130317103053.296ad77b@multivac>
Message-ID: <CAFYw3-cptNWU8Z3fd5-e3F=U2Bm54v+X_R+fLvQBwqxWn377Pw@mail.gmail.com>

>
>
> How much mem do you have in your system?
>
>
Is a openvz instance with 512 ram and 512 swap, uppet to 2024 ram 512 swap
to get everything running (apache + mysql).

Will do some tests now.

Thanks

-- 
--
Adri? Vidal
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130317/91209ea1/attachment-0001.html>

From marcial at tecnimoviles.com  Sun Mar 17 11:12:38 2013
From: marcial at tecnimoviles.com (Marcial Barrios)
Date: Sun, 17 Mar 2013 15:12:38 +0000
Subject: [Spce-user] rate-o-mat
Message-ID: <CAMnKu2FSKYtHaD=oMi4z++rf0WUWy-qju8v5VQQQLfUiM_dgTg@mail.gmail.com>

No calls are rated.  How can check if rate-o-mat is working properly?
-- 
*Marcial Barrios*
*Administrador de** Sistemas*
M?vil: (+34) 603 053 043
  Tel:  (+34) 902 955 290
 Fax:  (+34) 822 201 442
email: marcial at tecnimoviles.com
  web: www.tecnimoviles.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130317/35962318/attachment-0001.html>

From marcial at tecnimoviles.com  Sun Mar 17 11:41:53 2013
From: marcial at tecnimoviles.com (Marcial Barrios)
Date: Sun, 17 Mar 2013 15:41:53 +0000
Subject: [Spce-user] rate-o-mat
In-Reply-To: <CAKN+pHZC9q47uxx=i9A8YSY+1TAbzwuF89wzJ2jpv5G2Y0uVkQ@mail.gmail.com>
References: <CAMnKu2FSKYtHaD=oMi4z++rf0WUWy-qju8v5VQQQLfUiM_dgTg@mail.gmail.com>
	<CAKN+pHYLZ9eMD3+tMJw31eFm_Asu-6uSgL6esfjaVZbsUNn7CQ@mail.gmail.com>
	<CAMnKu2FgPde96M2UHX-B970bnLVT8-qA-CYDMOLNwpLGa4oa1A@mail.gmail.com>
	<CAKN+pHZC9q47uxx=i9A8YSY+1TAbzwuF89wzJ2jpv5G2Y0uVkQ@mail.gmail.com>
Message-ID: <CAMnKu2Gg=i=gfMtU=M3LFZ68VpC5XRhuyn8VpLO_D+ccXCbVHw@mail.gmail.com>

I Lorenzo I did what you suggested

#  service ngcp-rate-o-mat start

but this is the error message

rollback ineffective with AutoCommit enabled at /usr/sbin/rate
                                                                -o-mat line
91.
rollback ineffective with AutoCommit enabled at /usr/sbin/rate-o-mat line
92.
Error rating CDR batch: No fee info for profile 2 and user 'testuser_21' or
doma
      in 'testuser_21 at 176.31.119.223' found


2013/3/17 Lorenzo Mangani <lorenzo.mangani at gmail.com>

> I forgot to copy the list but as we wait for other more valuable
> suggestions, try simply:
>
> #  service ngcp-rate-o-mat start
>
> if you get a message about service being disabled, edit in your config.yml
> to enable rate-o-mat
>
>
>
> Lorenzo
>
>
> On Sun, Mar 17, 2013 at 4:28 PM, Marcial Barrios <marcial at tecnimoviles.com
> > wrote:
>
>> Hi Lorenzo, I apply this:
>>
>> # service ngcp-rate-o-mat status
>>
>> and this is the response:
>>
>> # rate-o-mat is not running ... failed!
>>
>>
>> 2013/3/17 Lorenzo Mangani <lorenzo.mangani at gmail.com>
>>
>>> Hi Marcial,
>>>
>>> You can first of all check and confirm if the service is running on your
>>> system:
>>>
>>> # service ngcp-rate-o-mat status
>>>
>>>
>>>
>>>  Lorenzo Mangani
>>>
>>> HOMER DEV TEAM
>>> QXIP - Network Engineering
>>>
>>>
>>> On Sun, Mar 17, 2013 at 4:12 PM, Marcial Barrios <
>>> marcial at tecnimoviles.com> wrote:
>>>
>>>>
>>>> No calls are rated.  How can check if rate-o-mat is working properly?
>>>> --
>>>> *Marcial Barrios*
>>>> *Administrador de** Sistemas*
>>>>  M?vil: (+34) 603 053 043
>>>>   Tel:  (+34) 902 955 290
>>>>  Fax:  (+34) 822 201 442
>>>> email: marcial at tecnimoviles.com
>>>>   web: www.tecnimoviles.com
>>>>
>>>> _______________________________________________
>>>> Spce-user mailing list
>>>> Spce-user at lists.sipwise.com
>>>> http://lists.sipwise.com/listinfo/spce-user
>>>>
>>>>
>>>
>>>
>>> --
>>>
>>>
>>>
>>>
>>
>>
>> --
>> *Marcial Barrios*
>> *Administrador de** Sistemas*
>> M?vil: (+34) 603 053 043
>>   Tel:  (+34) 902 955 290
>>  Fax:  (+34) 822 201 442
>> email: marcial at tecnimoviles.com
>>   web: www.tecnimoviles.com
>>
>
>
>
> --
>
>
> Lorenzo Mangani
>
> HOMER DEV TEAM
> QXIP - Network Engineering
>
>


-- 
*Marcial Barrios*
*Administrador de** Sistemas*
M?vil: (+34) 603 053 043
  Tel:  (+34) 902 955 290
 Fax:  (+34) 822 201 442
email: marcial at tecnimoviles.com
  web: www.tecnimoviles.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130317/f1bbdff8/attachment-0001.html>

From dgrotti at sipwise.com  Sun Mar 17 11:34:59 2013
From: dgrotti at sipwise.com (Daniel Grotti)
Date: Sun, 17 Mar 2013 16:34:59 +0100
Subject: [Spce-user] rate-o-mat
Message-ID: <j75utrfckctuw1qj68s8qy18.1363534499413@email.android.com>

Hi,
please check what /var/log/ngcp/rate-o-mat.log is saying.

Daniel



Sent from Samsung Mobile

Marcial Barrios <marcial at tecnimoviles.com> wrote:


No calls are rated. ?How can check if rate-o-mat is working properly?
-- 
Marcial Barrios
Administrador de?Sistemas

M?vil: (+34) 603 053 043
? Tel: ?(+34) 902 955 290
?Fax: ?(+34) 822 201 442
email: marcial at tecnimoviles.com
? web:?www.tecnimoviles.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130317/faa71c0a/attachment-0001.html>

From marcial at tecnimoviles.com  Sun Mar 17 12:07:26 2013
From: marcial at tecnimoviles.com (Marcial Barrios)
Date: Sun, 17 Mar 2013 16:07:26 +0000
Subject: [Spce-user] rate-o-mat
In-Reply-To: <CAMnKu2FTfpMAVeCQyXR+0wxOGSqCY5W1uPRi1MJ3KVr8SfEPUQ@mail.gmail.com>
References: <j75utrfckctuw1qj68s8qy18.1363534499413@email.android.com>
	<CAMnKu2FTfpMAVeCQyXR+0wxOGSqCY5W1uPRi1MJ3KVr8SfEPUQ@mail.gmail.com>
Message-ID: <CAMnKu2F03uNr-rr69jHV1G+YVjcm3O7ZE1JwA0UcnykqK422fw@mail.gmail.com>

2013/3/17 Marcial Barrios <marcial at tecnimoviles.com>

> Hi, Daniel this is what rate-o-mat.log is saying:
>
> Trying to connect to billing db...
> Successfully connected to billing db...
> Trying to connect to accounting db...
> Successfully connected to accounting db...
> Up and running.
> rate cdr #13
> No reseller info for user id 10d81223-c19c-4030-b2dd-429af4dd8376 found
> Error rating CDR batch: No reseller info for user id
> 10d81223-c19c-4030-b2dd-429af4dd8376 found
>
>
>
> 2013/3/17 Daniel Grotti <dgrotti at sipwise.com>
>
>> Hi,
>> please check what /var/log/ngcp/rate-o-mat.log is saying.
>>
>> Daniel
>>
>>
>>
>> Sent from Samsung Mobile
>>
>>
>>
>> Marcial Barrios <marcial at tecnimoviles.com> wrote:
>>
>>
>>
>> No calls are rated.  How can check if rate-o-mat is working properly?
>> --
>> *Marcial Barrios*
>> *Administrador de** Sistemas*
>> M?vil: (+34) 603 053 043
>>   Tel:  (+34) 902 955 290
>>  Fax:  (+34) 822 201 442
>> email: marcial at tecnimoviles.com
>>   web: www.tecnimoviles.com
>>
>
>
>
> --
> *Marcial Barrios*
> *Administrador de** Sistemas*
> M?vil: (+34) 603 053 043
>   Tel:  (+34) 902 955 290
>  Fax:  (+34) 822 201 442
> email: marcial at tecnimoviles.com
>   web: www.tecnimoviles.com
>



-- 
*Marcial Barrios*
*Administrador de** Sistemas*
M?vil: (+34) 603 053 043
  Tel:  (+34) 902 955 290
 Fax:  (+34) 822 201 442
email: marcial at tecnimoviles.com
  web: www.tecnimoviles.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130317/bfbdeeef/attachment-0001.html>

From jbonilla at sipwise.com  Sun Mar 17 13:09:35 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Sun, 17 Mar 2013 18:09:35 +0100
Subject: [Spce-user] login error Sipwise NGCP Administration Interface
In-Reply-To: <CAFYw3-cptNWU8Z3fd5-e3F=U2Bm54v+X_R+fLvQBwqxWn377Pw@mail.gmail.com>
References: <CAFYw3-duvK2yiPn_WaLOGzQoE58fNO65wu04LO8D3hcZd1yEOA@mail.gmail.com>
	<CAKN+pHZ2p4a3upNKOJrckrM=7yUzsisz=-zKbcy2eaKLe66-GA@mail.gmail.com>
	<CAFYw3-eazoF7HQP+Cz=H_NyOj55z5kRDq5fE7i=s3M9NS_webw@mail.gmail.com>
	<20130317103053.296ad77b@multivac>
	<CAFYw3-cptNWU8Z3fd5-e3F=U2Bm54v+X_R+fLvQBwqxWn377Pw@mail.gmail.com>
Message-ID: <20130317180935.060ff1ad@multivac>

El Sun, 17 Mar 2013 11:26:54 +0100
Adri? Vidal <adriavidal at gmail.com> escribi?:

> >
> >
> > How much mem do you have in your system?
> >
> >
> Is a openvz instance with 512 ram and 512 swap, uppet to 2024 ram 512 swap
> to get everything running (apache + mysql).
> 
> Will do some tests now.
> 
> Thanks
> 

512 ram is too low to run a spce with the default configuration. To lower the
requisites try to play with the following in your config.yml:

- Disable tls if you're not using it
- Disable presence if you're not using it
- Set lb and kamailio number of children to 2 instead of 8
- Lower the buffer_pool_size asigned to mysql to 128MB
- Lower the pkg_mem of the lb to 8
- set the shm mem of proxy and lb to 64


apply the changes and see if everything is running fine.

cheers,

Jon




From kmasse at questblue.com  Sun Mar 17 13:22:57 2013
From: kmasse at questblue.com (Kevin Masse)
Date: Sun, 17 Mar 2013 13:22:57 -0400
Subject: [Spce-user] rate-o-mat
References: <j75utrfckctuw1qj68s8qy18.1363534499413@email.android.com><CAMnKu2FTfpMAVeCQyXR+0wxOGSqCY5W1uPRi1MJ3KVr8SfEPUQ@mail.gmail.com>
	<CAMnKu2F03uNr-rr69jHV1G+YVjcm3O7ZE1JwA0UcnykqK422fw@mail.gmail.com>
Message-ID: <CEEE718799582B47A55D7897DCB264FC28024C@bvusaex.bestvoipusa.local>

Are you using XML or SOAP to update subscribers?  If so it is possible that you have deleted the subscriber before the rating process went over the CDRs.  


The only way to get your rating back running again is to update the accounting.cdr table manually by setting the rating_status to "ok" for source_subscriber_id=10d81223-c19c-4030-b2dd-429af4dd8376

 

 

If you are not using the XML or SOAP to update it still could be possible you deleted a subscriber before calls were rated.

 

This may have to be repeated a few times to get the status "ok" in all fields.

 

Kevin

 

 

 

 

From: spce-user-bounces at lists.sipwise.com [mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of Marcial Barrios
Sent: Sunday, March 17, 2013 12:07 PM
To: Daniel Grotti; spce-user at lists.sipwise.com
Subject: Re: [Spce-user] rate-o-mat

 

 

 

2013/3/17 Marcial Barrios <marcial at tecnimoviles.com>

Hi, Daniel this is what rate-o-mat.log is saying:

 

Trying to connect to billing db...

Successfully connected to billing db...

Trying to connect to accounting db...

Successfully connected to accounting db...

Up and running.

rate cdr #13

No reseller info for user id 10d81223-c19c-4030-b2dd-429af4dd8376 found

Error rating CDR batch: No reseller info for user id 10d81223-c19c-4030-b2dd-429af4dd8376 found

 

 

2013/3/17 Daniel Grotti <dgrotti at sipwise.com>

Hi,

please check what /var/log/ngcp/rate-o-mat.log is saying.

 

Daniel




Sent from Samsung Mobile 




Marcial Barrios <marcial at tecnimoviles.com> wrote:



 

No calls are rated.  How can check if rate-o-mat is working properly?

-- 

Marcial Barrios

Administrador de Sistemas

M?vil: (+34) 603 053 043 <tel:%28%2B34%29%20603%20053%20043> 

  Tel:  (+34) 902 955 290 <tel:%28%2B34%29%20902%20955%20290> 

 Fax:  (+34) 822 201 442 <tel:%28%2B34%29%20822%20201%20442> 

email: marcial at tecnimoviles.com

  web: www.tecnimoviles.com <http://www.tecnimoviles.com/> 





 

-- 

Marcial Barrios

Administrador de Sistemas

 <http://tecnimoviles.com/wp-content/uploads/2012/10/cropped-cropped-headertecnimoviles1-e1351371092469.png> 

M?vil: (+34) 603 053 043 <tel:%28%2B34%29%20603%20053%20043> 

  Tel:  (+34) 902 955 290 <tel:%28%2B34%29%20902%20955%20290> 

 Fax:  (+34) 822 201 442 <tel:%28%2B34%29%20822%20201%20442> 

email: marcial at tecnimoviles.com

  web: www.tecnimoviles.com





 

-- 

Marcial Barrios

Administrador de Sistemas

 <http://tecnimoviles.com/wp-content/uploads/2012/10/cropped-cropped-headertecnimoviles1-e1351371092469.png> 

M?vil: (+34) 603 053 043 <tel:%28%2B34%29%20603%20053%20043> 

  Tel:  (+34) 902 955 290 <tel:%28%2B34%29%20902%20955%20290> 

 Fax:  (+34) 822 201 442 <tel:%28%2B34%29%20822%20201%20442> 

email: marcial at tecnimoviles.com

  web: www.tecnimoviles.com

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130317/d553e818/attachment-0001.html>

From marcial at tecnimoviles.com  Sun Mar 17 16:09:20 2013
From: marcial at tecnimoviles.com (Marcial Barrios)
Date: Sun, 17 Mar 2013 20:09:20 +0000
Subject: [Spce-user] rate-o-mat
In-Reply-To: <CEEE718799582B47A55D7897DCB264FC28024C@bvusaex.bestvoipusa.local>
References: <j75utrfckctuw1qj68s8qy18.1363534499413@email.android.com>
	<CAMnKu2FTfpMAVeCQyXR+0wxOGSqCY5W1uPRi1MJ3KVr8SfEPUQ@mail.gmail.com>
	<CAMnKu2F03uNr-rr69jHV1G+YVjcm3O7ZE1JwA0UcnykqK422fw@mail.gmail.com>
	<CEEE718799582B47A55D7897DCB264FC28024C@bvusaex.bestvoipusa.local>
Message-ID: <CAMnKu2Gp40A3KXkoVzUbow29tS1f0e=OW3aV0tfxdenUFMxqVg@mail.gmail.com>

Hi Kevin, I do the following and works

update cdr set rating_status ="ok" where
 source_user_id="10d81223-c19c-4030-b2dd-429af4dd8376"

but when I tried to

#service ngcp-rate-o-mat start

I have this other message:

rollback ineffective with AutoCommit enabled at /usr/sbin/rate-o-mat line
91.
rollback ineffective with AutoCommit enabled at /usr/sbin/rate-o-mat line
92.
Error rating CDR batch: No fee info for profile 2 and user '0034922388318'
or domain '0034922388318 at 91.192.110.234' found





2013/3/17 Kevin Masse <kmasse at questblue.com>

> Are you using XML or SOAP to update subscribers?  If so it is possible
> that you have deleted the subscriber before the rating process went over
> the CDRs.  ****
>
>
> The only way to get your rating back running again is to update the
> accounting.cdr table manually by setting the rating_status to ?ok? for
> source_subscriber_id=10d81223-c19c-4030-b2dd-429af4dd8376****
>
> ** **
>
> ** **
>
> If you are not using the XML or SOAP to update it still could be possible
> you deleted a subscriber before calls were rated.****
>
> ** **
>
> This may have to be repeated a few times to get the status ?ok? in all
> fields.****
>
> ** **
>
> Kevin****
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>
> *From:* spce-user-bounces at lists.sipwise.com [mailto:
> spce-user-bounces at lists.sipwise.com] *On Behalf Of *Marcial Barrios
> *Sent:* Sunday, March 17, 2013 12:07 PM
> *To:* Daniel Grotti; spce-user at lists.sipwise.com
> *Subject:* Re: [Spce-user] rate-o-mat****
>
> ** **
>
> ** **
>
> ** **
>
> 2013/3/17 Marcial Barrios <marcial at tecnimoviles.com>****
>
> Hi, Daniel this is what rate-o-mat.log is saying:****
>
> ** **
>
> Trying to connect to billing db...****
>
> Successfully connected to billing db...****
>
> Trying to connect to accounting db...****
>
> Successfully connected to accounting db...****
>
> Up and running.****
>
> rate cdr #13****
>
> No reseller info for user id 10d81223-c19c-4030-b2dd-429af4dd8376 found***
> *
>
> Error rating CDR batch: No reseller info for user id
> 10d81223-c19c-4030-b2dd-429af4dd8376 found****
>
> ** **
>
> ** **
>
> 2013/3/17 Daniel Grotti <dgrotti at sipwise.com>****
>
> Hi,****
>
> please check what /var/log/ngcp/rate-o-mat.log is saying.****
>
> ** **
>
> Daniel****
>
>
>
>
> Sent from Samsung Mobile ****
>
>
>
>
> Marcial Barrios <marcial at tecnimoviles.com> wrote:
>
> ****
>
> ** **
>
> No calls are rated.  How can check if rate-o-mat is working properly?****
>
> -- ****
>
> *Marcial Barrios*****
>
> *Administrador de Sistemas*****
>
> ****
>
> M?vil: (+34) 603 053 043****
>
>   Tel:  (+34) 902 955 290****
>
>  Fax:  (+34) 822 201 442****
>
> email: marcial at tecnimoviles.com****
>
>   web: www.tecnimoviles.com****
>
>
>
> ****
>
> ** **
>
> -- ****
>
> *Marcial Barrios*****
>
> *Administrador de Sistemas*****
>
> ****
>
> M?vil: (+34) 603 053 043****
>
>   Tel:  (+34) 902 955 290****
>
>  Fax:  (+34) 822 201 442****
>
> email: marcial at tecnimoviles.com****
>
>   web: www.tecnimoviles.com****
>
>
>
> ****
>
> ** **
>
> -- ****
>
> *Marcial Barrios*****
>
> *Administrador de Sistemas*****
>
> ****
>
> M?vil: (+34) 603 053 043****
>
>   Tel:  (+34) 902 955 290****
>
>  Fax:  (+34) 822 201 442****
>
> email: marcial at tecnimoviles.com****
>
>   web: www.tecnimoviles.com****
>



-- 
*Marcial Barrios*
*Administrador de** Sistemas*
M?vil: (+34) 603 053 043
  Tel:  (+34) 902 955 290
 Fax:  (+34) 822 201 442
email: marcial at tecnimoviles.com
  web: www.tecnimoviles.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130317/052fa8a1/attachment-0001.html>

From dgrotti at sipwise.com  Sun Mar 17 16:18:50 2013
From: dgrotti at sipwise.com (Daniel Grotti)
Date: Sun, 17 Mar 2013 21:18:50 +0100
Subject: [Spce-user] rate-o-mat
Message-ID: <1xtygsph2mi02o3q5h6d2hy5.1363551530821@email.android.com>

Hi,
this means that you don't have any billing fees for calls to 0034*.
Please add a proper billing fees and restart rate-o-mat.

Daniel



Sent from Samsung Mobile

Marcial Barrios <marcial at tecnimoviles.com> wrote:

Hi Kevin, I do the following and works?

update cdr set rating_status ="ok" where ?source_user_id="10d81223-c19c-4030-b2dd-429af4dd8376"

but when I tried to?

#service ngcp-rate-o-mat start

I have this other message:

rollback ineffective with AutoCommit enabled at /usr/sbin/rate-o-mat line 91.
rollback ineffective with AutoCommit enabled at /usr/sbin/rate-o-mat line 92.
Error rating CDR batch: No fee info for profile 2 and user '0034922388318' or domain '0034922388318 at 91.192.110.234' found





2013/3/17 Kevin Masse <kmasse at questblue.com>
Are you using XML or SOAP to update subscribers?? If so it is possible that you have deleted the subscriber before the rating process went over the CDRs.?


The only way to get your rating back running again is to update the accounting.cdr table manually by setting the rating_status to ?ok? for source_subscriber_id=10d81223-c19c-4030-b2dd-429af4dd8376

?

?

If you are not using the XML or SOAP to update it still could be possible you deleted a subscriber before calls were rated.

?

This may have to be repeated a few times to get the status ?ok? in all fields.

?

Kevin

?

?

?

?

From: spce-user-bounces at lists.sipwise.com [mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of Marcial Barrios
Sent: Sunday, March 17, 2013 12:07 PM
To: Daniel Grotti; spce-user at lists.sipwise.com
Subject: Re: [Spce-user] rate-o-mat

?

?

?

2013/3/17 Marcial Barrios <marcial at tecnimoviles.com>

Hi, Daniel this is what rate-o-mat.log is saying:

?

Trying to connect to billing db...

Successfully connected to billing db...

Trying to connect to accounting db...

Successfully connected to accounting db...

Up and running.

rate cdr #13

No reseller info for user id 10d81223-c19c-4030-b2dd-429af4dd8376 found

Error rating CDR batch: No reseller info for user id 10d81223-c19c-4030-b2dd-429af4dd8376 found

?

?

2013/3/17 Daniel Grotti <dgrotti at sipwise.com>

Hi,

please check what /var/log/ngcp/rate-o-mat.log is saying.

?

Daniel




Sent from Samsung Mobile




Marcial Barrios <marcial at tecnimoviles.com> wrote:


?

No calls are rated. ?How can check if rate-o-mat is working properly?

--

Marcial Barrios

Administrador de?Sistemas


M?vil: (+34) 603 053 043

? Tel: ?(+34) 902 955 290

?Fax: ?(+34) 822 201 442

email: marcial at tecnimoviles.com

? web:?www.tecnimoviles.com




?

--

Marcial Barrios

Administrador de?Sistemas



M?vil: (+34) 603 053 043

? Tel: ?(+34) 902 955 290

?Fax: ?(+34) 822 201 442

email: marcial at tecnimoviles.com

? web:?www.tecnimoviles.com




?

--

Marcial Barrios

Administrador de?Sistemas



M?vil: (+34) 603 053 043

? Tel: ?(+34) 902 955 290

?Fax: ?(+34) 822 201 442

email: marcial at tecnimoviles.com

? web:?www.tecnimoviles.com




-- 
Marcial Barrios
Administrador de?Sistemas

M?vil: (+34) 603 053 043
? Tel: ?(+34) 902 955 290
?Fax: ?(+34) 822 201 442
email: marcial at tecnimoviles.com
? web:?www.tecnimoviles.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130317/1a056e0b/attachment-0001.html>

From vinceoscar at gmail.com  Sun Mar 17 17:05:38 2013
From: vinceoscar at gmail.com (Vince Oscar)
Date: Sun, 17 Mar 2013 17:05:38 -0400
Subject: [Spce-user] setup pstn gateway in and out
Message-ID: <CABAd6xvQi5hsdtQYgRdbN8EyQ68m=hGMPfxPubABy+r3rkwNkQ@mail.gmail.com>

I would like to setup gateways to pstn also to freeswitch and asterisk.
How would I do this? Where would I find the documentation.

I have tried to setup peering but with failure.  I am surely doing things
wrong.  Also what log files should I be checking to observe
success/failures.

Does a default install include all modules to do this?

Please advise

Thank you
Vince
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130317/54ef25cd/attachment-0001.html>

From marcial at tecnimoviles.com  Sun Mar 17 17:07:10 2013
From: marcial at tecnimoviles.com (Marcial Barrios)
Date: Sun, 17 Mar 2013 21:07:10 +0000
Subject: [Spce-user] rate-o-mat
In-Reply-To: <1xtygsph2mi02o3q5h6d2hy5.1363551530821@email.android.com>
References: <1xtygsph2mi02o3q5h6d2hy5.1363551530821@email.android.com>
Message-ID: <CAMnKu2FAYhoKUvvLiLLEbo+HEYYctM4HoPOpy+1b0VEisd73XA@mail.gmail.com>

Yes you were right I add the billing fees and work but this is like the
story that never end. This is the new error message:

 #rollback ineffective with AutoCommit enabled at /usr/sbin/rate-o-mat line
91.
rollback ineffective with AutoCommit enabled at /usr/sbin/rate-o-mat line
92.
Error rating CDR batch: No reseller info for user id
909d45f8-1b67-4b44-9655-27a8ccd755d5 found


2013/3/17 Daniel Grotti <dgrotti at sipwise.com>

> Hi,
> this means that you don't have any billing fees for calls to 0034*.
> Please add a proper billing fees and restart rate-o-mat.
>
> Daniel
>
>
>
> Sent from Samsung Mobile
>
>
>
> Marcial Barrios <marcial at tecnimoviles.com> wrote:
>
>
> Hi Kevin, I do the following and works
>
> update cdr set rating_status ="ok" where
>  source_user_id="10d81223-c19c-4030-b2dd-429af4dd8376"
>
> but when I tried to
>
> #service ngcp-rate-o-mat start
>
> I have this other message:
>
> rollback ineffective with AutoCommit enabled at /usr/sbin/rate-o-mat line
> 91.
> rollback ineffective with AutoCommit enabled at /usr/sbin/rate-o-mat line
> 92.
> Error rating CDR batch: No fee info for profile 2 and user '0034922388318'
> or domain '0034922388318 at 91.192.110.234' found
>
>
>
>
>
> 2013/3/17 Kevin Masse <kmasse at questblue.com>
>
>> Are you using XML or SOAP to update subscribers?  If so it is possible
>> that you have deleted the subscriber before the rating process went over
>> the CDRs.  ****
>>
>>
>> The only way to get your rating back running again is to update the
>> accounting.cdr table manually by setting the rating_status to ?ok? for
>> source_subscriber_id=10d81223-c19c-4030-b2dd-429af4dd8376****
>>
>> ** **
>>
>> ** **
>>
>> If you are not using the XML or SOAP to update it still could be possible
>> you deleted a subscriber before calls were rated.****
>>
>> ** **
>>
>> This may have to be repeated a few times to get the status ?ok? in all
>> fields.****
>>
>> ** **
>>
>> Kevin****
>>
>> ** **
>>
>> ** **
>>
>> ** **
>>
>> ** **
>>
>> *From:* spce-user-bounces at lists.sipwise.com [mailto:
>> spce-user-bounces at lists.sipwise.com] *On Behalf Of *Marcial Barrios
>> *Sent:* Sunday, March 17, 2013 12:07 PM
>> *To:* Daniel Grotti; spce-user at lists.sipwise.com
>> *Subject:* Re: [Spce-user] rate-o-mat****
>>
>> ** **
>>
>> ** **
>>
>> ** **
>>
>> 2013/3/17 Marcial Barrios <marcial at tecnimoviles.com>****
>>
>> Hi, Daniel this is what rate-o-mat.log is saying:****
>>
>> ** **
>>
>> Trying to connect to billing db...****
>>
>> Successfully connected to billing db...****
>>
>> Trying to connect to accounting db...****
>>
>> Successfully connected to accounting db...****
>>
>> Up and running.****
>>
>> rate cdr #13****
>>
>> No reseller info for user id 10d81223-c19c-4030-b2dd-429af4dd8376 found**
>> **
>>
>> Error rating CDR batch: No reseller info for user id
>> 10d81223-c19c-4030-b2dd-429af4dd8376 found****
>>
>> ** **
>>
>> ** **
>>
>> 2013/3/17 Daniel Grotti <dgrotti at sipwise.com>****
>>
>> Hi,****
>>
>> please check what /var/log/ngcp/rate-o-mat.log is saying.****
>>
>> ** **
>>
>> Daniel****
>>
>>
>>
>>
>> Sent from Samsung Mobile ****
>>
>>
>>
>>
>> Marcial Barrios <marcial at tecnimoviles.com> wrote:
>>
>> ****
>>
>> ** **
>>
>> No calls are rated.  How can check if rate-o-mat is working properly?****
>>
>> -- ****
>>
>> *Marcial Barrios*****
>>
>> *Administrador de Sistemas*****
>>
>> ****
>>
>> M?vil: (+34) 603 053 043****
>>
>>   Tel:  (+34) 902 955 290****
>>
>>  Fax:  (+34) 822 201 442****
>>
>> email: marcial at tecnimoviles.com****
>>
>>   web: www.tecnimoviles.com****
>>
>>
>>
>> ****
>>
>> ** **
>>
>> -- ****
>>
>> *Marcial Barrios*****
>>
>> *Administrador de Sistemas*****
>>
>> ****
>>
>> M?vil: (+34) 603 053 043****
>>
>>   Tel:  (+34) 902 955 290****
>>
>>  Fax:  (+34) 822 201 442****
>>
>> email: marcial at tecnimoviles.com****
>>
>>   web: www.tecnimoviles.com****
>>
>>
>>
>> ****
>>
>> ** **
>>
>> -- ****
>>
>> *Marcial Barrios*****
>>
>> *Administrador de Sistemas*****
>>
>> ****
>>
>> M?vil: (+34) 603 053 043****
>>
>>   Tel:  (+34) 902 955 290****
>>
>>  Fax:  (+34) 822 201 442****
>>
>> email: marcial at tecnimoviles.com****
>>
>>   web: www.tecnimoviles.com****
>>
>
>
>
> --
> *Marcial Barrios*
> *Administrador de** Sistemas*
> M?vil: (+34) 603 053 043
>   Tel:  (+34) 902 955 290
>  Fax:  (+34) 822 201 442
> email: marcial at tecnimoviles.com
>   web: www.tecnimoviles.com
>



-- 
*Marcial Barrios*
*Administrador de** Sistemas*
M?vil: (+34) 603 053 043
  Tel:  (+34) 902 955 290
 Fax:  (+34) 822 201 442
email: marcial at tecnimoviles.com
  web: www.tecnimoviles.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130317/aae1e1d4/attachment-0001.html>

From jbonilla at sipwise.com  Sun Mar 17 17:51:19 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Sun, 17 Mar 2013 22:51:19 +0100
Subject: [Spce-user] rate-o-mat
In-Reply-To: <CAMnKu2FAYhoKUvvLiLLEbo+HEYYctM4HoPOpy+1b0VEisd73XA@mail.gmail.com>
References: <1xtygsph2mi02o3q5h6d2hy5.1363551530821@email.android.com>
	<CAMnKu2FAYhoKUvvLiLLEbo+HEYYctM4HoPOpy+1b0VEisd73XA@mail.gmail.com>
Message-ID: <20130317225119.70b3b605@quenya>

El Sun, 17 Mar 2013 21:07:10 +0000
Marcial Barrios <marcial at tecnimoviles.com> escribi?:

> Yes you were right I add the billing fees and work but this is like the
> story that never end. This is the new error message:
> 
>  #rollback ineffective with AutoCommit enabled at /usr/sbin/rate-o-mat line
> 91.
> rollback ineffective with AutoCommit enabled at /usr/sbin/rate-o-mat line
> 92.
> Error rating CDR batch: No reseller info for user id
> 909d45f8-1b67-4b44-9655-27a8ccd755d5 found
> 
> 

You can alway use a "match everything" approach. It's normal that you hav all
the destination your customers are using in your billing profile. 

Don't delete the subscribers before the accounting has been finished. You can
terminate them without deleting them.

You can always disable rating and get plain cdrs if you're not interested in
rating.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130317/c1a540a9/attachment-0001.asc>

From jbonilla at sipwise.com  Sun Mar 17 17:53:05 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Sun, 17 Mar 2013 22:53:05 +0100
Subject: [Spce-user] setup pstn gateway in and out
In-Reply-To: <CABAd6xvQi5hsdtQYgRdbN8EyQ68m=hGMPfxPubABy+r3rkwNkQ@mail.gmail.com>
References: <CABAd6xvQi5hsdtQYgRdbN8EyQ68m=hGMPfxPubABy+r3rkwNkQ@mail.gmail.com>
Message-ID: <20130317225305.60ecf372@quenya>

El Sun, 17 Mar 2013 17:05:38 -0400
Vince Oscar <vinceoscar at gmail.com> escribi?:

> I would like to setup gateways to pstn also to freeswitch and asterisk.
> How would I do this? Where would I find the documentation.
> 
> I have tried to setup peering but with failure.  I am surely doing things
> wrong.  Also what log files should I be checking to observe
> success/failures.
> 
> Does a default install include all modules to do this?
> 
> Please advise
> 
> Thank you
> Vince


Yes. The spce is a full platform and you indeed add as many peering groups with
as many billing profiles as you want.

Please detail what you did and what you expected. The peering part is quite
easy and it's present in the handbook.


cheers,

Jon

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130317/87835360/attachment-0001.asc>

From marcial at tecnimoviles.com  Sun Mar 17 18:22:43 2013
From: marcial at tecnimoviles.com (Marcial Barrios)
Date: Sun, 17 Mar 2013 22:22:43 +0000
Subject: [Spce-user] rate-o-mat
In-Reply-To: <CAMnKu2FAYhoKUvvLiLLEbo+HEYYctM4HoPOpy+1b0VEisd73XA@mail.gmail.com>
References: <1xtygsph2mi02o3q5h6d2hy5.1363551530821@email.android.com>
	<CAMnKu2FAYhoKUvvLiLLEbo+HEYYctM4HoPOpy+1b0VEisd73XA@mail.gmail.com>
Message-ID: <CAMnKu2Hvh9wheoWc8sRA6nB22fwsvO6SrKCxuN0_dKifXTaLdw@mail.gmail.com>

Thanks to all, now is working rate-o-mat and the system is billing.

Resume:

Service rate-o-mat stopped because some users were eliminated with calls
not billed (they had not been associated with any fee).

The solution was to manually delete records from MySql database "accounting"
that present problems.

example:

#service ngcp-rate-o-mat start
#rollback ineffective with AutoCommit enabled at /usr/sbin/rate-o-mat line
91. rollback ineffective with AutoCommit enabled at /usr/sbin/rate-o-mat
line 92. Error rating CDR batch: No reseller info for user id
909d45f8-1b67-4b44-9655-27a8ccd755d5
found

MySql query:
delete from cdr where source_user_id="909d45f8-1b67-4b44-9655-27a8ccd755d5"





2013/3/17 Marcial Barrios <marcial at tecnimoviles.com>

> Yes you were right I add the billing fees and work but this is like the
> story that never end. This is the new error message:
>
>  #rollback ineffective with AutoCommit enabled at /usr/sbin/rate-o-mat
> line 91.
>  rollback ineffective with AutoCommit enabled at /usr/sbin/rate-o-mat
> line 92.
> Error rating CDR batch: No reseller info for user id
> 909d45f8-1b67-4b44-9655-27a8ccd755d5 found
>
>
> 2013/3/17 Daniel Grotti <dgrotti at sipwise.com>
>
>> Hi,
>> this means that you don't have any billing fees for calls to 0034*.
>> Please add a proper billing fees and restart rate-o-mat.
>>
>> Daniel
>>
>>
>>
>> Sent from Samsung Mobile
>>
>>
>>
>> Marcial Barrios <marcial at tecnimoviles.com> wrote:
>>
>>
>> Hi Kevin, I do the following and works
>>
>> update cdr set rating_status ="ok" where
>>  source_user_id="10d81223-c19c-4030-b2dd-429af4dd8376"
>>
>> but when I tried to
>>
>> #service ngcp-rate-o-mat start
>>
>> I have this other message:
>>
>> rollback ineffective with AutoCommit enabled at /usr/sbin/rate-o-mat line
>> 91.
>> rollback ineffective with AutoCommit enabled at /usr/sbin/rate-o-mat line
>> 92.
>> Error rating CDR batch: No fee info for profile 2 and user '0034922388318'
>> or domain '0034922388318 at 91.192.110.234' found
>>
>>
>>
>>
>>
>> 2013/3/17 Kevin Masse <kmasse at questblue.com>
>>
>>> Are you using XML or SOAP to update subscribers?  If so it is possible
>>> that you have deleted the subscriber before the rating process went over
>>> the CDRs.  ****
>>>
>>>
>>> The only way to get your rating back running again is to update the
>>> accounting.cdr table manually by setting the rating_status to ?ok? for
>>> source_subscriber_id=10d81223-c19c-4030-b2dd-429af4dd8376****
>>>
>>> ** **
>>>
>>> ** **
>>>
>>> If you are not using the XML or SOAP to update it still could be
>>> possible you deleted a subscriber before calls were rated.****
>>>
>>> ** **
>>>
>>> This may have to be repeated a few times to get the status ?ok? in all
>>> fields.****
>>>
>>> ** **
>>>
>>> Kevin****
>>>
>>> ** **
>>>
>>> ** **
>>>
>>> ** **
>>>
>>> ** **
>>>
>>> *From:* spce-user-bounces at lists.sipwise.com [mailto:
>>> spce-user-bounces at lists.sipwise.com] *On Behalf Of *Marcial Barrios
>>> *Sent:* Sunday, March 17, 2013 12:07 PM
>>> *To:* Daniel Grotti; spce-user at lists.sipwise.com
>>> *Subject:* Re: [Spce-user] rate-o-mat****
>>>
>>> ** **
>>>
>>> ** **
>>>
>>> ** **
>>>
>>> 2013/3/17 Marcial Barrios <marcial at tecnimoviles.com>****
>>>
>>> Hi, Daniel this is what rate-o-mat.log is saying:****
>>>
>>> ** **
>>>
>>> Trying to connect to billing db...****
>>>
>>> Successfully connected to billing db...****
>>>
>>> Trying to connect to accounting db...****
>>>
>>> Successfully connected to accounting db...****
>>>
>>> Up and running.****
>>>
>>> rate cdr #13****
>>>
>>> No reseller info for user id 10d81223-c19c-4030-b2dd-429af4dd8376 found*
>>> ***
>>>
>>> Error rating CDR batch: No reseller info for user id
>>> 10d81223-c19c-4030-b2dd-429af4dd8376 found****
>>>
>>> ** **
>>>
>>> ** **
>>>
>>> 2013/3/17 Daniel Grotti <dgrotti at sipwise.com>****
>>>
>>> Hi,****
>>>
>>> please check what /var/log/ngcp/rate-o-mat.log is saying.****
>>>
>>> ** **
>>>
>>> Daniel****
>>>
>>>
>>>
>>>
>>> Sent from Samsung Mobile ****
>>>
>>>
>>>
>>>
>>> Marcial Barrios <marcial at tecnimoviles.com> wrote:
>>>
>>> ****
>>>
>>> ** **
>>>
>>> No calls are rated.  How can check if rate-o-mat is working properly?***
>>> *
>>>
>>> -- ****
>>>
>>> *Marcial Barrios*****
>>>
>>> *Administrador de Sistemas*****
>>>
>>> ****
>>>
>>> M?vil: (+34) 603 053 043****
>>>
>>>   Tel:  (+34) 902 955 290****
>>>
>>>  Fax:  (+34) 822 201 442****
>>>
>>> email: marcial at tecnimoviles.com****
>>>
>>>   web: www.tecnimoviles.com****
>>>
>>>
>>>
>>> ****
>>>
>>> ** **
>>>
>>> -- ****
>>>
>>> *Marcial Barrios*****
>>>
>>> *Administrador de Sistemas*****
>>>
>>> ****
>>>
>>> M?vil: (+34) 603 053 043****
>>>
>>>   Tel:  (+34) 902 955 290****
>>>
>>>  Fax:  (+34) 822 201 442****
>>>
>>> email: marcial at tecnimoviles.com****
>>>
>>>   web: www.tecnimoviles.com****
>>>
>>>
>>>
>>> ****
>>>
>>> ** **
>>>
>>> -- ****
>>>
>>> *Marcial Barrios*****
>>>
>>> *Administrador de Sistemas*****
>>>
>>> ****
>>>
>>> M?vil: (+34) 603 053 043****
>>>
>>>   Tel:  (+34) 902 955 290****
>>>
>>>  Fax:  (+34) 822 201 442****
>>>
>>> email: marcial at tecnimoviles.com****
>>>
>>>   web: www.tecnimoviles.com****
>>>
>>
>>
>>
>>  --
>> *Marcial Barrios*
>> *Administrador de** Sistemas*
>> M?vil: (+34) 603 053 043
>>   Tel:  (+34) 902 955 290
>>  Fax:  (+34) 822 201 442
>> email: marcial at tecnimoviles.com
>>   web: www.tecnimoviles.com
>>
>
>
>
> --
> *Marcial Barrios*
> *Administrador de** Sistemas*
> M?vil: (+34) 603 053 043
>   Tel:  (+34) 902 955 290
>  Fax:  (+34) 822 201 442
> email: marcial at tecnimoviles.com
>   web: www.tecnimoviles.com
>



-- 
*Marcial Barrios*
*Administrador de** Sistemas*
M?vil: (+34) 603 053 043
  Tel:  (+34) 902 955 290
 Fax:  (+34) 822 201 442
email: marcial at tecnimoviles.com
  web: www.tecnimoviles.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130317/11e2bc0f/attachment-0001.html>

From john.murray at skyracktelecom.com  Mon Mar 18 05:33:35 2013
From: john.murray at skyracktelecom.com (John Murray)
Date: Mon, 18 Mar 2013 09:33:35 -0000
Subject: [Spce-user] Send peer inbound call straight to voicemail
In-Reply-To: <20130315102630.32c7a024@quenya>
References: <021301ce1b80$79b93db0$6d2bb910$@murray@skyracktelecom.com>
	<20130315102630.32c7a024@quenya>
Message-ID: <5146df74.c95fb40a.075c.16e1@mx.google.com>

Jon,

The problem initially to be authentication.

The PBX is actually a gateway and does not register with SPCE. I need to give each gateway subscriber an SPCE subscriber so I use the trusted source model.
IP address of gateway + userpart of From: field matches the subscriber for them to make calls.
For calls to the subscriber I have a static contact/location pointing at the gateway.

However when a call is put out to the subscriber on the gateway and they are not available a call comes in from the gateway to SPCE with 'From:' the original caller 'To:' the voicemail number and 'Divert:' the subscriber details.

This call is rejected with '407 Proxy Authentication Required' as there is no PSTN peer and no trusted source matching the 'From:' because the From: is the originating caller in the PSTN.

I cannot add a peer with the gateway's address as in the logic this is checked before trusted sources and non of my subscribers are then matched.

Would it not be better to check trusted sources first?

Many thanks for your help.

Regards

John

-----Original Message-----
From: spce-user-bounces at lists.sipwise.com [mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of Jon Bonilla (Manwe)
Sent: 15 March 2013 09:27
To: spce-user at lists.sipwise.com
Subject: Re: [Spce-user] Send peer inbound call straight to voicemail

El Thu, 7 Mar 2013 22:09:44 -0000
"John Murray" <john.murray at skyracktelecom.com> escribi?:

> Hi,
> 
>  
> 
> I want to send a specific call type to voicemail.
> 
>  
> 
> I send a call out to a PBX peer who has no voicemail. If the handset 
> is not answered I get a call from the peer with RURI 500 which is the 
> voicemail number, From is the original caller and a Divert: header 
> with the subscriber ddi.
> 
>  
> 
> I know how to take off the Divert header and get the subscriber ddi 
> but what would I put in the RURI to send straight to the subscribers voicemail?
> 
> I've tried prefixing 'vmu' to the ddi but that didn't work.
> 
>  

what was the ngrep-sip (including port 5062), the kamailio-proxy log and the asterisk console output of that call? 





From vinceoscar at gmail.com  Mon Mar 18 10:10:15 2013
From: vinceoscar at gmail.com (Vince Oscar)
Date: Mon, 18 Mar 2013 10:10:15 -0400
Subject: [Spce-user] peer registration fails
Message-ID: <CABAd6xtS0edr_mti=+AKpWHjCarHWA39o7SJ6_R9rOyRdkhmtA@mail.gmail.com>

I cannot get a peer server to PSTN to register.
Server settings are correct with x.x.x.x:5080
5080 is the correct required port

I think the peering rules are incorrect.  First who is outbound, caller or
callee.  I am confused who is who.

second I am confused how to form the peering rules.  for example how would
I allow all incoming, but only allow 10 digit (ac = 3 and sn=7) and 411 and
911

Thank you
Vince
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130318/48928ba7/attachment-0001.html>

From gavin.sweet at skyracktelecom.com  Mon Mar 18 17:24:45 2013
From: gavin.sweet at skyracktelecom.com (Gavin Sweet)
Date: Mon, 18 Mar 2013 21:24:45 -0000
Subject: [Spce-user] CSC www UI layout/customisation
Message-ID: <0ae101ce241f$0371d160$0a557420$@sweet@skyracktelecom.com>

Hi everyone - 

 

I can see from config.yml that some appearance/branding options are
available and that tabs in the view can be turned on/off (e.g.
calllist=1/0).

These options seem to be: account,   addressbook,  callblock, callforward,
calllist, desktop, device, reminder, voicebox

I was wondering, is there a deeper level of modification that can be done as
configuration, or does that require modification to code?

For instance, sub-components on pages, say, to show call list but not
charges.

If there is more, is there a quick "how-to" that someone is able to post to
extend the documentation regarding CSC admin config?

 

 

Many thanks,

Gavin

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130318/fae8aa0f/attachment-0001.html>

From joan at megatro.com  Tue Mar 19 06:13:30 2013
From: joan at megatro.com (Joan Jou)
Date: Tue, 19 Mar 2013 11:13:30 +0100
Subject: [Spce-user] voicemail
Message-ID: <51483A4A.3020809@megatro.com>

Hi,
How can , activate/ desactivate voicemail from user interface and/ admin 
interface /or from user phone?

Thank you
-- 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130319/949181af/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signatura1.gif
Type: image/gif
Size: 13564 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130319/949181af/attachment-0001.gif>

From rea1ity at kaist.ac.kr  Tue Mar 19 06:17:36 2013
From: rea1ity at kaist.ac.kr (=?UTF-8?B?RGF2aWQgSG8=?=)
Date: Tue, 19 Mar 2013 19:17:36 +0900 (KST)
Subject: [Spce-user] =?utf-8?q?Please_let_me_know_why_I_faced_401/403=28Un?=
	=?utf-8?q?authorized/Invalid_To-URI_Detected=29=2E?=
Message-ID: <51483ef63fea_@_imoxion.com>


p{margin:0;padding:0;}


Hi all,

I tried to register my PSTN Gateway with SIPWISE.
 
the following is communicate log between PSTN Gateway and SIPWISE.
 
Please see the following log;
============================================================================
REGISTER sip:sip.baekdoo.local SIP/2.0
Via:SIP/2.0/UDP 192.168.102.254:5060;branch=z9hG4bK1129594316
From:<sip:5799128 at sip.baekdoo.local>;tag=1561012923
To:<sip:5799128 at sip.baekdoo.local>
Call-ID:952467015 at 192.168.102.254
CSeq:20 REGISTER
Contact:*
Max-Forwards: 70
Expires: 0
user-agent: HUMAN TECHNOLOGY - HMG-1016
Allow:INVITE, CANCEL, PRACK, REFER, ACK, INFO, BYE, NOTIFY
Content-Length:0
 
SIP/2.0 100 Trying
Via:SIP/2.0/UDP 192.168.102.254:5060;rport=5060;branch=z9hG4bK1129594316
From:<sip:5799128 at sip.baekdoo.local>;tag=1561012923
To:<sip:5799128 at sip.baekdoo.local>
Call-ID:952467015 at 192.168.102.254
CSeq:20 REGISTER
Server: Sipwise NGCP Proxy 2.X
Content-Length: 0
 
SIP/2.0 401 Unauthorized
Via:SIP/2.0/UDP 192.168.102.254:5060;rport=5060;branch=z9hG4bK1129594316
From:<sip:5799128 at sip.baekdoo.local>;tag=1561012923
To:<sip:5799128 at sip.baekdoo.local>;tag=1d24a28a0bded6c40d31e6db8aab9ac6.1edb
Call-ID:952467015 at 192.168.102.254
CSeq:20 REGISTER
WWW-Authenticate: Digest realm="sip.baekdoo.local", nonce="UUgkOlFIIw4AagrF0Gf5ISH7uv3SM61l"
Server: Sipwise NGCP Proxy 2.X
Content-Length: 0
 
REGISTER sip:sip.baekdoo.local SIP/2.0
Via:SIP/2.0/UDP 192.168.102.254:5060;branch=z9hG4bK1374480614
From:<sip:5799128 at sip.baekdoo.local>;tag=1784607549
To:<sip:5799128 at sip.baekdoo.local>
Call-ID:952467015 at 192.168.102.254
CSeq:21 REGISTER
Contact:*
Authorization:Digest username="testuser4", realm="sip.baekdoo.local", nonce="UUgkOlFIIw4AagrF0Gf5ISH7uv3SM61l", uri="sip:sip.baekdoo.local", response="ff2c8b1a5c065d441453fed04e3904f6", algorithm=MD5
Max-Forwards: 70
Expires: 0
user-agent: HUMAN TECHNOLOGY - HMG-1016
Allow:INVITE, CANCEL, PRACK, REFER, ACK, INFO, BYE, NOTIFY
Content-Length:0
 
SIP/2.0 100 Trying
Via:SIP/2.0/UDP 192.168.102.254:5060;rport=5060;branch=z9hG4bK1374480614
From:<sip:5799128 at sip.baekdoo.local>;tag=1784607549
To:<sip:5799128 at sip.baekdoo.local>
Call-ID:952467015 at 192.168.102.254
CSeq:21 REGISTER
Server: Sipwise NGCP Proxy 2.X
Content-Length: 0
 
SIP/2.0 403 Invalid To-URI Detected
Via:SIP/2.0/UDP 192.168.102.254:5060;rport=5060;branch=z9hG4bK1374480614
From:<sip:5799128 at sip.baekdoo.local>;tag=1784607549
To:<sip:5799128 at sip.baekdoo.local>;tag=1d24a28a0bded6c40d31e6db8aab9ac6.4d11
Call-ID:952467015 at 192.168.102.254
CSeq:21 REGISTER
Server: Sipwise NGCP Proxy 2.X
Content-Length: 0
 
REGISTER sip:sip.baekdoo.local SIP/2.0
Via:SIP/2.0/UDP 192.168.102.254:5060;branch=z9hG4bK3917095761
From:<sip:5799128 at sip.baekdoo.local>;tag=1232014036
To:<sip:5799128 at sip.baekdoo.local>
Call-ID:94144624 at 192.168.102.254
CSeq:20 REGISTER
Contact:<sip:5799128 at 192.168.102.254:5060>
Max-Forwards: 70
Expires: 3600
user-agent: HUMAN TECHNOLOGY - HMG-1016
Allow:INVITE, CANCEL, PRACK, REFER, ACK, INFO, BYE, NOTIFY
Content-Length:0
 
SIP/2.0 100 Trying
Via:SIP/2.0/UDP 192.168.102.254:5060;rport=5060;branch=z9hG4bK3917095761
From:<sip:5799128 at sip.baekdoo.local>;tag=1232014036
To:<sip:5799128 at sip.baekdoo.local>
Call-ID:94144624 at 192.168.102.254
CSeq:20 REGISTER
Server: Sipwise NGCP Proxy 2.X
Content-Length: 0
 
SIP/2.0 401 Unauthorized
Via:SIP/2.0/UDP 192.168.102.254:5060;rport=5060;branch=z9hG4bK3917095761
From:<sip:5799128 at sip.baekdoo.local>;tag=1232014036
To:<sip:5799128 at sip.baekdoo.local>;tag=1d24a28a0bded6c40d31e6db8aab9ac6.98b4
Call-ID:94144624 at 192.168.102.254
CSeq:20 REGISTER
WWW-Authenticate: Digest realm="sip.baekdoo.local", nonce="UUgkP1FIIxO9M9CJvgTf7zCTDIambZIq"
Server: Sipwise NGCP Proxy 2.X
Content-Length: 0
 
REGISTER sip:sip.baekdoo.local SIP/2.0
Via:SIP/2.0/UDP 192.168.102.254:5060;branch=z9hG4bK517439459
From:<sip:5799128 at sip.baekdoo.local>;tag=192689838
To:<sip:5799128 at sip.baekdoo.local>
Call-ID:94144624 at 192.168.102.254
CSeq:21 REGISTER
Contact:<sip:5799128 at 192.168.102.254:5060>
Authorization:Digest username="testuser4", realm="sip.baekdoo.local", nonce="UUgkP1FIIxO9M9CJvgTf7zCTDIambZIq", uri="sip:sip.baekdoo.local", response="ef1a60da2d0e72ad5bad87ebaa551626", algorithm=MD5
Max-Forwards: 70
Expires: 3600
user-agent: HUMAN TECHNOLOGY - HMG-1016
Allow:INVITE, CANCEL, PRACK, REFER, ACK, INFO, BYE, NOTIFY
Content-Length:0
 
SIP/2.0 100 Trying
Via:SIP/2.0/UDP 192.168.102.254:5060;rport=5060;branch=z9hG4bK517439459
From:<sip:5799128 at sip.baekdoo.local>;tag=192689838
To:<sip:5799128 at sip.baekdoo.local>
Call-ID:94144624 at 192.168.102.254
CSeq:21 REGISTER
Server: Sipwise NGCP Proxy 2.X
Content-Length: 0
 
SIP/2.0 403 Invalid To-URI Detected
Via:SIP/2.0/UDP 192.168.102.254:5060;rport=5060;branch=z9hG4bK517439459
From:<sip:5799128 at sip.baekdoo.local>;tag=192689838
To:<sip:5799128 at sip.baekdoo.local>;tag=1d24a28a0bded6c40d31e6db8aab9ac6.a338
Call-ID:94144624 at 192.168.102.254
CSeq:21 REGISTER
Server: Sipwise NGCP Proxy 2.X
Content-Length: 0
=============================================================================
 
As refer the upper log, PSTN gateway tried three times registerating to SIPWISE but The SIPWISE server didn't register PSTN gateway caused by 
 
unathorized(401) and last time received message as Invalid To-URI Detected(403).
 
So I tried several times that I modified values it related authorized with WISESIP server at PSTN Gateway.
 
You could see the values what I could modified it attached as image file name is PSTN_Gateway.png
 
There have three values(Represented number, ID, Password) it needed to modify if PSTN gateway could register to SIPWISE.
 
But results are still same like upper log.
 
Also I attached the image file(test-gw-1.png) it could you know how to configured at SIP Peerings menu in SIPWISE server.
 
Sorry for long message so it could take your time if you could read it.
 
But I really want to know what is problemed.
 
It'll be appreciate give me advise to solve this problem.

Regards,

David Ho.

 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130319/e29aee08/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: test-gw-1.png
Type: application/octet-stream
Size: 51372 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130319/e29aee08/attachment-0002.png>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: PSTN_Gateway.png
Type: application/octet-stream
Size: 11219 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130319/e29aee08/attachment-0003.png>

From agranig at sipwise.com  Tue Mar 19 07:02:45 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Tue, 19 Mar 2013 12:02:45 +0100
Subject: [Spce-user] Delete subscriber does not delete 'trusted' entry
In-Reply-To: <035601ce2027$8e13aab0$aa3b0010$@murray@skyracktelecom.com>
References: <035601ce2027$8e13aab0$aa3b0010$@murray@skyracktelecom.com>
Message-ID: <514845D5.5040901@sipwise.com>

Hi John,

On 03/13/2013 09:15 PM, John Murray wrote:
> In v2.7 when I subscriber is deleted the entry in the trusted table is
> not deleted therefore when adding the same pattern to a new subscriber
> the original (deleted) subscriber is matched.

Another stupid mysql "feature" where triggers are not executed on 
foreign key delete cascades.

We've pushed a fix to 2.7, please do:

	apt-get update && apt-get upgrade && ngcp-update-db-schema

You'll still need to remove zombie entries from the kamailio.trusted 
table manually.

Let me know if it works for you.

Andreas



From rea1ity at kaist.ac.kr  Tue Mar 19 10:31:19 2013
From: rea1ity at kaist.ac.kr (=?UTF-8?B?RGF2aWQgSG8=?=)
Date: Tue, 19 Mar 2013 23:31:19 +0900 (KST)
Subject: [Spce-user] =?utf-8?q?peer_registration_fails?=
Message-ID: <51487a803fdf_@_imoxion.com>


p{margin:0;padding:0;}


Vince,
How could you tried to use WireShark?
So you could capture the more detailed packet between PSTN and Peer server it would be help to figure out what was problemed. 
David


-----------------------????-----------------------
???? : Vince Oscar <vinceoscar at gmail.com>
????: spce-user at lists.sipwise.com
????: 2013-03-18 23:10:15 GMT +0900 (Asia/Seoul)
??: [Spce-user] peer registration fails

I cannot get a peer server to PSTN to register.Server settings are correct with x.x.x.x:5080 5080 is the correct required port I think the peering rules are incorrect.  First who is outbound, caller or callee.  I am confused who is who.
 second I am confused how to form the peering rules.  for example how would I allow all incoming, but only allow 10 digit (ac = 3 and sn=7) and 411 and 911 Thank youVince
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130319/563e362e/attachment-0001.html>

From thilo.bangert at gmail.com  Tue Mar 19 14:17:53 2013
From: thilo.bangert at gmail.com (Thilo Bangert)
Date: Tue, 19 Mar 2013 19:17:53 +0100
Subject: [Spce-user] CSC www UI layout/customisation
In-Reply-To: <51478649.41ae0e0a.09f2.55e2SMTPIN_ADDED_BROKEN@mx.google.com>
References: <51478649.41ae0e0a.09f2.55e2SMTPIN_ADDED_BROKEN@mx.google.com>
Message-ID: <2642087.DcGUBIufWZ@marsupilami>

On Monday 18 March 2013 21:24:45 Gavin Sweet wrote:
> Hi everyone -
> 
> 
> 
> I can see from config.yml that some appearance/branding options are
> available and that tabs in the view can be turned on/off (e.g.
> calllist=1/0).
> 
> These options seem to be: account,   addressbook,  callblock, callforward,
> calllist, desktop, device, reminder, voicebox
> 
> I was wondering, is there a deeper level of modification that can be done as
> configuration, or does that require modification to code?
> 
> For instance, sub-components on pages, say, to show call list but not
> charges.
> 
> If there is more, is there a quick "how-to" that someone is able to post to
> extend the documentation regarding CSC admin config?


this is not a how-to per se, but:

check out the files at /usr/share/ngcp-www-csc

it is the path to the perl catalyst files implementing the csc. for example by 
editing the .tt files in lib/csc/root/tt/ and the layout files in 
lib/csc/root/layout will allow you to customize the appearance to your own 
whishes.

good luck!

> 
> 
> 
> 
> 
> Many thanks,
> 
> Gavin



From gavin.sweet at skyracktelecom.com  Tue Mar 19 17:06:55 2013
From: gavin.sweet at skyracktelecom.com (Gavin Sweet)
Date: Tue, 19 Mar 2013 21:06:55 -0000
Subject: [Spce-user] CSC www UI layout/customisation
In-Reply-To: <2642087.DcGUBIufWZ@marsupilami>
References: <51478649.41ae0e0a.09f2.55e2SMTPIN_ADDED_BROKEN@mx.google.com>
	<2642087.DcGUBIufWZ@marsupilami>
Message-ID: <0c6e01ce24e5$aff9af90$0fed0eb0$@sweet@skyracktelecom.com>

Thanks Thilo - very useful.

I wanted to let users login, but not display charges, only calls (as charges
not yet reliable for us).
I just edited calllist.tt, and removed the elements containing the charge
items (fairly obvious).

This:
            [% IF subscriber.account.billing_profile %]
            <td class="anrufkosten">[% Catalyst.loc('charges') %]</td>
            [% END %]

And this:
            [% IF subscriber.account.billing_profile %]
            <td class="anrufkosten">[%
subscriber.account.billing_profile.data.currency %] [% callentry.call_fee
%]</td>
            [% END %]


Works well!



> -----Original Message-----
> From: Thilo Bangert [mailto:thilo.bangert at gmail.com]
> Sent: 19 March 2013 18:18
> To: spce-user at lists.sipwise.com; gavin.sweet at skyracktelecom.com
> Subject: Re: [Spce-user] CSC www UI layout/customisation
> 
> On Monday 18 March 2013 21:24:45 Gavin Sweet wrote:
> > Hi everyone -
> >
> >
> >
> > I can see from config.yml that some appearance/branding options are
> > available and that tabs in the view can be turned on/off (e.g.
> > calllist=1/0).
> >
> > These options seem to be: account,   addressbook,  callblock,
> callforward,
> > calllist, desktop, device, reminder, voicebox
> >
> > I was wondering, is there a deeper level of modification that can be
> > done as configuration, or does that require modification to code?
> >
> > For instance, sub-components on pages, say, to show call list but not
> > charges.
> >
> > If there is more, is there a quick "how-to" that someone is able to
> > post to extend the documentation regarding CSC admin config?
> 
> 
> this is not a how-to per se, but:
> 
> check out the files at /usr/share/ngcp-www-csc
> 
> it is the path to the perl catalyst files implementing the csc. for
> example by editing the .tt files in lib/csc/root/tt/ and the layout
> files in lib/csc/root/layout will allow you to customize the appearance
> to your own whishes.
> 
> good luck!
> 
> >
> >
> >
> >
> >
> > Many thanks,
> >
> > Gavin
> -----
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2013.0.2904 / Virus Database: 2641/6166 - Release Date:
> 03/12/13 Internal Virus Database is out of date.




From jbonilla at sipwise.com  Tue Mar 19 18:05:24 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Tue, 19 Mar 2013 23:05:24 +0100
Subject: [Spce-user] voicemail
In-Reply-To: <51483A4A.3020809@megatro.com>
References: <51483A4A.3020809@megatro.com>
Message-ID: <20130319230524.65a5a596@quenya>

El Tue, 19 Mar 2013 11:13:30 +0100
Joan Jou <joan at megatro.com> escribi?:

> Hi,
> How can , activate/ desactivate voicemail from user interface and/ admin 
> interface /or from user phone?
> 
> Thank you


You can disable it globally in config.yml


For each subscriber, if you don't forward the calls to voicemail it will never
show.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130319/47ef0cc4/attachment-0001.asc>

From jbonilla at sipwise.com  Tue Mar 19 18:06:28 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Tue, 19 Mar 2013 23:06:28 +0100
Subject: [Spce-user] Please let me know why I faced
 401/403(Unauthorized/Invalid To-URI Detected).
In-Reply-To: <51483ef63fea_@_imoxion.com>
References: <51483ef63fea_@_imoxion.com>
Message-ID: <20130319230628.317e5009@quenya>

El Tue, 19 Mar 2013 19:17:36 +0900 (KST)
David Ho<rea1ity at kaist.ac.kr> escribi?:

> 
> p{margin:0;padding:0;}
> 
> 
> Hi all,
> 
> I tried to register my PSTN Gateway with SIPWISE.
>  
>

why? pstn gateways are peers and peer do not register against the spce. They
are authenticated by ip address.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130319/33361afd/attachment-0001.asc>

From jbonilla at sipwise.com  Tue Mar 19 18:08:10 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Tue, 19 Mar 2013 23:08:10 +0100
Subject: [Spce-user] peer registration fails
In-Reply-To: <CABAd6xtS0edr_mti=+AKpWHjCarHWA39o7SJ6_R9rOyRdkhmtA@mail.gmail.com>
References: <CABAd6xtS0edr_mti=+AKpWHjCarHWA39o7SJ6_R9rOyRdkhmtA@mail.gmail.com>
Message-ID: <20130319230810.24ddb12b@quenya>

El Mon, 18 Mar 2013 10:10:15 -0400
Vince Oscar <vinceoscar at gmail.com> escribi?:

> I cannot get a peer server to PSTN to register.
> Server settings are correct with x.x.x.x:5080
> 5080 is the correct required port
> 
> I think the peering rules are incorrect.  First who is outbound, caller or
> callee.  I am confused who is who.
> 
> second I am confused how to form the peering rules.  for example how would
> I allow all incoming, but only allow 10 digit (ac = 3 and sn=7) and 411 and
> 911
> 
> Thank you
> Vince


Do you want to register the spce to a peer? Or a peer to the spce? If it's the
second option don't. Peers don't register.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130319/0ac56956/attachment-0001.asc>

From jbonilla at sipwise.com  Tue Mar 19 18:11:41 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Tue, 19 Mar 2013 23:11:41 +0100
Subject: [Spce-user] BYE not correct when sent by called party
In-Reply-To: <7BA44D46-0ECA-4012-8B14-406B5295A4E9@mac.com>
References: <mailman.1176.1363363037.2547.spce-user@lists.sipwise.com>
	<7BA44D46-0ECA-4012-8B14-406B5295A4E9@mac.com>
Message-ID: <20130319231141.2d5aa4e8@quenya>

El Fri, 15 Mar 2013 17:30:51 +0100
Jirka Jirout <jirka.jirout at mac.com> escribi?:

> Hello,
> 
> > If you have it enabled, please send us the invite sent by the proxy to sems.
> > sems is controlled by custom headers sent by the proxy (be careful as they
> > contain sensitive info). 
> > 
> > If the headers are right, enable sems debug and let's see what it says.
> 
> here is the INVITE. As far as I can tell, the sst headers are there in
> P-App-Param:
> 

Yes. They look good to me.


> I do not see anything in the log. As a matter of fact, the logging seems to
> have stopped since i set the log level to 3 like you advised me :-)
> 

Please put that in sems.conf and restart sems. Sems debug level 3 is really
verbose.



-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130319/489bb053/attachment-0001.asc>

From m.rys at tri-lakes.net  Tue Mar 19 18:29:41 2013
From: m.rys at tri-lakes.net (Marc Rys)
Date: Tue, 19 Mar 2013 17:29:41 -0500 (CDT)
Subject: [Spce-user] Please let me know why I
	faced401/403(Unauthorized/Invalid To-URI Detected).
In-Reply-To: <20130319230628.317e5009@quenya>
Message-ID: <78ca2f18-8c92-4b31-b7d5-4de651d94f1e@usbsnpzimbra01v.servers.rystec.com>

I'm having the same problem.  error 403 on my test phone when trying to dial a 10 digit north America number.

I just setup SPCE from scratch yesterday, my installation went really well, i got my test phone registered successfully, but when I got to the Peer setup my progress came to a stop.  My peer is another soft switch I have on my network and I have it setup using the same method I've used for trunking to other SIP devices...  (Cisco IADs, Asterisk, Broadworks, SIP LD term providers)

I'm so green, can somebody please point me in the direction of where to look at the SIP messaging logs?  I figure this would explain why Im getting the 403 forbidden error.

I'm going to also wireshark my single NIC interface on SPCE right now to see if that gives me any clues

Thanks in advance..

Marc

http://www.tri-lakes.net
http://www.rystec.com




----- Original Message -----
From: "Jon Bonilla" <jbonilla at sipwise.com>
To: spce-user at lists.sipwise.com
Sent: Tuesday, March 19, 2013 5:06:28 PM
Subject: Re: [Spce-user] Please let me know why I faced401/403(Unauthorized/Invalid To-URI Detected).

El Tue, 19 Mar 2013 19:17:36 +0900 (KST)
David Ho<rea1ity at kaist.ac.kr> escribi?:

> 
> p{margin:0;padding:0;}
> 
> 
> Hi all,
> 
> I tried to register my PSTN Gateway with SIPWISE.
>  
>

why? pstn gateways are peers and peer do not register against the spce. They
are authenticated by ip address.


_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com
http://lists.sipwise.com/listinfo/spce-user



From jbonilla at sipwise.com  Tue Mar 19 18:33:02 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Tue, 19 Mar 2013 23:33:02 +0100
Subject: [Spce-user] Send peer inbound call straight to voicemail
In-Reply-To: <5146df74.c95fb40a.075c.16e1@mx.google.com>
References: <021301ce1b80$79b93db0$6d2bb910$@murray@skyracktelecom.com>
	<20130315102630.32c7a024@quenya>
	<5146df74.c95fb40a.075c.16e1@mx.google.com>
Message-ID: <20130319233302.3f768ba9@quenya>

El Mon, 18 Mar 2013 09:33:35 -0000
"John Murray" <john.murray at skyracktelecom.com> escribi?:

> Jon,
> 
> The problem initially to be authentication.
> 
> The PBX is actually a gateway and does not register with SPCE. I need to give
> each gateway subscriber an SPCE subscriber so I use the trusted source model.
> IP address of gateway + userpart of From: field matches the subscriber for
> them to make calls. For calls to the subscriber I have a static
> contact/location pointing at the gateway.
> 
> However when a call is put out to the subscriber on the gateway and they are
> not available a call comes in from the gateway to SPCE with 'From:' the
> original caller 'To:' the voicemail number and 'Divert:' the subscriber
> details.
> 
> This call is rejected with '407 Proxy Authentication Required' as there is no
> PSTN peer and no trusted source matching the 'From:' because the From: is the
> originating caller in the PSTN.
> 
> I cannot add a peer with the gateway's address as in the logic this is
> checked before trusted sources and non of my subscribers are then matched.
> 
> Would it not be better to check trusted sources first?
> 
>

You could modify the proxy configuration and modify ROUTE_AUTH. There you have
the "allow_trusted" function call.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130319/f6ae6a2d/attachment-0001.asc>

From jbonilla at sipwise.com  Tue Mar 19 18:33:54 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Tue, 19 Mar 2013 23:33:54 +0100
Subject: [Spce-user] SOAP array problem
In-Reply-To: <51308D35.4070804@hfcnetwork.hu>
References: <5119FF27.8020107@hfcnetwork.hu> <20130301103828.3b3b31ce@quenya>
	<51308D35.4070804@hfcnetwork.hu>
Message-ID: <20130319233354.7fee53dd@quenya>

El Fri, 01 Mar 2013 12:12:53 +0100
HFC-Network Kft - Szab? J?nos <szabojanos at hfcnetwork.hu> escribi?:

> Hi,
> 
> No, it's still under investigation.
> 
>

Is this still open?

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130319/3a34acdb/attachment-0001.asc>

From jbonilla at sipwise.com  Tue Mar 19 18:37:50 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Tue, 19 Mar 2013 23:37:50 +0100
Subject: [Spce-user] Please let me know why I
 faced401/403(Unauthorized/Invalid To-URI Detected).
In-Reply-To: <78ca2f18-8c92-4b31-b7d5-4de651d94f1e@usbsnpzimbra01v.servers.rystec.com>
References: <20130319230628.317e5009@quenya>
	<78ca2f18-8c92-4b31-b7d5-4de651d94f1e@usbsnpzimbra01v.servers.rystec.com>
Message-ID: <20130319233750.043dd6c7@quenya>

El Tue, 19 Mar 2013 17:29:41 -0500 (CDT)
Marc Rys <m.rys at tri-lakes.net> escribi?:

> I'm having the same problem.  error 403 on my test phone when trying to dial
> a 10 digit north America number.
> 
> I just setup SPCE from scratch yesterday, my installation went really well, i
> got my test phone registered successfully, but when I got to the Peer setup
> my progress came to a stop.  My peer is another soft switch I have on my
> network and I have it setup using the same method I've used for trunking to
> other SIP devices...  (Cisco IADs, Asterisk, Broadworks, SIP LD term
> providers)
> 
> I'm so green, can somebody please point me in the direction of where to look
> at the SIP messaging logs?  I figure this would explain why Im getting the
> 403 forbidden error.
> 
> I'm going to also wireshark my single NIC interface on SPCE right now to see
> if that gives me any clues
> 
> Thanks in advance..
> 
>

ngrep-sip is the quickest way to check what's going on in the network.

checking the proxy logs is usually the best method to see what happened with a
particular call. If the problem is related to permisions or routing there you
have the answer.  You can find the logs under /var/log/ngcp/*

cheers,

Jon

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130319/7df369ab/attachment-0001.asc>

From john.murray at skyracktelecom.com  Tue Mar 19 19:41:00 2013
From: john.murray at skyracktelecom.com (John Murray)
Date: Tue, 19 Mar 2013 23:41:00 -0000
Subject: [Spce-user] Delete subscriber does not delete 'trusted' entry
In-Reply-To: <514845D5.5040901@sipwise.com>
References: <035601ce2027$8e13aab0$aa3b0010$@murray@skyracktelecom.com>
	<514845D5.5040901@sipwise.com>
Message-ID: <5148f78d.8a0db40a.2cc6.180c@mx.google.com>

Andreas, looking good!

Many thanks again.

John

-----Original Message-----
From: spce-user-bounces at lists.sipwise.com
[mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of Andreas Granig
Sent: 19 March 2013 11:03
To: spce-user at lists.sipwise.com
Subject: Re: [Spce-user] Delete subscriber does not delete 'trusted' entry

Hi John,

On 03/13/2013 09:15 PM, John Murray wrote:
> In v2.7 when I subscriber is deleted the entry in the trusted table is 
> not deleted therefore when adding the same pattern to a new subscriber 
> the original (deleted) subscriber is matched.

Another stupid mysql "feature" where triggers are not executed on foreign
key delete cascades.

We've pushed a fix to 2.7, please do:

	apt-get update && apt-get upgrade && ngcp-update-db-schema

You'll still need to remove zombie entries from the kamailio.trusted table
manually.

Let me know if it works for you.

Andreas

_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com
http://lists.sipwise.com/listinfo/spce-user




From john.murray at skyracktelecom.com  Tue Mar 19 19:58:01 2013
From: john.murray at skyracktelecom.com (John Murray)
Date: Tue, 19 Mar 2013 23:58:01 -0000
Subject: [Spce-user] Send peer inbound call straight to voicemail
In-Reply-To: <20130319233302.3f768ba9@quenya>
References: <021301ce1b80$79b93db0$6d2bb910$@murray@skyracktelecom.com>
	<20130315102630.32c7a024@quenya>
	<5146df74.c95fb40a.075c.16e1@mx.google.com>
	<20130319233302.3f768ba9@quenya>
Message-ID: <5148fb8a.a567b40a.0ed1.1a23@mx.google.com>

Jon hi,

So I got this fixed.
I modified ROUTE_AUTH as:

route[ROUTE_AUTH]
{
	if(allow_trusted("$avp(s:ip)", "$avp(s:proto)"))
	{
                if(is_avp_set("$avp(s:peer_uuid)/s"))
                {
                        # use tag-column from trusted-table as uuid for this caller
                        avp_copy("$avp(s:peer_uuid)", "$avp(s:caller_uuid)/d");
  NEW --------->        avp_db_query("select username, domain from subscriber where uuid='$avp(S:caller_uuid)'", "$avp(s:caller_username), $avp(s:caller_domain)");
  NEW --------->         xlog("L_INFO", "Trusted peer with username '$avp(s:caller_username)' - [% logreq -%]\n");
                }
                else
                {
                        # if no uuid is set, use "0" as default uuid
                        $avp(s:caller_uuid) = "0";
                }

Then in route[ROUTE_LOAD_CALLER_PREF]:

if($var(forward) == 1 || $var(cf_loop) == 1 || $avp(s:from_faxserver) == 1)
        {
                $var(pref_domain) = "sip:" + $fU + "@" + $avp(s:acc_caller_domain);
        }
        else
        {
                if ($var(from_trusted) != 1)
                {
                        $var(pref_domain) = "sip:" + $fU + "@" + $avp(s:caller_domain);
                }
                else
                {
NEW ------------->      $var(pref_domain) = "sip:" + $avp(s:caller_username) + "@" + $avp(s:caller_domain);
                        #$var(pref_domain) = $fu;
                }



Many thanks

John


-----Original Message-----
From: Jon Bonilla (Manwe) [mailto:jbonilla at sipwise.com] 
Sent: 19 March 2013 22:33
To: spce-user at lists.sipwise.com
Cc: John Murray
Subject: Re: [Spce-user] Send peer inbound call straight to voicemail

El Mon, 18 Mar 2013 09:33:35 -0000
"John Murray" <john.murray at skyracktelecom.com> escribi?:

> Jon,
> 
> The problem initially to be authentication.
> 
> The PBX is actually a gateway and does not register with SPCE. I need 
> to give each gateway subscriber an SPCE subscriber so I use the trusted source model.
> IP address of gateway + userpart of From: field matches the subscriber 
> for them to make calls. For calls to the subscriber I have a static 
> contact/location pointing at the gateway.
> 
> However when a call is put out to the subscriber on the gateway and 
> they are not available a call comes in from the gateway to SPCE with 
> 'From:' the original caller 'To:' the voicemail number and 'Divert:' 
> the subscriber details.
> 
> This call is rejected with '407 Proxy Authentication Required' as 
> there is no PSTN peer and no trusted source matching the 'From:' 
> because the From: is the originating caller in the PSTN.
> 
> I cannot add a peer with the gateway's address as in the logic this is 
> checked before trusted sources and non of my subscribers are then matched.
> 
> Would it not be better to check trusted sources first?
> 
>

You could modify the proxy configuration and modify ROUTE_AUTH. There you have the "allow_trusted" function call.





From rea1ity at kaist.ac.kr  Wed Mar 20 05:00:45 2013
From: rea1ity at kaist.ac.kr (=?UTF-8?B?RGF2aWQgSG8=?=)
Date: Wed, 20 Mar 2013 18:00:45 +0900 (KST)
Subject: [Spce-user]
	=?utf-8?q?Please_let_me_know_why_Ifaced401/403=28Unau?=
	=?utf-8?q?thorized/Invalid_To-URI_Detected=29=2E?=
Message-ID: <51497c2b3fe4_@_imoxion.com>

Hi Jon,
 
When I open at "/var/log/ngcp/sem/sems.log" file, there are messages same like following;
-----------------------------------------------------------------------------------------------------------------
Mar 20 08:44:14 spce sems[2164]: [#7f601e77d700] [doRegistration, AmSipRegistration.cpp:139] ERROR: failed to send registration.
-----------------------------------------------------------------------------------------------------------------
It seems to occurred "doRegistration" function error in AmSipRegistration.cpp when I tried to register the PSTN gateway.
How do you think it?
Please let me know if you have to solve this problem.
 
Regards,
David Ho
 


-----------------------????-----------------------
???? : "Jon Bonilla (Manwe)" <jbonilla at sipwise.com>
????: spce-user at lists.sipwise.com
????: 2013-03-20 07:37:50 GMT +0900 (Asia/Seoul)
??: Re: [Spce-user] Please let me know why Ifaced401/403(Unauthorized/Invalid To-URI Detected).

El Tue, 19 Mar 2013 17:29:41 -0500 (CDT)

Marc Rys <m.rys at tri-lakes.net> escribi?:



> I'm having the same problem.  error 403 on my test phone when trying to dial

> a 10 digit north America number.

> 

> I just setup SPCE from scratch yesterday, my installation went really well, i

> got my test phone registered successfully, but when I got to the Peer setup

> my progress came to a stop.  My peer is another soft switch I have on my

> network and I have it setup using the same method I've used for trunking to

> other SIP devices...  (Cisco IADs, Asterisk, Broadworks, SIP LD term

> providers)

> 

> I'm so green, can somebody please point me in the direction of where to look

> at the SIP messaging logs?  I figure this would explain why Im getting the

> 403 forbidden error.

> 

> I'm going to also wireshark my single NIC interface on SPCE right now to see

> if that gives me any clues

> 

> Thanks in advance..

> 

>



ngrep-sip is the quickest way to check what's going on in the network.



checking the proxy logs is usually the best method to see what happened with a

particular call. If the problem is related to permisions or routing there you

have the answer.  You can find the logs under /var/log/ngcp/*



cheers,



Jon



_______________________________________________

Spce-user mailing list

Spce-user at lists.sipwise.com

http://lists.sipwise.com/listinfo/spce-user

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130320/ddc242ad/attachment-0001.html>

From jbonilla at sipwise.com  Wed Mar 20 05:08:06 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Wed, 20 Mar 2013 10:08:06 +0100
Subject: [Spce-user] Please let me know why
 Ifaced401/403(Unauthorized/Invalid To-URI Detected).
In-Reply-To: <51497c273fec_@_imoxion.com>
References: <51497c273fec_@_imoxion.com>
Message-ID: <20130320100806.03427c63@quenya>

El Wed, 20 Mar 2013 18:00:45 +0900 (KST)
David Ho<rea1ity at kaist.ac.kr> escribi?:

> Hi Jon,
>  
> When I open at "/var/log/ngcp/sem/sems.log" file, there are messages same
> like following;
> -----------------------------------------------------------------------------------------------------------------
> Mar 20 08:44:14 spce sems[2164]: [#7f601e77d700] [doRegistration,
> AmSipRegistration.cpp:139] ERROR: failed to send registration.
> -----------------------------------------------------------------------------------------------------------------
> It seems to occurred "doRegistration" function error in AmSipRegistration.cpp
> when I tried to register the PSTN gateway. How do you think it? Please let me
> know if you have to solve this problem. Regards,
> David Ho
>  
> 

IMHO you're mixing things. 403 to your subscribers, registration against a peer
gw... please let's go one by one. 

Let's start first with you peer registration. Do you need to register there?
What did you do for that? How does you configuration files look like?

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130320/8414d01a/attachment-0001.asc>

From rea1ity at kaist.ac.kr  Wed Mar 20 06:18:21 2013
From: rea1ity at kaist.ac.kr (=?UTF-8?B?RGF2aWQgSG8=?=)
Date: Wed, 20 Mar 2013 19:18:21 +0900 (KST)
Subject: [Spce-user]
	=?utf-8?q?Please_let_me_know_whyIfaced401/403=28Unaut?=
	=?utf-8?q?horized/Invalid_To-URI_Detected=29=2E?=
Message-ID: <51498ece3fd8_@_imoxion.com>


p{margin:0;padding:0;}


Oh my GOD!
You are right.
I think that I faced seriousley critical problemed.
When I opened the other files as "kamailio-options-proxy_log" and "kamailio-proxy_log" and "kamailio-lb_log" at "/var/log/ngcp/" path,
There have so many logs made it appear New request and request reply per a second.
Moreover, there have the user that I didn't set it but it could working internally.
I.E."R=sip:nagios at voip.sipwise.local F=sip:sipsak at 127.0.0.1:8485 T=sip:nagios at voip.sipwise.local"
My SIPWISE server operate it downloaded from vmware image.
So may I have some missing to configurate of sipwise. :(
I'm just tried to configured it as following SIPWISE document version 2.7 because I don't know well about VoIP.
Now I understand why the document recommended manually install SIPWISE.
Maybe, this problem must be solve first before that I configuration of peer registration modified.
Anyway, I set a peer same like the image files that I attached it name is "test-gw-1.png".
 
 


-----------------------????-----------------------
???? : "Jon Bonilla (Manwe)" <jbonilla at sipwise.com>
????: "spce-user at lists.sipwise.com" <spce-user at lists.sipwise.com>
? ?: David Ho <rea1ity at kaist.ac.kr>
????: 2013-03-20 18:08:06 GMT +0900 (Asia/Seoul)
??: Re: [Spce-user] Please let me know whyIfaced401/403(Unauthorized/Invalid To-URI Detected).

El Wed, 20 Mar 2013 18:00:45 +0900 (KST)

David Ho<rea1ity at kaist.ac.kr> escribi?:



> Hi Jon,

>  

> When I open at "/var/log/ngcp/sem/sems.log" file, there are messages same

> like following;

> -----------------------------------------------------------------------------------------------------------------

> Mar 20 08:44:14 spce sems[2164]: [#7f601e77d700] [doRegistration,

> AmSipRegistration.cpp:139] ERROR: failed to send registration.

> -----------------------------------------------------------------------------------------------------------------

> It seems to occurred "doRegistration" function error in AmSipRegistration.cpp

> when I tried to register the PSTN gateway. How do you think it? Please let me

> know if you have to solve this problem. Regards,

> David Ho

>  

> 



IMHO you're mixing things. 403 to your subscribers, registration against a peer

gw... please let's go one by one. 



Let's start first with you peer registration. Do you need to register there?

What did you do for that? How does you configuration files look like?




-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130320/5e822de2/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: kamailio-options-proxy_log.txt
Type: application/octet-stream
Size: 4981 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130320/5e822de2/attachment-0003.txt>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: kamailio-proxy_log.txt
Type: application/octet-stream
Size: 6458 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130320/5e822de2/attachment-0004.txt>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: kamailio-lb_log.txt
Type: application/octet-stream
Size: 3943 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130320/5e822de2/attachment-0005.txt>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: test-gw-1.png
Type: application/octet-stream
Size: 51372 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130320/5e822de2/attachment-0001.png>

From jbonilla at sipwise.com  Wed Mar 20 06:35:53 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Wed, 20 Mar 2013 11:35:53 +0100
Subject: [Spce-user] Please let me know
 whyIfaced401/403(Unauthorized/Invalid To-URI Detected).
In-Reply-To: <51498ecb3fe0_@_imoxion.com>
References: <51498ecb3fe0_@_imoxion.com>
Message-ID: <20130320113553.0345a804@quenya>

El Wed, 20 Mar 2013 19:18:21 +0900 (KST)
David Ho<rea1ity at kaist.ac.kr> escribi?:

> 
> p{margin:0;padding:0;}
> 
> 
> Oh my GOD!
> You are right.
> I think that I faced seriousley critical problemed.
> When I opened the other files as "kamailio-options-proxy_log" and
> "kamailio-proxy_log" and "kamailio-lb_log" at "/var/log/ngcp/" path, There
> have so many logs made it appear New request and request reply per a second.
> Moreover, there have the user that I didn't set it but it could working
> internally. I.E."R=sip:nagios at voip.sipwise.local F=sip:sipsak at 127.0.0.1:8485
> T=sip:nagios at voip.sipwise.local" My SIPWISE server operate it downloaded from
> vmware image. So may I have some missing to configurate of sipwise. :( I'm
> just tried to configured it as following SIPWISE document version 2.7 because
> I don't know well about VoIP. Now I understand why the document recommended
> manually install SIPWISE.


No, what you achieve manually installing the provider is more or less the same
as you get with the vmware image. The internal sipsak user you have found is an
internal user for monitoring the system. 

> Maybe, this problem must be solve first before that
> I configuration of peer registration modified. Anyway, I set a peer same like
> the image files that I attached it name is "test-gw-1.png". 


you need to go step by step. Follow the handbook, create you doain, create and
account, create two subscribers and call from one subscriber to the other. Once
you're there we'll talk about peering servers.

I think you are mixing too many things so either you follow the handbook
quick-start or you'll need to learn about Voip (or both).


cheers,

Jon

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130320/5863e37a/attachment-0001.asc>

From vinceoscar at gmail.com  Wed Mar 20 07:42:43 2013
From: vinceoscar at gmail.com (Vince Oscar)
Date: Wed, 20 Mar 2013 07:42:43 -0400
Subject: [Spce-user] confused how to form the peering rules
Message-ID: <CABAd6xsY44VhkXSTefa=aCiR2dGJk8tQvfiP4Qeanj4y0-c5KQ@mail.gmail.com>

I cannot get an outgoing call to complete to PSTN where vendor requires
port 5080
Server settings are correct with x.x.x.x:5080
5080 is the correct required port
Is there a config file the I need to modify?

I think maybe the peering rules are incorrect.

First, who is outbound, the caller or the callee?  I am confused who is who.

Second
I am .

For example,
how would I allow all incoming calls, but only allow 10 digit (ac = 3 and
sn=7) and 411 and
911 for outbound?
Thank you
Vince
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130320/52b2d81f/attachment-0001.html>

From jbonilla at sipwise.com  Wed Mar 20 10:22:43 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Wed, 20 Mar 2013 15:22:43 +0100
Subject: [Spce-user] confused how to form the peering rules
In-Reply-To: <CABAd6xsY44VhkXSTefa=aCiR2dGJk8tQvfiP4Qeanj4y0-c5KQ@mail.gmail.com>
References: <CABAd6xsY44VhkXSTefa=aCiR2dGJk8tQvfiP4Qeanj4y0-c5KQ@mail.gmail.com>
Message-ID: <20130320152243.354755ce@quenya>

El Wed, 20 Mar 2013 07:42:43 -0400
Vince Oscar <vinceoscar at gmail.com> escribi?:

> I cannot get an outgoing call to complete to PSTN where vendor requires
> port 5080

So your peer listens port 5080. No problem as when you define a gateway you can
put there ip and port. You have fields for that.

> Server settings are correct with x.x.x.x:5080
> 5080 is the correct required port
> Is there a config file the I need to modify?

Not for configuring a peer server inside a peer host.


> 
> I think maybe the peering rules are incorrect.
> 
> First, who is outbound, the caller or the callee?  I am confused who is who.
> 

Caller makes the call. Callee is the destination of the call.


> Second
> I am .
> 
> For example,
> how would I allow all incoming calls, but only allow 10 digit (ac = 3 and
> sn=7) and 411 and
> 911 for outbound?
> Thank you
> Vince

Incoming calls are always accepted rom pstn peers by default. Peering rules
tell the system which outgoing calls can be sent to the peering servers in that
group.

If you use callee_preffix you can use just prefixes like 43
 which would be Vienna.

If you want to allow only 10 digit destinations you need to use callee_patter
instead. Something like ^[0.9]{10} 

But peering rules are not the way to allow/disallow your customers to dial to
certain destinations. That's NCOS and call barring. Peering rules is for
sending calls to certain peers for peering contract reasons.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130320/f9888ffe/attachment-0001.asc>

From csaroli at isysp.com  Wed Mar 20 11:11:56 2013
From: csaroli at isysp.com (Christopher Saroli)
Date: Wed, 20 Mar 2013 15:11:56 +0000
Subject: [Spce-user] Billing Profiles/Fees
In-Reply-To: <20130315102112.37d2c298@quenya>
References: <EE7149FDD60B3045A4377F05AAFFF3F118342A46@ISS-Server.ISS.local>
	<20130315102112.37d2c298@quenya>
Message-ID: <EE7149FDD60B3045A4377F05AAFFF3F118358CB8@ISS-Server.ISS.local>

Thanks for responding Jon.  So I started the rate-o-mat service.  Guess it was off. Is this supposed to run on startup? Either way half of it is working now.

So outgoing calls are all being rated and showing up in the exported cdr but incoming calls are not. 

Interesting enough, when I check the customer portal the outgoing calls are listed in the following format +11xxxxxxxxxx while incoming are +xxxxxxxxxx.  The duration of the call is listed in the customer portal for both. Rewrite rule?  

Thanks

---------------------------------
Christopher Saroli
csaroli at isysp.com

???????????????????????????????? 586-226-7800 (Main Office)
                                 586-884-0800 (Direct)


-----Original Message-----
From: spce-user-bounces at lists.sipwise.com [mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of Jon Bonilla
Sent: Friday, March 15, 2013 5:21 AM
To: spce-user at lists.sipwise.com
Subject: Re: [Spce-user] Billing Profiles/Fees

El Wed, 13 Mar 2013 15:52:10 +0000
Christopher Saroli <csaroli at isysp.com> escribi?:

> Little confused on this, every call I make/receive does not get a fee 
> applied to it.  Just trying right now to apply a fee to all calls.  I 
> might be totally wrong here, any info would be appreciated.
> 
> This is my config:
> 
> Billing Profile:
> Interval Charge:  0.02
> Freetime: 	  0
> Free Cash: 	  0
> Monthly Lim	  0
> Lock		  none
> Currency: 	 $
> 
> 
> Bililng Fees
> Destination:  	.*
> Zone:      	All
> Init Rate:	1
> Init interval:	1
> Follow Rate: 	1
> Follow interval: 1
> 
> 
> Also I have no outgoing calls recorded in the CDR under the admin 
> interface but in the customer portal they are there. Is this by design 
> or is something wrong.
>

Well, he cdrs should be shown in both interfaces. We can go there later on. But anyways, to check if your calls are rated fine you should check the exported cdrs (if any).

Is rate-o-mat running in your system? If not, your calls won't be rated and you won't get exported cdrs.


From jbonilla at sipwise.com  Wed Mar 20 11:20:06 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Wed, 20 Mar 2013 16:20:06 +0100
Subject: [Spce-user] Billing Profiles/Fees
In-Reply-To: <EE7149FDD60B3045A4377F05AAFFF3F118358CB8@ISS-Server.ISS.local>
References: <EE7149FDD60B3045A4377F05AAFFF3F118342A46@ISS-Server.ISS.local>
	<20130315102112.37d2c298@quenya>
	<EE7149FDD60B3045A4377F05AAFFF3F118358CB8@ISS-Server.ISS.local>
Message-ID: <20130320162006.1d0f2522@quenya>

El Wed, 20 Mar 2013 15:11:56 +0000
Christopher Saroli <csaroli at isysp.com> escribi?:

> Thanks for responding Jon.  So I started the rate-o-mat service.  Guess it
> was off. Is this supposed to run on startup? Either way half of it is working
> now.
> 

It should start on startup unless you disable it in config.yml. But in that
case the cdrs would be exported without rating information.

> So outgoing calls are all being rated and showing up in the exported cdr but
> incoming calls are not. 

If you want incoming calls to be exported you need to enable that option in
config.yml under the cdrexport section. By default only outgoing sucessfull
calls are exported.

> 
> Interesting enough, when I check the customer portal the outgoing calls are
> listed in the following format +11xxxxxxxxxx while incoming are +xxxxxxxxxx.
> The duration of the call is listed in the customer portal for both. Rewrite
> rule?  
> 

I *strongly* suggest you and every spce user to *always* normalize the incoming
numbers to E164 format and send them in any way you want denormalizing them via
rewrite rules.

The system and all it's components should always work in E164 format. And you
should always work in that way when creating peeing rules, billing profiles...
etc


PD: Did I mention *strongly suggest E164 format*? I meant it.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130320/445df4ec/attachment-0001.asc>

From csaroli at isysp.com  Wed Mar 20 11:58:02 2013
From: csaroli at isysp.com (Christopher Saroli)
Date: Wed, 20 Mar 2013 15:58:02 +0000
Subject: [Spce-user] Billing Profiles/Fees
In-Reply-To: <20130320162006.1d0f2522@quenya>
References: <EE7149FDD60B3045A4377F05AAFFF3F118342A46@ISS-Server.ISS.local>
	<20130315102112.37d2c298@quenya>
	<EE7149FDD60B3045A4377F05AAFFF3F118358CB8@ISS-Server.ISS.local>
	<20130320162006.1d0f2522@quenya>
Message-ID: <EE7149FDD60B3045A4377F05AAFFF3F118358D13@ISS-Server.ISS.local>

Thanks Jon, I will check the config and enable the incoming.

As for rewrite rules, when my peering sends north American calls to me it is not in E.164.  It is without the CC. So I have a rewrite rule in place to add the 1. All the phone systems here do not want the CC so I have to strip the 1 when I send it back out. Is there a better way for me to go about this. I have the following rewrite rules in place

(Applied to domain)
Outbound Rewite Rules for Callee 
^(1)([1-9][0-9]+)$                   \2                          
Inbound Rewrite Rules for Callee
^([1-9][0-9]{9})                        1\1   (needed this in order for subscribers within sipwise to call each other)


(Applied to my peering)
Outbound rewrite rules for Calle
^([0-9]{7})$                             ${caller_cc}${caller_ac}\1
Inbound Rewrite rules for Callee
^([1-9][0-9]{9})                         1\1

Thanks for all your help








---------------------------------
Christopher Saroli
csaroli at isysp.com

?????????????????


-----Original Message-----
From: Jon Bonilla [mailto:jbonilla at sipwise.com] 
Sent: Wednesday, March 20, 2013 11:20 AM
To: spce-user@
Cc: Christopher Saroli
Subject: Re: [Spce-user] Billing Profiles/Fees

El Wed, 20 Mar 2013 15:11:56 +0000
Christopher Saroli <csaroli at isysp.com> escribi?:

> Thanks for responding Jon.  So I started the rate-o-mat service.  
> Guess it was off. Is this supposed to run on startup? Either way half 
> of it is working now.
> 

It should start on startup unless you disable it in config.yml. But in that case the cdrs would be exported without rating information.

> So outgoing calls are all being rated and showing up in the exported 
> cdr but incoming calls are not.

If you want incoming calls to be exported you need to enable that option in config.yml under the cdrexport section. By default only outgoing sucessfull calls are exported.

> 
> Interesting enough, when I check the customer portal the outgoing 
> calls are listed in the following format +11xxxxxxxxxx while incoming are +xxxxxxxxxx.
> The duration of the call is listed in the customer portal for both. 
> Rewrite rule?
> 

I *strongly* suggest you and every spce user to *always* normalize the incoming numbers to E164 format and send them in any way you want denormalizing them via rewrite rules.

The system and all it's components should always work in E164 format. And you should always work in that way when creating peeing rules, billing profiles...
etc


PD: Did I mention *strongly suggest E164 format*? I meant it.


From jbonilla at sipwise.com  Wed Mar 20 12:31:27 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Wed, 20 Mar 2013 17:31:27 +0100
Subject: [Spce-user] Billing Profiles/Fees
In-Reply-To: <EE7149FDD60B3045A4377F05AAFFF3F118358D13@ISS-Server.ISS.local>
References: <EE7149FDD60B3045A4377F05AAFFF3F118342A46@ISS-Server.ISS.local>
	<20130315102112.37d2c298@quenya>
	<EE7149FDD60B3045A4377F05AAFFF3F118358CB8@ISS-Server.ISS.local>
	<20130320162006.1d0f2522@quenya>
	<EE7149FDD60B3045A4377F05AAFFF3F118358D13@ISS-Server.ISS.local>
Message-ID: <20130320173127.606ca7e7@quenya>

El Wed, 20 Mar 2013 15:58:02 +0000
Christopher Saroli <csaroli at isysp.com> escribi?:

> Thanks Jon, I will check the config and enable the incoming.
> 
> As for rewrite rules, when my peering sends north American calls to me it is
> not in E.164.  It is without the CC. So I have a rewrite rule in place to add
> the 1. All the phone systems here do not want the CC so I have to strip the 1
> when I send it back out.

Yes. that's exactly how it should be and that's why we have rewrite rules
there. In every install I need to normalize what peers/subscribers send and
also denormalize what I send them to the format they expect. 

This might be confusing at the beginning but once you're done rewrite rules are
your best friends.


> Is there a better way for me to go about this. I
> have the following rewrite rules in place
> 
> (Applied to domain)
> Outbound Rewite Rules for Callee 
> ^(1)([1-9][0-9]+)$                   \2                          
> Inbound Rewrite Rules for Callee
> ^([1-9][0-9]{9})                        1\1   (needed this in order for
> subscribers within sipwise to call each other)
> 

Looks good as starting point. remember that you have to normalize everything
you'd expect a user to dial. International, emergency, national, short
numbers... etc


> 
> (Applied to my peering)
> Outbound rewrite rules for Calle
> ^([0-9]{7})$                             ${caller_cc}${caller_ac}\1
> Inbound Rewrite rules for Callee
> ^([1-9][0-9]{9})                         1\1
> 
> Thanks for all your help
> 
> 

again looks good as starting point.


You'll see it in kamailio-proxy logs. How the numbers are normalized and once
you create peering rules and ncos levels in E164 format you'll see that calls
cannot be routed or rated in case they were not normalized.


You're in the right path :)

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130320/e369f3ba/attachment-0001.asc>

From rea1ity at kaist.ac.kr  Wed Mar 20 23:24:34 2013
From: rea1ity at kaist.ac.kr (=?UTF-8?B?RGF2aWQgSG8=?=)
Date: Thu, 21 Mar 2013 12:24:34 +0900 (KST)
Subject: [Spce-user]
	=?utf-8?q?Please_let_me_knowwhyIfaced401/403=28Unauth?=
	=?utf-8?q?orized/Invalid_To-URI_Detected=29=2E?=
Message-ID: <514a7ee23fe6_@_imoxion.com>

 
 
Jon,
 
I've already followed up the document that you provide "The sip:provider CE Handbook 2.7" so I'm followed setup 1 to 6 chapter and almost setting same like the document.
And now I'm tried to modify values of Peering Servers several times and you could know where I'm modified the values at "6.4.4. Authenticating and Registering against Peering Servers".
the URL following;
http://www.sipwise.com/doc/2.7/spce/ar01s06.html#_authenticating_and_registering_against_peering_servers
 
So manytimes, SIPWISE server denied my PSTN gateway caused by Unauthorized(401).
And I found what is problemed via sems log file("/var/log/ngcp/sem/sems.log") it appeared error message before I mentioned same like following;
-----------------------------------------------------------------------------------------------------------------
Mar 20 08:44:14 spce sems[2164]: [#7f601e77d700] [doRegistration, AmSipRegistration.cpp:139] ERROR: failed to send registration.
-----------------------------------------------------------------------------------------------------------------
So 'doRegistration function in AmSipRegistration.cpp file occurred ERROR message.
If you were me, how could solve the error occurred?
 
And it is another one that I afraid thing.
the log files in SIPWISE as kamailio-lb.log and kamailio-options-proxy.log increased capacity too much.
my kamailio-lb.log file capacity is over than 28MB and kamailio-options-proxy.log file capacity is more than 7MB.
The reason of the log too much increased capacity caused by stacked approax. four to five log messages per a second!
Is it alright to leave it even though it will be huge capacity log file after than this log file could occurr system error for leak of hard disk capacity.
 
David


-----------------------????-----------------------
???? : "Jon Bonilla (Manwe)" <jbonilla at sipwise.com>
????: "spce-user at lists.sipwise.com" <spce-user at lists.sipwise.com>
? ?: David Ho <rea1ity at kaist.ac.kr>
????: 2013-03-20 19:35:53 GMT +0900 (Asia/Seoul)
??: Re: [Spce-user] Please let me knowwhyIfaced401/403(Unauthorized/Invalid To-URI Detected).

El Wed, 20 Mar 2013 19:18:21 +0900 (KST)

David Ho<rea1ity at kaist.ac.kr> escribi?:



> 

> p{margin:0;padding:0;}

> 

> 

> Oh my GOD!

> You are right.

> I think that I faced seriousley critical problemed.

> When I opened the other files as "kamailio-options-proxy_log" and

> "kamailio-proxy_log" and "kamailio-lb_log" at "/var/log/ngcp/" path, There

> have so many logs made it appear New request and request reply per a second.

> Moreover, there have the user that I didn't set it but it could working

> internally. I.E."R=sip:nagios at voip.sipwise.local F=sip:sipsak at 127.0.0.1:8485

> T=sip:nagios at voip.sipwise.local" My SIPWISE server operate it downloaded from

> vmware image. So may I have some missing to configurate of sipwise. :( I'm

> just tried to configured it as following SIPWISE document version 2.7 because

> I don't know well about VoIP. Now I understand why the document recommended

> manually install SIPWISE.





No, what you achieve manually installing the provider is more or less the same

as you get with the vmware image. The internal sipsak user you have found is an

internal user for monitoring the system. 



> Maybe, this problem must be solve first before that

> I configuration of peer registration modified. Anyway, I set a peer same like

> the image files that I attached it name is "test-gw-1.png". 





you need to go step by step. Follow the handbook, create you doain, create and

account, create two subscribers and call from one subscriber to the other. Once

you're there we'll talk about peering servers.



I think you are mixing too many things so either you follow the handbook

quick-start or you'll need to learn about Voip (or both).





cheers,



Jon



-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130321/c752c8f0/attachment-0001.html>

From jbonilla at sipwise.com  Thu Mar 21 06:24:36 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Thu, 21 Mar 2013 11:24:36 +0100
Subject: [Spce-user] Please let me
 knowwhyIfaced401/403(Unauthorized/Invalid To-URI Detected).
In-Reply-To: <514a7edf3fe1_@_imoxion.com>
References: <514a7edf3fe1_@_imoxion.com>
Message-ID: <20130321112436.5a1fcc54@quenya>

El Thu, 21 Mar 2013 12:24:34 +0900 (KST)
David Ho<rea1ity at kaist.ac.kr> escribi?:

>  
>  
> Jon,
>  
> I've already followed up the document that you provide "The sip:provider CE
> Handbook 2.7" so I'm followed setup 1 to 6 chapter and almost setting same
> like the document. And now I'm tried to modify values of Peering Servers
> several times and you could know where I'm modified the values at "6.4.4.
> Authenticating and Registering against Peering Servers". the URL following;
> http://www.sipwise.com/doc/2.7/spce/ar01s06.html#_authenticating_and_registering_against_peering_servers
> So manytimes, SIPWISE server denied my PSTN gateway caused by
> Unauthorized(401). And I found what is problemed via sems log
> file("/var/log/ngcp/sem/sems.log") it appeared error message before I
> mentioned same like following;


So you already created two subscribers and you ca call from one to the other?

If so, send us what you did to register against your peer server and all the
related information, network capture and configuration files. We cannot guess
what's going on without information.


> -----------------------------------------------------------------------------------------------------------------
> Mar 20 08:44:14 spce sems[2164]: [#7f601e77d700] [doRegistration,
> AmSipRegistration.cpp:139] ERROR: failed to send registration.
> -----------------------------------------------------------------------------------------------------------------
> So 'doRegistration function in AmSipRegistration.cpp file occurred ERROR
> message. If you were me, how could solve the error occurred? And it is
> another one that I afraid thing. the log files in SIPWISE as kamailio-lb.log
> and kamailio-options-proxy.log increased capacity too much. my
> kamailio-lb.log file capacity is over than 28MB and
> kamailio-options-proxy.log file capacity is more than 7MB. The reason of the
> log too much increased capacity caused by stacked approax. four to five log
> messages per a second! Is it alright to leave it even though it will be huge
> capacity log file after than this log file could occurr system error for leak
> of hard disk capacity. David


You can expect several GB of logs per day in a production system. 28MB is a
very small log. THe spce logs a lot of information for each request.




-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130321/a0e215ad/attachment-0001.asc>

From nh.developer at gmail.com  Thu Mar 21 07:29:09 2013
From: nh.developer at gmail.com (Developing Tomorrow)
Date: Thu, 21 Mar 2013 12:29:09 +0100
Subject: [Spce-user] Running Sipwise on Centos
Message-ID: <001c01ce2627$4eb52f00$ec1f8d00$@gmail.com>

Hi.,

 

I have tested sipwise on debian, as per the recommendation.

But the truth is that I prefer centos 6

 

Could someone direct me to the source code to compile or to some
instructions on 

 How to install it on centos?

 

Help would be much appreciated.

cheers

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130321/9f4c9e9f/attachment-0001.html>

From jbonilla at sipwise.com  Thu Mar 21 08:39:58 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Thu, 21 Mar 2013 13:39:58 +0100
Subject: [Spce-user] Running Sipwise on Centos
In-Reply-To: <001c01ce2627$4eb52f00$ec1f8d00$@gmail.com>
References: <001c01ce2627$4eb52f00$ec1f8d00$@gmail.com>
Message-ID: <20130321133958.7967731a@quenya>

El Thu, 21 Mar 2013 12:29:09 +0100
"Developing Tomorrow" <nh.developer at gmail.com> escribi?:

> Hi.,
> 
>  
> 
> I have tested sipwise on debian, as per the recommendation.
> 
> But the truth is that I prefer centos 6
> 
>  
> 
> Could someone direct me to the source code to compile or to some
> instructions on 
> 
>  How to install it on centos?
> 
>  
> 
> Help would be much appreciated.
> 


Sorry but I won't help you here. I don't know if it's even possible to compile
all the packages on centos and change the paths of the files that should be in
different places. This is more about religion, holy war, vim vs emacs or
whatever you'd like to call the rpm vs deb or debian vs any_other_distro.

Debian is the only distro we support and amd64 the only arch.  

This goes far away from what the spce mailing list can offer as support.

Maybe we could change the thread and talk about something more important: How
can anyone prefer Centos over Debian? 

:P


cheers,

Jon

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130321/ebacb4a7/attachment-0001.asc>

From alesg at coit.es  Thu Mar 21 11:43:58 2013
From: alesg at coit.es (Alejandro Sanchez Gimenez)
Date: Thu, 21 Mar 2013 16:43:58 +0100 (CET)
Subject: [Spce-user] Fwd: ngcp-fraud-auto-lock
In-Reply-To: <22675ad6-47d1-4690-ace6-dc6f663687db@mail.coit.es>
Message-ID: <84392596-5fc6-46c3-b4e2-c30c75f3b65b@mail.coit.es>

Hi again!! 


I?m still blocked at this point. I can?t get fraud auto-lock working. In my test configuration I have in my billing profile: 


Fraud Monthly Limit: 1 (this is "one cent", as I understand from documentation) 
Fraund Monthly Lock: incoming and outgoing 


I have only one catch-all fee for this billing profile, where init rate is 15 (cents), so in the first call, the account reaches its limit. I can see cdrs for the subscriber working correctly, with correct data in duration and charges. For example, for the subscriber?s first call: 


duration: 00:05 
charges: 0.15? 


At this point I execute "ngcp-fraud-auto-lock" but the subscriber can still make and receive calls. Where could be the problem? Rating is also working properly, because I have also tested with interval free time/cash and I get correct data. 


Thanks a lot. 


----- Mensaje original -----

De: "Alejandro Sanchez Gimenez" <alesg at coit.es> 
Para: spce-user at lists.sipwise.com 
Enviados: Martes, 26 de Febrero 2013 12:59:41 
Asunto: ngcp-fraud-auto-lock 


Hi all!! 


I have read it is possible to change the cronjob ngcp-fraud-auto-lock to be executed more than once a day. Would it be any problem if it is executed for exmple each five minutes? Could the system play any announcement if the subscriber?s balance is about to expire? 


Thanks a lot!! 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130321/63456217/attachment-0001.html>

From rfuchs at sipwise.com  Thu Mar 21 13:33:09 2013
From: rfuchs at sipwise.com (Richard Fuchs)
Date: Thu, 21 Mar 2013 13:33:09 -0400
Subject: [Spce-user] Fwd: ngcp-fraud-auto-lock
In-Reply-To: <84392596-5fc6-46c3-b4e2-c30c75f3b65b@mail.coit.es>
References: <84392596-5fc6-46c3-b4e2-c30c75f3b65b@mail.coit.es>
Message-ID: <514B4455.4090007@sipwise.com>

Hi,

Can you enter your mysql console and post the output of the following
commands, substituting $ID for the account ID of your subscriber?

> select * from billing.contracts where id = $ID;

> select * from billing.contract_balances where contract_id = $ID;

> select * from billing.billing_mappings m, billing.billing_profiles p
where p.id = m.billing_profile_id and m.contract_id = $ID;

Thanks


On 03/21/13 11:43, Alejandro Sanchez Gimenez wrote:
> Hi again!!
> 
> I?m still blocked at this point. I can?t get fraud auto-lock working. In
> my test configuration I have in my billing profile:
> 
> Fraud Monthly Limit: 1 (this is "one cent", as I understand from
> documentation)
> Fraund Monthly Lock: incoming and outgoing
> 
> I have only one catch-all fee for this billing profile, where init rate
> is 15 (cents), so in the first call, the account reaches its limit. I
> can see cdrs for the subscriber working correctly, with correct data in
> duration and charges. For example, for the subscriber?s first call:
> 
> duration: 00:05
> charges: 0.15?
> 
> At this point I execute "ngcp-fraud-auto-lock" but the subscriber can
> still make and receive calls. Where could be the problem? Rating is also
> working properly, because I have also tested with interval free
> time/cash and I get correct data.
> 
> Thanks a lot.
> 
> 
> ------------------------------------------------------------------------
> *De: *"Alejandro Sanchez Gimenez" <alesg at coit.es>
> *Para: *spce-user at lists.sipwise.com
> *Enviados: *Martes, 26 de Febrero 2013 12:59:41
> *Asunto: *ngcp-fraud-auto-lock
> 
> Hi all!!
> 
> I have read it is possible to change the cronjob ngcp-fraud-auto-lock to
> be executed more than once a day. Would it be any problem if it is
> executed for exmple each five minutes? Could the system play any
> announcement if the subscriber?s balance is about to expire?
> 
> Thanks a lot!!
> 
> 
> 
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 900 bytes
Desc: OpenPGP digital signature
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130321/e36cefe0/attachment-0001.asc>

From alesg at coit.es  Thu Mar 21 15:00:45 2013
From: alesg at coit.es (Alejandro Sanchez Gimenez)
Date: Thu, 21 Mar 2013 20:00:45 +0100 (CET)
Subject: [Spce-user]  Fwd: ngcp-fraud-auto-lock
Message-ID: <52a220ff-2caa-4f08-a955-667f985fb213@mail.coit.es>

Hello Richard, 


this is what I get: 



mysql> select * from contracts where id = 22; 
+----+-------------+-------------+------------+----------+--------+-------------+---------------------+---------------------+---------------------+---------------------+ 
| id | customer_id | reseller_id | contact_id | order_id | status | external_id | modify_timestamp | create_timestamp | activate_timestamp | terminate_timestamp | 
+----+-------------+-------------+------------+----------+--------+-------------+---------------------+---------------------+---------------------+---------------------+ 
| 22 | NULL | 1 | NULL | NULL | active | asdfasdfsdf | 2013-03-21 15:02:14 | 2013-03-21 15:02:14 | 2013-03-21 15:02:14 | NULL | 
+----+-------------+-------------+------------+----------+--------+-------------+---------------------+---------------------+---------------------+---------------------+ 
1 row in set (0.00 sec) 




mysql> select * from contract_balances where contract_id = 22; 
+----+-------------+--------------+-----------------------+-------------------+----------------------------+---------------------+---------------------+------------+ 
| id | contract_id | cash_balance | cash_balance_interval | free_time_balance | free_time_balance_interval | start | end | invoice_id | 
+----+-------------+--------------+-----------------------+-------------------+----------------------------+---------------------+---------------------+------------+ 
| 21 | 22 | 0 | 0 | 0 | 0 | 2013-03-01 00:00:00 | 2013-03-31 23:59:59 | NULL | 
+----+-------------+--------------+-----------------------+-------------------+----------------------------+---------------------+---------------------+------------+ 
1 row in set (0.00 sec) 




mysql> select * from billing.billing_mappings m, billing.billing_profiles p 
-> where p.id = m.billing_profile_id and m.contract_id = 22; 
+----+------------+----------+--------------------+-------------+------------+----+-------------+--------------+--------------+---------+-----------------+--------------------+--------------------+---------------+----------------+----------------------+---------------------+-------------------------------+-------------------+------------------+--------------------+----------+----------+--------------+ 
| id | start_date | end_date | billing_profile_id | contract_id | product_id | id | reseller_id | handle | name | prepaid | interval_charge | interval_free_time | interval_free_cash | interval_unit | interval_count | fraud_interval_limit | fraud_interval_lock | fraud_interval_notify | fraud_daily_limit | fraud_daily_lock | fraud_daily_notify | currency | vat_rate | vat_included | 
+----+------------+----------+--------------------+-------------+------------+----+-------------+--------------+--------------+---------+-----------------+--------------------+--------------------+---------------+----------------+----------------------+---------------------+-------------------------------+-------------------+------------------+--------------------+----------+----------+--------------+ 
| 22 | NULL | NULL | 4 | 22 | NULL | 4 | 1 | Tarifa_plana | Tarifa plana | 0 | 0 | 0 | 0 | month | 1 | 100 | 3 | | 100 | 3 | NULL | ?? | 0 | 0 | 
+----+------------+----------+--------------------+-------------+------------+----+-------------+--------------+--------------+---------+-----------------+--------------------+--------------------+---------------+----------------+----------------------+---------------------+-------------------------------+-------------------+------------------+--------------------+----------+----------+--------------+ 
1 row in set (0.00 sec) 




Thank you very much!!! 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130321/c59db2c1/attachment-0001.html>

From jbonilla at sipwise.com  Thu Mar 21 14:48:35 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Thu, 21 Mar 2013 19:48:35 +0100
Subject: [Spce-user] Fwd: ngcp-fraud-auto-lock
In-Reply-To: <52a220ff-2caa-4f08-a955-667f985fb213@mail.coit.es>
References: <52a220ff-2caa-4f08-a955-667f985fb213@mail.coit.es>
Message-ID: <20130321194835.0807d079@quenya>

El Thu, 21 Mar 2013 20:00:45 +0100 (CET)
Alejandro Sanchez Gimenez <alesg at coit.es> escribi?:

> Hello Richard, 
> 
> 
> this is what I get: 
> 

Please, can you repeat the same queries with \G at the end?

Like

mysql> select * from contracts where id = 22\G;



btw I don't know why your mail is not an answer to the one Richard sent. It's
in a new thread.

 

> 
> 
> 


> +----+-------------+-------------+------------+----------+--------+-------------+---------------------+---------------------+---------------------+---------------------+ 
> | id | customer_id | reseller_id | contact_id | order_id | status |
> external_id | modify_timestamp | create_timestamp | activate_timestamp |
> terminate_timestamp |
> +----+-------------+-------------+------------+----------+--------+-------------+---------------------+---------------------+---------------------+---------------------+
> | 22 | NULL | 1 | NULL | NULL | active | asdfasdfsdf | 2013-03-21 15:02:14 |
> 2013-03-21 15:02:14 | 2013-03-21 15:02:14 | NULL |
> +----+-------------+-------------+------------+----------+--------+-------------+---------------------+---------------------+---------------------+---------------------+
> 1 row in set (0.00 sec) 
> 
> 
> 
> 
> mysql> select * from contract_balances where contract_id = 22; 
> +----+-------------+--------------+-----------------------+-------------------+----------------------------+---------------------+---------------------+------------+ 
> | id | contract_id | cash_balance | cash_balance_interval | free_time_balance
> | free_time_balance_interval | start | end | invoice_id |
> +----+-------------+--------------+-----------------------+-------------------+----------------------------+---------------------+---------------------+------------+
> | 21 | 22 | 0 | 0 | 0 | 0 | 2013-03-01 00:00:00 | 2013-03-31 23:59:59 | NULL
> |
> +----+-------------+--------------+-----------------------+-------------------+----------------------------+---------------------+---------------------+------------+
> 1 row in set (0.00 sec) 
> 
> 
> 
> 
> mysql> select * from billing.billing_mappings m, billing.billing_profiles p 
> -> where p.id = m.billing_profile_id and m.contract_id = 22; 
> +----+------------+----------+--------------------+-------------+------------+----+-------------+--------------+--------------+---------+-----------------+--------------------+--------------------+---------------+----------------+----------------------+---------------------+-------------------------------+-------------------+------------------+--------------------+----------+----------+--------------+ 
> | id | start_date | end_date | billing_profile_id | contract_id | product_id
> | id | reseller_id | handle | name | prepaid | interval_charge |
> interval_free_time | interval_free_cash | interval_unit | interval_count |
> fraud_interval_limit | fraud_interval_lock | fraud_interval_notify |
> fraud_daily_limit | fraud_daily_lock | fraud_daily_notify | currency |
> vat_rate | vat_included |
> +----+------------+----------+--------------------+-------------+------------+----+-------------+--------------+--------------+---------+-----------------+--------------------+--------------------+---------------+----------------+----------------------+---------------------+-------------------------------+-------------------+------------------+--------------------+----------+----------+--------------+
> | 22 | NULL | NULL | 4 | 22 | NULL | 4 | 1 | Tarifa_plana | Tarifa plana | 0
> | 0 | 0 | 0 | month | 1 | 100 | 3 | | 100 | 3 | NULL | ?? | 0 | 0 |
> +----+------------+----------+--------------------+-------------+------------+----+-------------+--------------+--------------+---------+-----------------+--------------------+--------------------+---------------+----------------+----------------------+---------------------+-------------------------------+-------------------+------------------+--------------------+----------+----------+--------------+
> 1 row in set (0.00 sec) 
> 
> 
> 
> 
> Thank you very much!!! 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130321/c6bdda5b/attachment-0001.asc>

From rfuchs at sipwise.com  Thu Mar 21 15:19:07 2013
From: rfuchs at sipwise.com (Richard Fuchs)
Date: Thu, 21 Mar 2013 15:19:07 -0400
Subject: [Spce-user] Fwd: ngcp-fraud-auto-lock
In-Reply-To: <52a220ff-2caa-4f08-a955-667f985fb213@mail.coit.es>
References: <52a220ff-2caa-4f08-a955-667f985fb213@mail.coit.es>
Message-ID: <514B5D2B.30008@sipwise.com>

There's a problem with the ratings, the contract_balances don't show any
charges being made to the account, everything is zero. If the CDRs are
rated correctly, then the association between calling subscriber and
account must be incorrect. Check the source_account_id field of the
CDRs, it should contain 22, and the customer_cost field should contain
the cost.

BR


On 03/21/13 15:00, Alejandro Sanchez Gimenez wrote:
> Hello Richard,
> 
> this is what I get:
> 
> mysql> select * from contracts where id = 22;
> +----+-------------+-------------+------------+----------+--------+-------------+---------------------+---------------------+---------------------+---------------------+
> | id | customer_id | reseller_id | contact_id | order_id | status |
> external_id | modify_timestamp    | create_timestamp    |
> activate_timestamp  | terminate_timestamp |
> +----+-------------+-------------+------------+----------+--------+-------------+---------------------+---------------------+---------------------+---------------------+
> | 22 |        NULL |           1 |       NULL |     NULL | active |
> asdfasdfsdf | 2013-03-21 15:02:14 | 2013-03-21 15:02:14 | 2013-03-21
> 15:02:14 |                NULL |
> +----+-------------+-------------+------------+----------+--------+-------------+---------------------+---------------------+---------------------+---------------------+
> 1 row in set (0.00 sec)
> 
> 
> mysql> select * from contract_balances where contract_id = 22;
> +----+-------------+--------------+-----------------------+-------------------+----------------------------+---------------------+---------------------+------------+
> | id | contract_id | cash_balance | cash_balance_interval |
> free_time_balance | free_time_balance_interval | start               |
> end                 | invoice_id |
> +----+-------------+--------------+-----------------------+-------------------+----------------------------+---------------------+---------------------+------------+
> | 21 |          22 |            0 |                     0 |            
>     0 |                          0 | 2013-03-01 00:00:00 | 2013-03-31
> 23:59:59 |       NULL |
> +----+-------------+--------------+-----------------------+-------------------+----------------------------+---------------------+---------------------+------------+
> 1 row in set (0.00 sec)
> 
> 
> mysql> select * from billing.billing_mappings m, billing.billing_profiles p
>     -> where p.id = m.billing_profile_id and m.contract_id = 22;
> +----+------------+----------+--------------------+-------------+------------+----+-------------+--------------+--------------+---------+-----------------+--------------------+--------------------+---------------+----------------+----------------------+---------------------+-------------------------------+-------------------+------------------+--------------------+----------+----------+--------------+
> | id | start_date | end_date | billing_profile_id | contract_id |
> product_id | id | reseller_id | handle       | name         | prepaid |
> interval_charge | interval_free_time | interval_free_cash |
> interval_unit | interval_count | fraud_interval_limit |
> fraud_interval_lock | fraud_interval_notify         | fraud_daily_limit
> | fraud_daily_lock | fraud_daily_notify | currency | vat_rate |
> vat_included |
> +----+------------+----------+--------------------+-------------+------------+----+-------------+--------------+--------------+---------+-----------------+--------------------+--------------------+---------------+----------------+----------------------+---------------------+-------------------------------+-------------------+------------------+--------------------+----------+----------+--------------+
> | 22 | NULL       | NULL     |                  4 |          22 |      
> NULL |  4 |           1 | Tarifa_plana | Tarifa plana |       0 |      
>         0 |                  0 |                  0 | month         |  
>            1 |                  100 |                   3 |            
>                   |               100 |                3 | NULL        
>       | ??        |        0 |            0 |
> +----+------------+----------+--------------------+-------------+------------+----+-------------+--------------+--------------+---------+-----------------+--------------------+--------------------+---------------+----------------+----------------------+---------------------+-------------------------------+-------------------+------------------+--------------------+----------+----------+--------------+
> 1 row in set (0.00 sec)
> 
> 
> Thank you very much!!!
> 
> 
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 900 bytes
Desc: OpenPGP digital signature
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130321/d2ec5288/attachment-0001.asc>

From nh.developer at gmail.com  Thu Mar 21 17:41:48 2013
From: nh.developer at gmail.com (Developing Tomorrow)
Date: Thu, 21 Mar 2013 22:41:48 +0100
Subject: [Spce-user] Running Sipwise on Centos
Message-ID: <004801ce267c$e4ecd6d0$aec68470$@gmail.com>

HI Jon.
[] 
Thank you for that.
It was really deep.

I guess I will have to set for Debian or .......

Thank for the answer




From alesg at coit.es  Thu Mar 21 18:14:44 2013
From: alesg at coit.es (Alejandro Sanchez Gimenez)
Date: Thu, 21 Mar 2013 23:14:44 +0100 (CET)
Subject: [Spce-user]  Fwd: ngcp-fraud-auto-lock
In-Reply-To: <40c228be-4d1d-4f0a-b91f-677f44f52e55@mail.coit.es>
Message-ID: <4d3c7dda-2db9-450f-bd1f-3fb71ff4647e@mail.coit.es>

Hi again Richard!! 


I?ve checked accounting.cdr table and source_account_id field of the contains 22, and the customer_cost field contains the cost. Here you can see the query with relevant data: 



mysql> select * from cdr where source_account_id like "22"\G; 
*************************** 1. row *************************** 
id: 52 
update_time: 2013-03-21 15:03:32 
source_user_id: 635a08f3-3df3-4559-a6fd-d5e280d9761e 
source_provider_id: 1 
source_external_subscriber_id: XXXXXXXX 
source_external_contract_id: 
source_account_id: 22 
source_user: XXXXXXXXX 
source_domain: xxx.xxx.es 
source_cli: 34XXXXXXXXX 
source_clir: 0 
source_ip: 192.168.0.24 
destination_user_id: 43317f99-0859-4e89-a63b-ddf3d9bffb20 
destination_provider_id: 1 
destination_external_subscriber_id: XXXXXXXXX 
destination_external_contract_id: XXXXX 
destination_account_id: 8 
destination_user: XXXXXXXXX 
destination_domain: xxx.xxx.es 
destination_user_dialed: XXXXXXXXX 
destination_user_in: XXXXXXXXX 
destination_domain_in: xxx.xxx.es 
peer_auth_user: 
peer_auth_realm: 
call_type: call 
call_status: ok 
call_code: 200 
init_time: 1363874581.577 
start_time: 1363874584.840 
duration: 15.680 
call_id: OWIzN2RmY2M5M2RlMmFmMWU1MzVhNzkyNTVkNzI2YmM. 
carrier_cost: 0.00 
reseller_cost: 0.00 
customer_cost: 16.50 
carrier_free_time: 0 
reseller_free_time: 0 
customer_free_time: 0 
carrier_billing_fee_id: NULL 
reseller_billing_fee_id: NULL 
customer_billing_fee_id: 7 
carrier_billing_zone_id: NULL 
reseller_billing_zone_id: NULL 
customer_billing_zone_id: 7 
frag_carrier_onpeak: NULL 
frag_reseller_onpeak: NULL 
frag_customer_onpeak: NULL 
is_fragmented: NULL 
split: 0 
rated_at: 2013-03-21 15:03:32 
rating_status: ok 
exported_at: NULL 
export_status: unexported 


Jon, I paste the query with \G: 



mysql> select * from contracts where id = 22\G; 
*************************** 1. row *************************** 
id: 22 
customer_id: NULL 
reseller_id: 1 
contact_id: NULL 
order_id: NULL 
status: active 
external_id: 
modify_timestamp: 2013-03-21 15:02:14 
create_timestamp: 2013-03-21 15:02:14 
activate_timestamp: 2013-03-21 15:02:14 
terminate_timestamp: NULL 
1 row in set (0.00 sec) 


mysql> select * from contract_balances where contract_id = 22\G; 
*************************** 1. row *************************** 
id: 21 
contract_id: 22 
cash_balance: 0 
cash_balance_interval: 0 
free_time_balance: 0 
free_time_balance_interval: 0 
start: 2013-03-01 00:00:00 
end: 2013-03-31 23:59:59 
invoice_id: NULL 
1 row in set (0.00 sec) 


mysql> select * from billing.billing_mappings m, billing.billing_profiles p 
-> where p.id = m.billing_profile_id and m.contract_id = 22\G; 
*************************** 1. row *************************** 
id: 22 
start_date: NULL 
end_date: NULL 
billing_profile_id: 4 
contract_id: 22 
product_id: NULL 
id: 4 
reseller_id: 1 
handle: Tarifa_plana 
name: Tarifa plana 
prepaid: 0 
interval_charge: 0 
interval_free_time: 0 
interval_free_cash: 0 
interval_unit: month 
interval_count: 1 
fraud_interval_limit: 100 
fraud_interval_lock: 3 
fraud_interval_notify: 
fraud_daily_limit: 100 
fraud_daily_lock: 3 
fraud_daily_notify: NULL 
currency: ?? 
vat_rate: 0 
vat_included: 0 
1 row in set (0.00 sec) 


Thank you very much!! 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130321/df7f1dc7/attachment-0001.html>

From jbonilla at sipwise.com  Thu Mar 21 17:58:07 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Thu, 21 Mar 2013 22:58:07 +0100
Subject: [Spce-user] Running Sipwise on Centos
In-Reply-To: <004801ce267c$e4ecd6d0$aec68470$@gmail.com>
References: <004801ce267c$e4ecd6d0$aec68470$@gmail.com>
Message-ID: <20130321225807.15a56dcb@quenya>

El Thu, 21 Mar 2013 22:41:48 +0100
"Developing Tomorrow" <nh.developer at gmail.com> escribi?:

> HI Jon.
> [] 
> Thank you for that.
> It was really deep.
> 
> I guess I will have to set for Debian or .......
> 
> Thank for the answer
> 
>

did you use the "answer" button to reply my message? It appears as a new thread
to me.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130321/571067a1/attachment-0001.asc>

From jbonilla at sipwise.com  Thu Mar 21 17:58:33 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Thu, 21 Mar 2013 22:58:33 +0100
Subject: [Spce-user] Fwd: ngcp-fraud-auto-lock
In-Reply-To: <4d3c7dda-2db9-450f-bd1f-3fb71ff4647e@mail.coit.es>
References: <40c228be-4d1d-4f0a-b91f-677f44f52e55@mail.coit.es>
	<4d3c7dda-2db9-450f-bd1f-3fb71ff4647e@mail.coit.es>
Message-ID: <20130321225833.342c05c1@quenya>

El Thu, 21 Mar 2013 23:14:44 +0100 (CET)
Alejandro Sanchez Gimenez <alesg at coit.es> escribi?:

> Hi again Richard!! 
> 
> 
> I?ve checked accounting.cdr table and source_account_id field of the contains
> 22, and the customer_cost field contains the cost. Here you can see the query
> with relevant data: 
> 
> 

did you use the "answer" button to reply the last message? It appears as a new
thread to me.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130321/746d2f7f/attachment-0001.asc>

From alesg at coit.es  Thu Mar 21 18:24:04 2013
From: alesg at coit.es (Alejandro Sanchez Gimenez)
Date: Thu, 21 Mar 2013 23:24:04 +0100 (CET)
Subject: [Spce-user] Fwd: ngcp-fraud-auto-lock
In-Reply-To: <20130321225833.342c05c1@quenya>
Message-ID: <b812af0c-3ced-41cf-9e33-a8f4a293b96f@mail.coit.es>

Sorry Jon, I didn?t.

----- Mensaje original -----
De: "Jon Bonilla" <jbonilla at sipwise.com>
Para: spce-user at lists.sipwise.com
Enviados: Jueves, 21 de Marzo 2013 22:58:33
Asunto: Re: [Spce-user] Fwd: ngcp-fraud-auto-lock

El Thu, 21 Mar 2013 23:14:44 +0100 (CET)
Alejandro Sanchez Gimenez <alesg at coit.es> escribi?:

> Hi again Richard!! 
> 
> 
> I?ve checked accounting.cdr table and source_account_id field of the contains
> 22, and the customer_cost field contains the cost. Here you can see the query
> with relevant data: 
> 
> 

did you use the "answer" button to reply the last message? It appears as a new
thread to me.


_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com
http://lists.sipwise.com/listinfo/spce-user



From rea1ity at kaist.ac.kr  Fri Mar 22 00:55:39 2013
From: rea1ity at kaist.ac.kr (=?UTF-8?B?RGF2aWQgSG8=?=)
Date: Fri, 22 Mar 2013 13:55:39 +0900 (KST)
Subject: [Spce-user]
	=?utf-8?q?Please_let_meknowwhyIfaced401/403=28Unautho?=
	=?utf-8?q?rized/Invalid_To-URI_Detected=29=2E?=
Message-ID: <514be7813fe6_@_imoxion.com>

Jon,
 
I was succeed to register with my PSTN gateway last night.
The reason I couldn't register to my PSTN gateway was have no peer about incorrectly ID/PW because my PSTN gateway ID only gets number(i.e. 0 to 9).
And my PSTN GW required three kinds of values it is 'Represent telephone number', 'ID','PW'.
So I'm put in the values to there that I set on SIPWISE peer same as ID and PW.
But the one of value as 'Represent telephone number' used at ID on SIPWISE.(It caused confused me.)
So when I tried input a same value as 'Represent TEL No' and 'ID', The PSTN GW as well registered on SIPWISE server. (hooray!!)
It so hard time for me because its a first time to set VoIP system it configured SIP server and registering with third party PSTN GW.
But It is alright. thank you for your support.
Registered problem was cleared.
 
And it is another one of the log capacities.
As you mentioned the log files capacity can increased over the several GB.
If I want to modified the log files capacity has limited as several MB, how can I do that?
I didn't want to it would be problemed that SIP server hard drive capacity was full of log file so it occurred system error.
 
David
 


-----------------------????-----------------------
???? : "Jon Bonilla (Manwe)" <jbonilla at sipwise.com>
????: "spce-user at lists.sipwise.com" <spce-user at lists.sipwise.com>
? ?: David Ho <rea1ity at kaist.ac.kr>
????: 2013-03-21 19:24:36 GMT +0900 (Asia/Seoul)
??: Re: [Spce-user] Please let meknowwhyIfaced401/403(Unauthorized/Invalid To-URI Detected).

El Thu, 21 Mar 2013 12:24:34 +0900 (KST)

David Ho<rea1ity at kaist.ac.kr> escribi?:



>  

>  

> Jon,

>  

> I've already followed up the document that you provide "The sip:provider CE

> Handbook 2.7" so I'm followed setup 1 to 6 chapter and almost setting same

> like the document. And now I'm tried to modify values of Peering Servers

> several times and you could know where I'm modified the values at "6.4.4.

> Authenticating and Registering against Peering Servers". the URL following;

> http://www.sipwise.com/doc/2.7/spce/ar01s06.html#_authenticating_and_registering_against_peering_servers

> So manytimes, SIPWISE server denied my PSTN gateway caused by

> Unauthorized(401). And I found what is problemed via sems log

> file("/var/log/ngcp/sem/sems.log") it appeared error message before I

> mentioned same like following;





So you already created two subscribers and you ca call from one to the other?



If so, send us what you did to register against your peer server and all the

related information, network capture and configuration files. We cannot guess

what's going on without information.





> -----------------------------------------------------------------------------------------------------------------

> Mar 20 08:44:14 spce sems[2164]: [#7f601e77d700] [doRegistration,

> AmSipRegistration.cpp:139] ERROR: failed to send registration.

> -----------------------------------------------------------------------------------------------------------------

> So 'doRegistration function in AmSipRegistration.cpp file occurred ERROR

> message. If you were me, how could solve the error occurred? And it is

> another one that I afraid thing. the log files in SIPWISE as kamailio-lb.log

> and kamailio-options-proxy.log increased capacity too much. my

> kamailio-lb.log file capacity is over than 28MB and

> kamailio-options-proxy.log file capacity is more than 7MB. The reason of the

> log too much increased capacity caused by stacked approax. four to five log

> messages per a second! Is it alright to leave it even though it will be huge

> capacity log file after than this log file could occurr system error for leak

> of hard disk capacity. David





You can expect several GB of logs per day in a production system. 28MB is a

very small log. THe spce logs a lot of information for each request.







-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130322/06fa8722/attachment-0001.html>

From agranig at sipwise.com  Fri Mar 22 03:26:05 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Fri, 22 Mar 2013 08:26:05 +0100
Subject: [Spce-user] Please let
 meknowwhyIfaced401/403(Unauthorized/Invalid To-URI Detected).
In-Reply-To: <514be7813fe6_@_imoxion.com>
References: <514be7813fe6_@_imoxion.com>
Message-ID: <514C078D.80102@sipwise.com>

Hi,

On 03/22/2013 05:55 AM, David Ho wrote:
> And it is another one of the log capacities.
>
> As you mentioned the log files capacity can increased over the several GB.
>
> If I want to modified the log files capacity has limited as several MB,
> how can I do that?
>
> I didn't want to it would be problemed that SIP server hard drive
> capacity was full of log file so it occurred system error.

It's true that the log files (especially kamailio-proxy.log and 
kamailio-lb.log) are pretty extensive, and can increase over several 
gigs per day. However, you'll appreciate that when you run into a 
customer issue for the first time, where you'll be able to clearly 
comprehend what was going on. And it's not like the SPCE is a tiny 
system, so be prepared for quite some data (e.g. you get CDRs for each 
call, which will be kept for 3 months by default, then backed up to 
/var/backups/cdr for 12 months by default etc.

Also note that log files get rotated and compressed nightly, and are 
kept for however long you choose to (3 months by default). You can 
fine-tune that by copying 
/etc/ngcp-config/templates/etc/logrotate.d/ngcp.tt2 to 
/etc/ngcp-config/templates/etc/logrotate.d/ngcp.customtt.tt2, adapt it 
to your needs and execute "ngcpcfg apply" afterwards.

That's the preferred way to keep your used disk space low, instead of 
starting to reduce log messages.

Andreas



From alesg at coit.es  Fri Mar 22 09:14:48 2013
From: alesg at coit.es (Alejandro Sanchez Gimenez)
Date: Fri, 22 Mar 2013 14:14:48 +0100 (CET)
Subject: [Spce-user] Fwd: ngcp-fraud-auto-lock
In-Reply-To: <b812af0c-3ced-41cf-9e33-a8f4a293b96f@mail.coit.es>
Message-ID: <c12b2fa4-a0a6-4c42-bdff-24d2bce3a7fb@mail.coit.es>

I think that the problem reported by Mario Contreras On 07/16/12 is still happening: "I have tested, and I can asume it works like I supposed. The fraud only compares de free cash you have assigned to the account...". As he said, if I assign free cash to the account, fraud lock works properly. If I don?t assign free cash account balance doesn?t increase, so the comparison in /usr/sbin/ngcp-fraud-auto-lock: "b.cash_balance_interval >= p.fraud_interval_limit" never happens. It was supposed to be fixed in 2.6. Is this correct?

Thank you very much!!

----- Mensaje original -----
De: "Alejandro Sanchez Gimenez" <alesg at coit.es>
Para: "Jon Bonilla" <jbonilla at sipwise.com>
CC: spce-user at lists.sipwise.com
Enviados: Jueves, 21 de Marzo 2013 23:24:04
Asunto: Re: [Spce-user] Fwd: ngcp-fraud-auto-lock

Sorry Jon, I didn?t.

----- Mensaje original -----
De: "Jon Bonilla" <jbonilla at sipwise.com>
Para: spce-user at lists.sipwise.com
Enviados: Jueves, 21 de Marzo 2013 22:58:33
Asunto: Re: [Spce-user] Fwd: ngcp-fraud-auto-lock

El Thu, 21 Mar 2013 23:14:44 +0100 (CET)
Alejandro Sanchez Gimenez <alesg at coit.es> escribi?:

> Hi again Richard!! 
> 
> 
> I?ve checked accounting.cdr table and source_account_id field of the contains
> 22, and the customer_cost field contains the cost. Here you can see the query
> with relevant data: 
> 
> 

did you use the "answer" button to reply the last message? It appears as a new
thread to me.


_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com
http://lists.sipwise.com/listinfo/spce-user

_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com
http://lists.sipwise.com/listinfo/spce-user



From vermeulen.deon at gmail.com  Fri Mar 22 09:16:19 2013
From: vermeulen.deon at gmail.com (Deon Vermeulen)
Date: Fri, 22 Mar 2013 15:16:19 +0200
Subject: [Spce-user] Invoicing and Billing Peers/Trunks
Message-ID: <514C59A3.7090101@gmail.com>

Good Day

Is there any possibility that one could generate an Invoice to customers 
with a Trunk to SPCE 2.7?

Is there any possibility to Generate a Bill to a Peering Partner for 
calls sent to SPCE 2.7. i.e Reverse billing?


Thank you.

KR
Deon
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130322/d84e9e77/attachment-0001.html>

From rfuchs at sipwise.com  Fri Mar 22 09:27:54 2013
From: rfuchs at sipwise.com (Richard Fuchs)
Date: Fri, 22 Mar 2013 09:27:54 -0400
Subject: [Spce-user] Fwd: ngcp-fraud-auto-lock
In-Reply-To: <c12b2fa4-a0a6-4c42-bdff-24d2bce3a7fb@mail.coit.es>
References: <c12b2fa4-a0a6-4c42-bdff-24d2bce3a7fb@mail.coit.es>
Message-ID: <514C5C5A.4090800@sipwise.com>

Can you check which version of rate-o-mat is installed?

dpkg -l ngcp-rate-o-mat


On 03/22/13 09:14, Alejandro Sanchez Gimenez wrote:
> I think that the problem reported by Mario Contreras On 07/16/12 is still happening: "I have tested, and I can asume it works like I supposed. The fraud only compares de free cash you have assigned to the account...". As he said, if I assign free cash to the account, fraud lock works properly. If I don?t assign free cash account balance doesn?t increase, so the comparison in /usr/sbin/ngcp-fraud-auto-lock: "b.cash_balance_interval >= p.fraud_interval_limit" never happens. It was supposed to be fixed in 2.6. Is this correct?
> 
> Thank you very much!!
> 
> ----- Mensaje original -----
> De: "Alejandro Sanchez Gimenez" <alesg at coit.es>
> Para: "Jon Bonilla" <jbonilla at sipwise.com>
> CC: spce-user at lists.sipwise.com
> Enviados: Jueves, 21 de Marzo 2013 23:24:04
> Asunto: Re: [Spce-user] Fwd: ngcp-fraud-auto-lock
> 
> Sorry Jon, I didn?t.
> 
> ----- Mensaje original -----
> De: "Jon Bonilla" <jbonilla at sipwise.com>
> Para: spce-user at lists.sipwise.com
> Enviados: Jueves, 21 de Marzo 2013 22:58:33
> Asunto: Re: [Spce-user] Fwd: ngcp-fraud-auto-lock
> 
> El Thu, 21 Mar 2013 23:14:44 +0100 (CET)
> Alejandro Sanchez Gimenez <alesg at coit.es> escribi?:
> 
>> Hi again Richard!! 
>>
>>
>> I?ve checked accounting.cdr table and source_account_id field of the contains
>> 22, and the customer_cost field contains the cost. Here you can see the query
>> with relevant data: 
>>
>>
> 
> did you use the "answer" button to reply the last message? It appears as a new
> thread to me.
> 
> 
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
> 
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
> 
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 900 bytes
Desc: OpenPGP digital signature
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130322/fa313295/attachment-0001.asc>

From alesg at coit.es  Fri Mar 22 09:56:53 2013
From: alesg at coit.es (Alejandro Sanchez Gimenez)
Date: Fri, 22 Mar 2013 14:56:53 +0100 (CET)
Subject: [Spce-user] Fwd: ngcp-fraud-auto-lock
In-Reply-To: <514C5C5A.4090800@sipwise.com>
Message-ID: <1c03da20-53f6-4e8b-8bc0-b5927529c899@mail.coit.es>

Hello Richard,

The version installed is 1.4.0

Thanks

----- Mensaje original -----
De: "Richard Fuchs" <rfuchs at sipwise.com>
Para: spce-user at lists.sipwise.com
Enviados: Viernes, 22 de Marzo 2013 14:27:54
Asunto: Re: [Spce-user] Fwd: ngcp-fraud-auto-lock

Can you check which version of rate-o-mat is installed?

dpkg -l ngcp-rate-o-mat


On 03/22/13 09:14, Alejandro Sanchez Gimenez wrote:
> I think that the problem reported by Mario Contreras On 07/16/12 is still happening: "I have tested, and I can asume it works like I supposed. The fraud only compares de free cash you have assigned to the account...". As he said, if I assign free cash to the account, fraud lock works properly. If I don?t assign free cash account balance doesn?t increase, so the comparison in /usr/sbin/ngcp-fraud-auto-lock: "b.cash_balance_interval >= p.fraud_interval_limit" never happens. It was supposed to be fixed in 2.6. Is this correct?
> 
> Thank you very much!!
> 
> ----- Mensaje original -----
> De: "Alejandro Sanchez Gimenez" <alesg at coit.es>
> Para: "Jon Bonilla" <jbonilla at sipwise.com>
> CC: spce-user at lists.sipwise.com
> Enviados: Jueves, 21 de Marzo 2013 23:24:04
> Asunto: Re: [Spce-user] Fwd: ngcp-fraud-auto-lock
> 
> Sorry Jon, I didn?t.
> 
> ----- Mensaje original -----
> De: "Jon Bonilla" <jbonilla at sipwise.com>
> Para: spce-user at lists.sipwise.com
> Enviados: Jueves, 21 de Marzo 2013 22:58:33
> Asunto: Re: [Spce-user] Fwd: ngcp-fraud-auto-lock
> 
> El Thu, 21 Mar 2013 23:14:44 +0100 (CET)
> Alejandro Sanchez Gimenez <alesg at coit.es> escribi?:
> 
>> Hi again Richard!! 
>>
>>
>> I?ve checked accounting.cdr table and source_account_id field of the contains
>> 22, and the customer_cost field contains the cost. Here you can see the query
>> with relevant data: 
>>
>>
> 
> did you use the "answer" button to reply the last message? It appears as a new
> thread to me.
> 
> 
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
> 
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
> 
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user


_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com
http://lists.sipwise.com/listinfo/spce-user



From rfuchs at sipwise.com  Fri Mar 22 10:30:06 2013
From: rfuchs at sipwise.com (Richard Fuchs)
Date: Fri, 22 Mar 2013 10:30:06 -0400
Subject: [Spce-user] Fwd: ngcp-fraud-auto-lock
In-Reply-To: <1c03da20-53f6-4e8b-8bc0-b5927529c899@mail.coit.es>
References: <1c03da20-53f6-4e8b-8bc0-b5927529c899@mail.coit.es>
Message-ID: <514C6AEE.7050705@sipwise.com>

Hey,

Turns out that this was fixed last July, but then broken again in
October. We'll have a (new) fix for it shortly.

cheers


On 03/22/13 09:56, Alejandro Sanchez Gimenez wrote:
> Hello Richard,
> 
> The version installed is 1.4.0
> 
> Thanks
> 
> ----- Mensaje original -----
> De: "Richard Fuchs" <rfuchs at sipwise.com>
> Para: spce-user at lists.sipwise.com
> Enviados: Viernes, 22 de Marzo 2013 14:27:54
> Asunto: Re: [Spce-user] Fwd: ngcp-fraud-auto-lock
> 
> Can you check which version of rate-o-mat is installed?
> 
> dpkg -l ngcp-rate-o-mat
> 
> 
> On 03/22/13 09:14, Alejandro Sanchez Gimenez wrote:
>> I think that the problem reported by Mario Contreras On 07/16/12 is still happening: "I have tested, and I can asume it works like I supposed. The fraud only compares de free cash you have assigned to the account...". As he said, if I assign free cash to the account, fraud lock works properly. If I don?t assign free cash account balance doesn?t increase, so the comparison in /usr/sbin/ngcp-fraud-auto-lock: "b.cash_balance_interval >= p.fraud_interval_limit" never happens. It was supposed to be fixed in 2.6. Is this correct?
>>
>> Thank you very much!!
>>
>> ----- Mensaje original -----
>> De: "Alejandro Sanchez Gimenez" <alesg at coit.es>
>> Para: "Jon Bonilla" <jbonilla at sipwise.com>
>> CC: spce-user at lists.sipwise.com
>> Enviados: Jueves, 21 de Marzo 2013 23:24:04
>> Asunto: Re: [Spce-user] Fwd: ngcp-fraud-auto-lock
>>
>> Sorry Jon, I didn?t.
>>
>> ----- Mensaje original -----
>> De: "Jon Bonilla" <jbonilla at sipwise.com>
>> Para: spce-user at lists.sipwise.com
>> Enviados: Jueves, 21 de Marzo 2013 22:58:33
>> Asunto: Re: [Spce-user] Fwd: ngcp-fraud-auto-lock
>>
>> El Thu, 21 Mar 2013 23:14:44 +0100 (CET)
>> Alejandro Sanchez Gimenez <alesg at coit.es> escribi?:
>>
>>> Hi again Richard!! 
>>>
>>>
>>> I?ve checked accounting.cdr table and source_account_id field of the contains
>>> 22, and the customer_cost field contains the cost. Here you can see the query
>>> with relevant data: 
>>>
>>>
>>
>> did you use the "answer" button to reply the last message? It appears as a new
>> thread to me.
>>
>>
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/listinfo/spce-user
>>
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/listinfo/spce-user
>>
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/listinfo/spce-user
> 
> 
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 900 bytes
Desc: OpenPGP digital signature
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130322/97815cab/attachment-0001.asc>

From gavin.sweet at skyracktelecom.com  Fri Mar 22 12:00:26 2013
From: gavin.sweet at skyracktelecom.com (Gavin Sweet)
Date: Fri, 22 Mar 2013 16:00:26 -0000
Subject: [Spce-user] Inbound CDR tariffs?
Message-ID: <113701ce2716$61b99880$252cc980$@sweet@skyracktelecom.com>

Hi - 

We have inbound CDR exporting switched on, and I had a couple of questions,
as we'd ideally like to be able to actually rate (apply tariff with cost) to
certain inbound calls, because they carry a cost to us (we have a mobile
network integrated for which there are transit charges in both directions).

1. It actually appears that all inbound calls are classified with
destinations and rates that are not in the billing profile fees? Can this be
modified?

E.g. where a call outbound to a specific mobile picks up the appropriate
destinations from billing fees:
carrier_zone	customer_zone	carrier_destination	customer_destination
UKN			UKN			UKNFM1
UKNFM1

However, calls *inbound* from that same mobile to the same user show:
carrier_zone	customer_zone	carrier_destination	customer_destination
onnet						platform internal	


2. Is source_account_id = 0 fully reliable as an indicator in a CDR file
that the call was from an external user? does anything break that
assumption?


Thanks as ever.

Gavin




From alesg at coit.es  Fri Mar 22 12:36:17 2013
From: alesg at coit.es (Alejandro Sanchez Gimenez)
Date: Fri, 22 Mar 2013 17:36:17 +0100 (CET)
Subject: [Spce-user] Fwd: ngcp-fraud-auto-lock
In-Reply-To: <514C6AEE.7050705@sipwise.com>
Message-ID: <b48c9ac9-ded7-4130-af92-b0106adb36a1@mail.coit.es>

Ok Richard, thank you very much.

Will the fix be available in a new version of sipwise:ce or via patch?

Regards



----- Mensaje original -----
De: Richard Fuchs <rfuchs at sipwise.com>
Para: spce-user at lists.sipwise.com
Enviado: Fri, 22 Mar 2013 15:30:06 +0100 (CET)
Asunto: Re: [Spce-user] Fwd: ngcp-fraud-auto-lock

Hey,

Turns out that this was fixed last July, but then broken again in
October. We'll have a (new) fix for it shortly.

cheers


On 03/22/13 09:56, Alejandro Sanchez Gimenez wrote:
> Hello Richard,
> 
> The version installed is 1.4.0
> 
> Thanks
> 
> ----- Mensaje original -----
> De: "Richard Fuchs" <rfuchs at sipwise.com>
> Para: spce-user at lists.sipwise.com
> Enviados: Viernes, 22 de Marzo 2013 14:27:54
> Asunto: Re: [Spce-user] Fwd: ngcp-fraud-auto-lock
> 
> Can you check which version of rate-o-mat is installed?
> 
> dpkg -l ngcp-rate-o-mat
> 
> 
> On 03/22/13 09:14, Alejandro Sanchez Gimenez wrote:
>> I think that the problem reported by Mario Contreras On 07/16/12 is still happening: "I have tested, and I can asume it works like I supposed. The fraud only compares de free cash you have assigned to the account...". As he said, if I assign free cash to the account, fraud lock works properly. If I don?t assign free cash account balance doesn?t increase, so the comparison in /usr/sbin/ngcp-fraud-auto-lock: "b.cash_balance_interval >= p.fraud_interval_limit" never happens. It was supposed to be fixed in 2.6. Is this correct?
>>
>> Thank you very much!!
>>
>> ----- Mensaje original -----
>> De: "Alejandro Sanchez Gimenez" <alesg at coit.es>
>> Para: "Jon Bonilla" <jbonilla at sipwise.com>
>> CC: spce-user at lists.sipwise.com
>> Enviados: Jueves, 21 de Marzo 2013 23:24:04
>> Asunto: Re: [Spce-user] Fwd: ngcp-fraud-auto-lock
>>
>> Sorry Jon, I didn?t.
>>
>> ----- Mensaje original -----
>> De: "Jon Bonilla" <jbonilla at sipwise.com>
>> Para: spce-user at lists.sipwise.com
>> Enviados: Jueves, 21 de Marzo 2013 22:58:33
>> Asunto: Re: [Spce-user] Fwd: ngcp-fraud-auto-lock
>>
>> El Thu, 21 Mar 2013 23:14:44 +0100 (CET)
>> Alejandro Sanchez Gimenez <alesg at coit.es> escribi?:
>>
>>> Hi again Richard!! 
>>>
>>>
>>> I?ve checked accounting.cdr table and source_account_id field of the contains
>>> 22, and the customer_cost field contains the cost. Here you can see the query
>>> with relevant data: 
>>>
>>>
>>
>> did you use the "answer" button to reply the last message? It appears as a new
>> thread to me.
>>
>>
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/listinfo/spce-user
>>
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/listinfo/spce-user
>>
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/listinfo/spce-user
> 
> 
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user





From agranig at sipwise.com  Fri Mar 22 12:44:45 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Fri, 22 Mar 2013 17:44:45 +0100
Subject: [Spce-user] Inbound CDR tariffs?
In-Reply-To: <113701ce2716$61b99880$252cc980$@sweet@skyracktelecom.com>
References: <113701ce2716$61b99880$252cc980$@sweet@skyracktelecom.com>
Message-ID: <514C8A7D.2080009@sipwise.com>

Hi,

On 03/22/2013 05:00 PM, Gavin Sweet wrote:
> 1. It actually appears that all inbound calls are classified with
> destinations and rates that are not in the billing profile fees? Can this be
> modified?

Inbound calls are not rated at all in 2.7. You've to wait for another 
week or so for 2.8 to come out :)


> 2. Is source_account_id = 0 fully reliable as an indicator in a CDR file
> that the call was from an external user? does anything break that
> assumption?

Should do the trick, but I'd go for source_user_id instead, which is 
more obvious.

Andreas



From gavin.sweet at skyracktelecom.com  Fri Mar 22 13:13:57 2013
From: gavin.sweet at skyracktelecom.com (Gavin Sweet)
Date: Fri, 22 Mar 2013 17:13:57 -0000
Subject: [Spce-user] Inbound CDR tariffs?
In-Reply-To: <514C8A7D.2080009@sipwise.com>
References: <113701ce2716$61b99880$252cc980$@sweet@skyracktelecom.com>
	<514C8A7D.2080009@sipwise.com>
Message-ID: <114a01ce2720$a3ba5ee0$eb2f1ca0$@sweet@skyracktelecom.com>

Thanks Andreas, look forward to that.
As soon as you have the new feature/fix list, would be great to see it.

Cheers,
Gavin

> -----Original Message-----
> From: spce-user-bounces at lists.sipwise.com [mailto:spce-user-
> bounces at lists.sipwise.com] On Behalf Of Andreas Granig
> Sent: 22 March 2013 16:45
> To: spce-user at lists.sipwise.com
> Subject: Re: [Spce-user] Inbound CDR tariffs?
> 
> Hi,
> 
> On 03/22/2013 05:00 PM, Gavin Sweet wrote:
> > 1. It actually appears that all inbound calls are classified with
> > destinations and rates that are not in the billing profile fees? Can
> > this be modified?
> 
> Inbound calls are not rated at all in 2.7. You've to wait for another
> week or so for 2.8 to come out :)
> 
> 
> > 2. Is source_account_id = 0 fully reliable as an indicator in a CDR
> file
> > that the call was from an external user? does anything break that
> > assumption?
> 
> Should do the trick, but I'd go for source_user_id instead, which is
> more obvious.
> 
> Andreas
> 
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
> -----
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2013.0.2904 / Virus Database: 2641/6166 - Release Date:
> 03/12/13
> Internal Virus Database is out of date.




From wfulton at thirdhatch.com  Fri Mar 22 19:27:39 2013
From: wfulton at thirdhatch.com (William Fulton)
Date: Fri, 22 Mar 2013 16:27:39 -0700
Subject: [Spce-user] Sipwise CE 2.7 SBC config revisited
Message-ID: <0F449623D8AAB145A3278882C750BA5831C713@sea2k3ex06.thirdhatch.corp>

Guys,

 

I have spent a couple of weeks trying to make this work with absolutely
no luck.  I have read the manual cover to cover and there is clearly a
section missing.  Section 6 references a section in Section 7 about
advanced network configuration that does not exist.

 

I have two interfaces:

 

eth0 and eth1

 

*         eth0 is on the external subnet and this configuration was easy
as I simply assigned the sip_ext, web_ext, ssh_ext, etc to this adaptor.

*         eth1 is on the internal subnet and I have added the role of
sip_int to this adaptor.  

*         I created a new subscriber and set the preferences to
authenticate to the peer and set the peer username, password and realm

*         Despite my best efforts, I cannot get the subscriber external
to the Sipwise CE system to authenticate to the Asterisk system internal
to the Sipwise CE system.  

 

What am I missing?  I'm sure it's pretty simple but I can't see it.

 

Thanks,

William

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130322/743fedd6/attachment-0001.html>

From jbonilla at sipwise.com  Sat Mar 23 06:13:16 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Sat, 23 Mar 2013 11:13:16 +0100
Subject: [Spce-user] Sipwise CE 2.7 SBC config revisited
In-Reply-To: <0F449623D8AAB145A3278882C750BA5831C713@sea2k3ex06.thirdhatch.corp>
References: <0F449623D8AAB145A3278882C750BA5831C713@sea2k3ex06.thirdhatch.corp>
Message-ID: <20130323111316.773fab5b@quenya>

El Fri, 22 Mar 2013 16:27:39 -0700
"William Fulton" <wfulton at thirdhatch.com> escribi?:

> Guys,
> 
>  
> 
> I have spent a couple of weeks trying to make this work with absolutely
> no luck.  I have read the manual cover to cover and there is clearly a
> section missing.  Section 6 references a section in Section 7 about
> advanced network configuration that does not exist.
> 
>  
> 
> I have two interfaces:
> 
>  
> 
> eth0 and eth1
> 
>  
> 
> *         eth0 is on the external subnet and this configuration was easy
> as I simply assigned the sip_ext, web_ext, ssh_ext, etc to this adaptor.
> 
> *         eth1 is on the internal subnet and I have added the role of
> sip_int to this adaptor.  


why did you do this? Why not lo interface for this role?

> 
> *         I created a new subscriber and set the preferences to
> authenticate to the peer and set the peer username, password and realm

Did you also check the "peer_auth_register" option?


> 
> *         Despite my best efforts, I cannot get the subscriber external
> to the Sipwise CE system to authenticate to the Asterisk system internal
> to the Sipwise CE system.  
> 

What? An external subscriber to authenticate against the internal asterisk? The
asterisk system running in the spce is just a voicebox endpoint.

Could you please explain what's you scenario and what you are trying to
achieve? Sorry but I didn't understand.

>  
> 
> What am I missing?  I'm sure it's pretty simple but I can't see it.
> 

Please tell us in more detail what your scenario is and what you did.


cheers,

Jon

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130323/42f36788/attachment-0001.asc>

From am at sipalto.com  Sat Mar 23 13:32:11 2013
From: am at sipalto.com (Ahmed Murad)
Date: Sat, 23 Mar 2013 17:32:11 -0000
Subject: [Spce-user] routing calls between peers
Message-ID: <04db01ce27ec$5a3f5c30$0ebe1490$@sipalto.com>

Hello,

 

I am new to SIPWISE, do you have any guides on how to route calls from 1
peer to another peer via the SIPWISE box?

 

Any guidance would be greatly appreciated.

 

Best regards,

 

Ahmed

 

Ahmed Murad | SiPalto Ltd. 

Tel: 0207 148 7525  |  Fax: 0207 148  7526  |  Email:
<mailto:am at sipalto.com> am at sipalto.com
6-16 Arbutus St, London, E8 4DT | Website:  <http://www.sipalto.com/>
www.sipalto.com

 

Description: Description: Description: Description: Description:
Description: Description:
http://www.sipalto.com/images/large-logo.png

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130323/9132c9cd/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.png
Type: image/png
Size: 5743 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130323/9132c9cd/attachment-0001.png>

From am at sipalto.com  Sat Mar 23 13:36:19 2013
From: am at sipalto.com (Ahmed Murad)
Date: Sat, 23 Mar 2013 17:36:19 -0000
Subject: [Spce-user] Connect softphone to SIPWISE
Message-ID: <04e101ce27ec$ee7b0d90$cb7128b0$@sipalto.com>

Hello,

 

I have been following your guides and successfully created & connected to a
couple of external peers.

 

However I am having trouble connecting a softphone to our first subscriber.
I have created the domain, account and subscriber.

 

But I get a timeout 500 error when I try to connect the softphone to the
subscriber, I have ensured there are no firewall rules causing the timeout.

 

Please could you point me in the right direction in regards to
troubleshooting, is there anything I've missed?

 

Best regards,

 

Ahmed

 

Ahmed Murad | SiPalto Ltd. 

Tel: 0207 148 7525  |  Fax: 0207 148  7526  |  Email:
<mailto:am at sipalto.com> am at sipalto.com
6-16 Arbutus St, London, E8 4DT | Website:  <http://www.sipalto.com/>
www.sipalto.com

 

Description: Description: Description: Description: Description:
Description: Description:
http://www.sipalto.com/images/large-logo.png

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130323/06e1a274/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.png
Type: image/png
Size: 5743 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130323/06e1a274/attachment-0001.png>

From nh.developer at gmail.com  Sat Mar 23 15:03:01 2013
From: nh.developer at gmail.com (Developing Tomorrow)
Date: Sat, 23 Mar 2013 20:03:01 +0100
Subject: [Spce-user] Sipwise over openvpn
Message-ID: <00b701ce27f9$0b2de460$2189ad20$@gmail.com>

Hi.,

 

I have "successfully" installed sipwise and am able to register users and
make calls using any "normal" providers.

Now

In the same box, I have created a tunnel to another network that has a
gateway.

I am able to ping that gateway from the station that has sipwise also able
to send traffic to the gateway, but I will get no answer because of the ip
used.

Let me put down the layout:

All the following ips are fake but will clearly show the layout.

 

Sipwise external ip: 200.200.200.200

Sipwise internal ip: 10.0.0.1

 

Tunnel router: 10.0.0.10

Gateway across tunel: 192.168.10.10

 

If I use a standard asterisk box, give it ip 10.0.1 and send calls to
gateway, the voice will come back thru tunnel because the gateway is
replying to 10.0.0.1

 

In my sipwise case, it does not work as sipwise is sending the call with ip
200.200.200.200 so the gateway is answering thru the normal internet and not
the tunnel.

 

How can I configure sipwise to send info to that gateway using ip in the
form ttttttttt at 10.0.0.1 <mailto:ttttttttt at 10.0.0.1>  and not
ttttttttt at 200.200.200.200 <mailto:ttttttttt at 200.200.200.200> .

 

The media relay is supposed to do that function but I have no clue on how to
configure it.

I could only create a socket with ip 10.0.0.1 and that allows me to send the
traffic thru the vpn, but does not change the ip that the gateway receives.

 

I would like to receive a call like TTTTTTTTTTTTTTTTTT at yyy.yyy.yyy.yyy
<mailto:TTTTTTTTTTTTTTTTTT at yyy.yyy.yyy.yyy>  and send it to the gateway like
TTTTTTTTTTTTTT at 10.0.0.1 <mailto:TTTTTTTTTTTTTT at 10.0.0.1> ....

So the gateway would answer thru the vpn to 10.0.0.1 and sipwise would send
it back to yyy.yyy.yyy.yyy

 

Media proxy? Media relay?

 

How can I configure that.

Guidance would be mostly appreciated.

Thank you.

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130323/d89d1b7d/attachment-0001.html>

From oliver at oliverv.com  Sat Mar 23 15:12:39 2013
From: oliver at oliverv.com (Oliver Vermeulen)
Date: Sat, 23 Mar 2013 20:12:39 +0100
Subject: [Spce-user] Why does it take 35sec to register
Message-ID: <011701ce27fa$63ef56a0$2bce03e0$@oliverv.com>

Hello All,

 

I tested sipwise yesterday.. install was successful very easy.

 

Does anybody have problems with sip registration ?

Why does it take 35sec to register.

 

Also when it's registered and I dial takes long time to process call.

 

Cheers,

 

Olivier Vermeulen

Description: oliverv.jpg



msn:  <mailto:oliver at oliverv.com> oliver at oliverv.com

 <http://www.wvg-tele.com/> http://www.oliverv.com

****************************************************************************
************ 
This E-mail message, including any attachments, is intended only for the
person or entity to which it is addressed, 
and may contain confidential Information. If you are not the intended
recipient, any review, retransmission, disclosure, copying, 
modification or other use of this E-mail message or attachments is strictly
forbidden. If you have received this E-mail message in error,
please contact the author and delete the message and any attachments from
your computer.
P Think before you print.

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130323/66b7ba3e/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image002.png
Type: application/octet-stream
Size: 21421 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130323/66b7ba3e/attachment-0001.png>

From jbonilla at sipwise.com  Sat Mar 23 15:21:10 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Sat, 23 Mar 2013 20:21:10 +0100
Subject: [Spce-user] Why does it take 35sec to register
In-Reply-To: <011701ce27fa$63ef56a0$2bce03e0$@oliverv.com>
References: <011701ce27fa$63ef56a0$2bce03e0$@oliverv.com>
Message-ID: <20130323202110.6f71ae8d@quenya>

El Sat, 23 Mar 2013 20:12:39 +0100
"Oliver Vermeulen" <oliver at oliverv.com> escribi?:

> Hello All,
> 
>  
> 
> I tested sipwise yesterday.. install was successful very easy.
> 
>  
> 
> Does anybody have problems with sip registration ?
> 
> Why does it take 35sec to register.
> 
>  
> 
> Also when it's registered and I dial takes long time to process call.
> 
> 

Please make a ngrep-sip capture of a registration and another one of a call. We
will be able to see how long takes.


-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130323/afee13bc/attachment-0001.asc>

From jbonilla at sipwise.com  Sat Mar 23 15:24:46 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Sat, 23 Mar 2013 20:24:46 +0100
Subject: [Spce-user] Sipwise over openvpn
In-Reply-To: <00b701ce27f9$0b2de460$2189ad20$@gmail.com>
References: <00b701ce27f9$0b2de460$2189ad20$@gmail.com>
Message-ID: <20130323202446.1bc5eeed@quenya>

El Sat, 23 Mar 2013 20:03:01 +0100
"Developing Tomorrow" <nh.developer at gmail.com> escribi?:

> Hi.,
> 
>  
> 
> I have "successfully" installed sipwise and am able to register users and
> make calls using any "normal" providers.
> 
> Now
> 
> In the same box, I have created a tunnel to another network that has a
> gateway.
> 
> I am able to ping that gateway from the station that has sipwise also able
> to send traffic to the gateway, but I will get no answer because of the ip
> used.
> 
> Let me put down the layout:
> 
> All the following ips are fake but will clearly show the layout.
> 
>  
> 
> Sipwise external ip: 200.200.200.200
> 
> Sipwise internal ip: 10.0.0.1
> 
>  
> 
> Tunnel router: 10.0.0.10
> 
> Gateway across tunel: 192.168.10.10
> 
>  
> 
> If I use a standard asterisk box, give it ip 10.0.1 and send calls to
> gateway, the voice will come back thru tunnel because the gateway is
> replying to 10.0.0.1
> 
>  
> 
> In my sipwise case, it does not work as sipwise is sending the call with ip
> 200.200.200.200 so the gateway is answering thru the normal internet and not
> the tunnel.
> 
>  
> 
> How can I configure sipwise to send info to that gateway using ip in the
> form ttttttttt at 10.0.0.1 <mailto:ttttttttt at 10.0.0.1>  and not
> ttttttttt at 200.200.200.200 <mailto:ttttttttt at 200.200.200.200> .
> 
>  

Why don't you configure in the other side of the tunnel. "Send the traffic to
20.200.200.200 via 10.0.0.1" ?

You can use the "extra_socket" option in the spce and that will work for you at
sip signaling level but at rtp level it won't work so you'd get signaling via
the tunel and media over internet. If that's what you want it's fine. 

Anyways I'd continue using the external main address and try to fix you issue
at networking level, with routes.


> 
> The media relay is supposed to do that function but I have no clue on how to
> configure it.
> 
> I could only create a socket with ip 10.0.0.1 and that allows me to send the
> traffic thru the vpn, but does not change the ip that the gateway receives.
> 



cheers,

Jon

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130323/d6223bd4/attachment-0001.asc>

From jbonilla at sipwise.com  Sat Mar 23 15:26:05 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Sat, 23 Mar 2013 20:26:05 +0100
Subject: [Spce-user] Connect softphone to SIPWISE
In-Reply-To: <04e101ce27ec$ee7b0d90$cb7128b0$@sipalto.com>
References: <04e101ce27ec$ee7b0d90$cb7128b0$@sipalto.com>
Message-ID: <20130323202605.57ed0cdd@quenya>

El Sat, 23 Mar 2013 17:36:19 -0000
"Ahmed Murad" <am at sipalto.com> escribi?:

> Hello,
> 
>  
> 
> I have been following your guides and successfully created & connected to a
> couple of external peers.
> 
>  
> 
> However I am having trouble connecting a softphone to our first subscriber.
> I have created the domain, account and subscriber.
> 
>  
> 
> But I get a timeout 500 error when I try to connect the softphone to the
> subscriber, I have ensured there are no firewall rules causing the timeout.
> 
>  
> 
> Please could you point me in the right direction in regards to
> troubleshooting, is there anything I've missed?
> 
>  
> 

Please do a ngre-sip capture in the systeem and let us check the
incoming/outgoing traffic.

Also, you can check /var/log/ngcp/kamaiio-proxy.log to check if REGISTER
requests are arriving and what's going on with them.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130323/8d7f84f5/attachment-0001.asc>

From jbonilla at sipwise.com  Sat Mar 23 15:40:18 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Sat, 23 Mar 2013 20:40:18 +0100
Subject: [Spce-user] routing calls between peers
In-Reply-To: <04db01ce27ec$5a3f5c30$0ebe1490$@sipalto.com>
References: <04db01ce27ec$5a3f5c30$0ebe1490$@sipalto.com>
Message-ID: <20130323204018.49f4428a@quenya>

El Sat, 23 Mar 2013 17:32:11 -0000
"Ahmed Murad" <am at sipalto.com> escribi?:

> Hello,
> 
>  
> 
> I am new to SIPWISE, do you have any guides on how to route calls from 1
> peer to another peer via the SIPWISE box?
> 
>  
> 
> Any guidance would be greatly appreciated.
> 
>  
> 
> Best regards,
> 
>  
>

Hi Ahmed

By default, the spce does not allow peer-peer calls. It only allows
peer-subscriber, subscriber-subscriber and subscriber-peer.

But you can indeed send calls from one peer to another. There are two ways. Let
me explain them and their differences:


- You have the option "force_outbound_calls_to_peer" you can set in the proxy
  preferences in the admin interface. If this option is set, any incoming call
  coming from this peer will be routed to pstn (to another or the same peer)
  based on the peer rules. This will be done even if the callee is a local
  subscriber.


- There's another option in config.yml which is called "allow_peer_relay".
  This is global and will work like this: When a call comes from a peer it
  will terminate the call locally if the calee is local or will send the call
  to the pstn following the peer rules if not.


The first method has the advantage of being specific for a single peer and you
can enable it per peer. I use it for sbc scenarios or pure peer-peer scenarios,
where you don't have local subscriber or those are "sbc subscribers with upper
registration". 

The second one is global, it will be enabled for all you peers. The advantage
is that it won't send out you local calls. I use this for migrations, where you
still need to send the traffic of legacy class5 systems via the new Sipwise
Class5.


In case you want to sell minutes to other peers and change them for it, the
above options have a problem: The rating engine is not prepared atm to bill and
charge peering servers. You'll get the cdrs and you will need to use an
external billing system for the charges. If this is your case you still have a
third option:

- Create your pstn gateways as peers. But other providers, you clients, the
  ones you want to charge create them as *subscribers*. Creating them as
  subscribers will allow you to rate them, apply ncos, prepaid (in case of
  pro) and lots of features not present in peer options. You just need to
  create a permanent contact for them, set them as trusted form the ip
  addresses they use and allow all the clis (if you want).


Please let me know which scenario you're looking for and if any of these three
options match it.

cheers,

Jon

 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130323/2ea74561/attachment-0001.asc>

From jbonilla at sipwise.com  Sat Mar 23 15:44:02 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Sat, 23 Mar 2013 20:44:02 +0100
Subject: [Spce-user] Invoicing and Billing Peers/Trunks
In-Reply-To: <514C59A3.7090101@gmail.com>
References: <514C59A3.7090101@gmail.com>
Message-ID: <20130323204402.4eeaf025@quenya>

El Fri, 22 Mar 2013 15:16:19 +0200
Deon Vermeulen <vermeulen.deon at gmail.com> escribi?:

> Good Day
> 
> Is there any possibility that one could generate an Invoice to customers 
> with a Trunk to SPCE 2.7?


You can get the cdrs, but not an invoice as PDF is you mean that. Any customer,
has to be created as subscriber and it will be rated if you don't disable it.

> 
> Is there any possibility to Generate a Bill to a Peering Partner for 
> calls sent to SPCE 2.7. i.e Reverse billing?
> 

Not at the moment. Peers can't be charged in this version of the spce.


cheers,

Jon
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130323/d3643d13/attachment-0001.asc>

From m.rys at tri-lakes.net  Sat Mar 23 16:21:46 2013
From: m.rys at tri-lakes.net (Marc Rys)
Date: Sat, 23 Mar 2013 15:21:46 -0500 (CDT)
Subject: [Spce-user] 483 Too Many Hops
In-Reply-To: <20130323204402.4eeaf025@quenya>
Message-ID: <c6ac07e3-43b0-4127-8a7d-8f562e0eef4c@usbsnpzimbra01v.servers.rystec.com>

I've been evaluating SPCE over the last week.  I've already setup a couple test subscribers and setup a peer with a provider we work with for SIP LD Term.  All of those test have worked very successful.

We also have our own media gateway which is interconnected with the local PSTN via TDM trunk, but I send incoming calls from the PSTN through our MediaGateway to SPCE, SPCE is responding back with 483 "Too Many Hops".  Attached is the wireshark cap.

Any help is appreciated.

Marc Rys
http://www.tri-lakes.net
http://www.rystec.com
-------------- next part --------------
A non-text attachment was scrubbed...
Name: SPCE-INBOUND-Taqua-483TooManyHops.pcap
Type: application/octet-stream
Size: 2220 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130323/3e72b68f/attachment-0001.pcap>

From jbonilla at sipwise.com  Sat Mar 23 16:45:10 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Sat, 23 Mar 2013 21:45:10 +0100
Subject: [Spce-user] 483 Too Many Hops
In-Reply-To: <c6ac07e3-43b0-4127-8a7d-8f562e0eef4c@usbsnpzimbra01v.servers.rystec.com>
References: <20130323204402.4eeaf025@quenya>
	<c6ac07e3-43b0-4127-8a7d-8f562e0eef4c@usbsnpzimbra01v.servers.rystec.com>
Message-ID: <20130323214510.5498f5dc@quenya>

El Sat, 23 Mar 2013 15:21:46 -0500 (CDT)
Marc Rys <m.rys at tri-lakes.net> escribi?:

> En-Respuesta-A: <20130323204402.4eeaf025 at quenya>


Please don't answer a mail to create a new thread. Use the "new mail" button
instead. 


Cheers,

Jon

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130323/df575ddc/attachment-0001.asc>

From lorenzo.mangani at gmail.com  Sat Mar 23 16:55:40 2013
From: lorenzo.mangani at gmail.com (Lorenzo Mangani)
Date: Sat, 23 Mar 2013 21:55:40 +0100
Subject: [Spce-user] 483 Too Many Hops
In-Reply-To: <c6ac07e3-43b0-4127-8a7d-8f562e0eef4c@usbsnpzimbra01v.servers.rystec.com>
References: <20130323204402.4eeaf025@quenya>
	<c6ac07e3-43b0-4127-8a7d-8f562e0eef4c@usbsnpzimbra01v.servers.rystec.com>
Message-ID: <CAKN+pHYO2a=jA5rJ=y52ONYW75Duh97o5b-RN4qK+kbSQo9NuA@mail.gmail.com>

Marc,

Your Tarqua invite has Max-Forwards set to 1.
Try increasing the allowed hops and the call will terminate.


Lorenzo Mangani

HOMER DEV TEAM
QXIP - Network Engineering

On Sat, Mar 23, 2013 at 9:21 PM, Marc Rys <m.rys at tri-lakes.net> wrote:

> I've been evaluating SPCE over the last week.  I've already setup a couple
> test subscribers and setup a peer with a provider we work with for SIP LD
> Term.  All of those test have worked very successful.
>
> We also have our own media gateway which is interconnected with the local
> PSTN via TDM trunk, but I send incoming calls from the PSTN through our
> MediaGateway to SPCE, SPCE is responding back with 483 "Too Many Hops".
>  Attached is the wireshark cap.
>
> Any help is appreciated.
>
> Marc Rys
> http://www.tri-lakes.net
> http://www.rystec.com
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
>
>


--
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130323/c66d0c64/attachment-0001.html>

From m.rys at tri-lakes.net  Sat Mar 23 17:33:36 2013
From: m.rys at tri-lakes.net (Marc Rys)
Date: Sat, 23 Mar 2013 16:33:36 -0500 (CDT)
Subject: [Spce-user] 404 Not Found
In-Reply-To: <dd2ad9e2-70fc-4987-b767-fc7de6d5bb13@usbsnpzimbra01v.servers.rystec.com>
Message-ID: <396bde5e-b30d-42cb-ad5d-092d77b22c90@usbsnpzimbra01v.servers.rystec.com>

Ok, I adjusted my Taqua/Gateway to 4 hops and now I dont get the 483 anymore..  Now I get a 404 Not Found.

I'm so close..

pcap attached.

Marc Rys
http://www.tri-lakes.net
http://www.rystec.com
-------------- next part --------------
A non-text attachment was scrubbed...
Name: SPCE-INBOUND-Taqua-404NotFound.pcap
Type: application/octet-stream
Size: 6396 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130323/a7d2aebf/attachment-0001.pcap>

From lorenzo.mangani at gmail.com  Sat Mar 23 17:39:50 2013
From: lorenzo.mangani at gmail.com (Lorenzo Mangani)
Date: Sat, 23 Mar 2013 22:39:50 +0100
Subject: [Spce-user] 483 Too Many Hops
In-Reply-To: <CAKN+pHYO2a=jA5rJ=y52ONYW75Duh97o5b-RN4qK+kbSQo9NuA@mail.gmail.com>
References: <20130323204402.4eeaf025@quenya>
	<c6ac07e3-43b0-4127-8a7d-8f562e0eef4c@usbsnpzimbra01v.servers.rystec.com>
	<CAKN+pHYO2a=jA5rJ=y52ONYW75Duh97o5b-RN4qK+kbSQo9NuA@mail.gmail.com>
Message-ID: <CAKN+pHaUiEuDtj9k=yKUcqSqWbHKCC=iTSoN5ZAOL6d7wF9fgA@mail.gmail.com>

Marc,

Please don't fork the messages by creating a new thread for each step of
the discussion and consult the documentation.
You need an inbound rewrite rule applied to strip the + from the INVITE in
order to match the local user, as well described in the Handbook, actually
this is exactly the example shown there:
http://www.sipwise.com/doc/2.7/spce/ar01s06.html#dialplans

Lorenzo Mangani

HOMER DEV TEAM
QXIP - Network Engineering

On Sat, Mar 23, 2013 at 9:55 PM, Lorenzo Mangani
<lorenzo.mangani at gmail.com>wrote:

> Marc,
>
> Your Tarqua invite has Max-Forwards set to 1.
> Try increasing the allowed hops and the call will terminate.
>
>
> Lorenzo Mangani
>
> HOMER DEV TEAM
> QXIP - Network Engineering
>
> On Sat, Mar 23, 2013 at 9:21 PM, Marc Rys <m.rys at tri-lakes.net> wrote:
>
>> I've been evaluating SPCE over the last week.  I've already setup a
>> couple test subscribers and setup a peer with a provider we work with for
>> SIP LD Term.  All of those test have worked very successful.
>>
>> We also have our own media gateway which is interconnected with the local
>> PSTN via TDM trunk, but I send incoming calls from the PSTN through our
>> MediaGateway to SPCE, SPCE is responding back with 483 "Too Many Hops".
>>  Attached is the wireshark cap.
>>
>> Any help is appreciated.
>>
>> Marc Rys
>> http://www.tri-lakes.net
>> http://www.rystec.com
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/listinfo/spce-user
>>
>>
>
>
> --
>
>
>
>


--
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130323/8385ad4c/attachment-0001.html>

From jbonilla at sipwise.com  Sat Mar 23 17:40:18 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Sat, 23 Mar 2013 22:40:18 +0100
Subject: [Spce-user] 404 Not Found
In-Reply-To: <396bde5e-b30d-42cb-ad5d-092d77b22c90@usbsnpzimbra01v.servers.rystec.com>
References: <dd2ad9e2-70fc-4987-b767-fc7de6d5bb13@usbsnpzimbra01v.servers.rystec.com>
	<396bde5e-b30d-42cb-ad5d-092d77b22c90@usbsnpzimbra01v.servers.rystec.com>
Message-ID: <20130323224018.4cc3eab0@quenya>

El Sat, 23 Mar 2013 16:33:36 -0500 (CDT)
Marc Rys <m.rys at tri-lakes.net> escribi?:

> Ok, I adjusted my Taqua/Gateway to 4 hops and now I dont get the 483
> anymore..  Now I get a 404 Not Found.
> 


Please check the kamailio-proxy logs for that call. Most probably the requested
number was not local and the system returns a 404.

Are you using rewrite rules to normalize the call to E164 format?

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130323/b0dba2a9/attachment-0001.asc>

From wfulton at thirdhatch.com  Sat Mar 23 17:34:07 2013
From: wfulton at thirdhatch.com (William Fulton)
Date: Sat, 23 Mar 2013 14:34:07 -0700
Subject: [Spce-user] Sipwise CE 2.7 SBC config revisited
In-Reply-To: <20130323111316.773fab5b@quenya>
References: <0F449623D8AAB145A3278882C750BA5831C713@sea2k3ex06.thirdhatch.corp>
	<20130323111316.773fab5b@quenya>
Message-ID: <0F449623D8AAB145A3278882C750BA5831C719@sea2k3ex06.thirdhatch.corp>

Jon,

The SPCE system is multi-homed with eth0 running on a public routable ip address and eth1 running on a private class c network.  We have an asterisk server running in the class c network as well.  We are trying to use the SPCE system to protect the asterisk server while allowing access through the public internet to authenticated users.

So, let?s say SPCE looks like this:
Eth0: x.x.x.x
Eth1: y.y.y.y

The Asterisk Server:
y.y.y.x

The client:
x.x.x.y

I want the client to authenticate against the SPCE, then the SPCE authenticates the client to the Asterisk server.  I can get the client to authenticate against the SPCE but I cannot get the SPCE to authenticate the client against the Asterisk server.

I have set peer_auth_register flag and also provided the credentials to the SPCE server for the client's account on the asterisk server along with the realm address for the asterisk server.  The SPCE never even tries to authenticate to the asterisk server.

Thank you,
William

-----Original Message-----
From: spce-user-bounces at lists.sipwise.com [mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of Jon Bonilla (Manwe)
Sent: Saturday, March 23, 2013 3:13 AM
To: spce-user at lists.sipwise.com
Subject: Re: [Spce-user] Sipwise CE 2.7 SBC config revisited

El Fri, 22 Mar 2013 16:27:39 -0700
"William Fulton" <wfulton at thirdhatch.com> escribi?:

> Guys,
> 
>  
> 
> I have spent a couple of weeks trying to make this work with 
> absolutely no luck.  I have read the manual cover to cover and there 
> is clearly a section missing.  Section 6 references a section in 
> Section 7 about advanced network configuration that does not exist.
> 
>  
> 
> I have two interfaces:
> 
>  
> 
> eth0 and eth1
> 
>  
> 
> *         eth0 is on the external subnet and this configuration was easy
> as I simply assigned the sip_ext, web_ext, ssh_ext, etc to this adaptor.
> 
> *         eth1 is on the internal subnet and I have added the role of
> sip_int to this adaptor.  


why did you do this? Why not lo interface for this role?

> 
> *         I created a new subscriber and set the preferences to
> authenticate to the peer and set the peer username, password and realm

Did you also check the "peer_auth_register" option?


> 
> *         Despite my best efforts, I cannot get the subscriber external
> to the Sipwise CE system to authenticate to the Asterisk system 
> internal to the Sipwise CE system.
> 

What? An external subscriber to authenticate against the internal asterisk? The
asterisk system running in the spce is just a voicebox endpoint.

Could you please explain what's you scenario and what you are trying to
achieve? Sorry but I didn't understand.

>  
> 
> What am I missing?  I'm sure it's pretty simple but I can't see it.
> 

Please tell us in more detail what your scenario is and what you did.


cheers,

Jon


From jbonilla at sipwise.com  Sat Mar 23 18:57:10 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Sat, 23 Mar 2013 23:57:10 +0100
Subject: [Spce-user] Sipwise CE 2.7 SBC config revisited
In-Reply-To: <0F449623D8AAB145A3278882C750BA5831C719@sea2k3ex06.thirdhatch.corp>
References: <0F449623D8AAB145A3278882C750BA5831C713@sea2k3ex06.thirdhatch.corp>
	<20130323111316.773fab5b@quenya>
	<0F449623D8AAB145A3278882C750BA5831C719@sea2k3ex06.thirdhatch.corp>
Message-ID: <20130323235710.275098f4@quenya>

El Sat, 23 Mar 2013 14:34:07 -0700
"William Fulton" <wfulton at thirdhatch.com> escribi?:

> Jon,
> 
> The SPCE system is multi-homed with eth0 running on a public routable ip
> address and eth1 running on a private class c network.  We have an asterisk
> server running in the class c network as well.  We are trying to use the SPCE
> system to protect the asterisk server while allowing access through the
> public internet to authenticated users.
> 
> So, let?s say SPCE looks like this:
> Eth0: x.x.x.x
> Eth1: y.y.y.y
> 
> The Asterisk Server:
> y.y.y.x
> 
> The client:
> x.x.x.y
> 
> I want the client to authenticate against the SPCE, then the SPCE
> authenticates the client to the Asterisk server.  I can get the client to
> authenticate against the SPCE but I cannot get the SPCE to authenticate the
> client against the Asterisk server.
> 
> I have set peer_auth_register flag and also provided the credentials to the
> SPCE server for the client's account on the asterisk server along with the
> realm address for the asterisk server.  The SPCE never even tries to
> authenticate to the asterisk server.
> 
> 

Th spce will always use sip_ext as incoming and outgoing interface for the sip
traffic. sip_int is only for internal communication of the spce elements.

If you want the spce to communicate to the asterisk server and register there,
it will use x.x.x.x ip to go to y.y.y.x


Do you have the network and the routes prepared for that? You router should
have the routes for the spce to reach the asterisk server in the internal
network.

I think this is more a networking issue here.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130323/bcca0a19/attachment-0001.asc>

From nh.developer at gmail.com  Sat Mar 23 20:19:16 2013
From: nh.developer at gmail.com (Developing Tomorrow)
Date: Sun, 24 Mar 2013 01:19:16 +0100
Subject: [Spce-user] Spce-user Digest, Vol 28, Issue 52
In-Reply-To: <mailman.1220.1364071519.2547.spce-user@lists.sipwise.com>
References: <mailman.1220.1364071519.2547.spce-user@lists.sipwise.com>
Message-ID: <00d101ce2825$390cbc40$ab2634c0$@gmail.com>


Hi Jon 

Thanks

When I discovered the fact that the rtp was sent thru the normal internet,
that is what I thought
For the router to send the packets thru the vpn.
The problem that came is that, I did not know how to do that.

Technically I know the command to force a host thru the vpn, but I cannot
apply that because the vpn is created thru the internet
Now, if I tell the router to force the vpn host also thru the vpn, then it
cannot create itself and also go thru its own tunnel.

I than thought that, any udp packet sent from gateway should go thru the vpn
tunnel.
Anyone know how to translate that to iptables command?
Udp packets from 192.168.10.10 all thru route 10.0.0.2 (tunnel on router
side of gateway)

thx


_________________________________________________________________________
Please consider the environment before printing this email.
?
Notice: Please do not include customer identifying information (i.e.,
account number, social security, etc.) in email communications.
?
Privacy Statement: This message may contain confidential information that is
protected by law. If you received this message in error, please notify the
sender by fax or email and delete this message. If you properly received
this message, you may use its contents only in strictaccordance with our
instructions and privacy policy.
If any person makes a false or misleading representation to obtain customer
information, that person may have committed a federal crime, and we may
report that person to the proper authorities.

-----Original Message-----
From: spce-user-bounces at lists.sipwise.com
[mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of
spce-user-request at lists.sipwise.com
Sent: 23 March 2013 21:45
To: spce-user at lists.sipwise.com
Subject: Spce-user Digest, Vol 28, Issue 52

Send Spce-user mailing list submissions to
	spce-user at lists.sipwise.com

To subscribe or unsubscribe via the World Wide Web, visit
	http://lists.sipwise.com/listinfo/spce-user
or, via email, send a message with subject or body 'help' to
	spce-user-request at lists.sipwise.com

You can reach the person managing the list at
	spce-user-owner at lists.sipwise.com

When replying, please edit your Subject line so it is more specific than
"Re: Contents of Spce-user digest..."


Today's Topics:

   1. Re: Why does it take 35sec to register (Jon Bonilla (Manwe))
   2. Re: Sipwise over openvpn (Jon Bonilla (Manwe))
   3. Re: Connect softphone to SIPWISE (Jon Bonilla (Manwe))
   4. Re: routing calls between peers (Jon Bonilla (Manwe))
   5. Re: Invoicing and Billing Peers/Trunks (Jon Bonilla (Manwe))
   6. 483 Too Many Hops (Marc Rys)
   7. Re: 483 Too Many Hops (Jon Bonilla (Manwe))


----------------------------------------------------------------------

Message: 1
Date: Sat, 23 Mar 2013 20:21:10 +0100
From: Jon Bonilla (Manwe) <jbonilla at sipwise.com>
Subject: Re: [Spce-user] Why does it take 35sec to register
To: spce-user at lists.sipwise.com
Message-ID: <20130323202110.6f71ae8d at quenya>
Content-Type: text/plain; charset="utf-8"

El Sat, 23 Mar 2013 20:12:39 +0100
"Oliver Vermeulen" <oliver at oliverv.com> escribi?:

> Hello All,
> 
>  
> 
> I tested sipwise yesterday.. install was successful very easy.
> 
>  
> 
> Does anybody have problems with sip registration ?
> 
> Why does it take 35sec to register.
> 
>  
> 
> Also when it's registered and I dial takes long time to process call.
> 
> 

Please make a ngrep-sip capture of a registration and another one of a call.
We will be able to see how long takes.


-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130323/afee13bc/
attachment-0001.pgp>

------------------------------

Message: 2
Date: Sat, 23 Mar 2013 20:24:46 +0100
From: Jon Bonilla (Manwe) <jbonilla at sipwise.com>
Subject: Re: [Spce-user] Sipwise over openvpn
To: spce-user at lists.sipwise.com
Message-ID: <20130323202446.1bc5eeed at quenya>
Content-Type: text/plain; charset="utf-8"

El Sat, 23 Mar 2013 20:03:01 +0100
"Developing Tomorrow" <nh.developer at gmail.com> escribi?:

> Hi.,
> 
>  
> 
> I have "successfully" installed sipwise and am able to register users 
> and make calls using any "normal" providers.
> 
> Now
> 
> In the same box, I have created a tunnel to another network that has a 
> gateway.
> 
> I am able to ping that gateway from the station that has sipwise also 
> able to send traffic to the gateway, but I will get no answer because 
> of the ip used.
> 
> Let me put down the layout:
> 
> All the following ips are fake but will clearly show the layout.
> 
>  
> 
> Sipwise external ip: 200.200.200.200
> 
> Sipwise internal ip: 10.0.0.1
> 
>  
> 
> Tunnel router: 10.0.0.10
> 
> Gateway across tunel: 192.168.10.10
> 
>  
> 
> If I use a standard asterisk box, give it ip 10.0.1 and send calls to 
> gateway, the voice will come back thru tunnel because the gateway is 
> replying to 10.0.0.1
> 
>  
> 
> In my sipwise case, it does not work as sipwise is sending the call 
> with ip
> 200.200.200.200 so the gateway is answering thru the normal internet 
> and not the tunnel.
> 
>  
> 
> How can I configure sipwise to send info to that gateway using ip in 
> the form ttttttttt at 10.0.0.1 <mailto:ttttttttt at 10.0.0.1>  and not
> ttttttttt at 200.200.200.200 <mailto:ttttttttt at 200.200.200.200> .
> 
>  

Why don't you configure in the other side of the tunnel. "Send the traffic
to
20.200.200.200 via 10.0.0.1" ?

You can use the "extra_socket" option in the spce and that will work for you
at sip signaling level but at rtp level it won't work so you'd get signaling
via the tunel and media over internet. If that's what you want it's fine. 

Anyways I'd continue using the external main address and try to fix you
issue at networking level, with routes.


> 
> The media relay is supposed to do that function but I have no clue on 
> how to configure it.
> 
> I could only create a socket with ip 10.0.0.1 and that allows me to 
> send the traffic thru the vpn, but does not change the ip that the gateway
receives.
> 



cheers,

Jon

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130323/d6223bd4/
attachment-0001.pgp>

------------------------------

Message: 3
Date: Sat, 23 Mar 2013 20:26:05 +0100
From: Jon Bonilla (Manwe) <jbonilla at sipwise.com>
Subject: Re: [Spce-user] Connect softphone to SIPWISE
To: spce-user at lists.sipwise.com
Message-ID: <20130323202605.57ed0cdd at quenya>
Content-Type: text/plain; charset="utf-8"

El Sat, 23 Mar 2013 17:36:19 -0000
"Ahmed Murad" <am at sipalto.com> escribi?:

> Hello,
> 
>  
> 
> I have been following your guides and successfully created & connected 
> to a couple of external peers.
> 
>  
> 
> However I am having trouble connecting a softphone to our first
subscriber.
> I have created the domain, account and subscriber.
> 
>  
> 
> But I get a timeout 500 error when I try to connect the softphone to 
> the subscriber, I have ensured there are no firewall rules causing the
timeout.
> 
>  
> 
> Please could you point me in the right direction in regards to 
> troubleshooting, is there anything I've missed?
> 
>  
> 

Please do a ngre-sip capture in the systeem and let us check the
incoming/outgoing traffic.

Also, you can check /var/log/ngcp/kamaiio-proxy.log to check if REGISTER
requests are arriving and what's going on with them.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130323/8d7f84f5/
attachment-0001.pgp>

------------------------------

Message: 4
Date: Sat, 23 Mar 2013 20:40:18 +0100
From: Jon Bonilla (Manwe) <jbonilla at sipwise.com>
Subject: Re: [Spce-user] routing calls between peers
To: spce-user at lists.sipwise.com
Message-ID: <20130323204018.49f4428a at quenya>
Content-Type: text/plain; charset="utf-8"

El Sat, 23 Mar 2013 17:32:11 -0000
"Ahmed Murad" <am at sipalto.com> escribi?:

> Hello,
> 
>  
> 
> I am new to SIPWISE, do you have any guides on how to route calls from 
> 1 peer to another peer via the SIPWISE box?
> 
>  
> 
> Any guidance would be greatly appreciated.
> 
>  
> 
> Best regards,
> 
>  
>

Hi Ahmed

By default, the spce does not allow peer-peer calls. It only allows
peer-subscriber, subscriber-subscriber and subscriber-peer.

But you can indeed send calls from one peer to another. There are two ways.
Let me explain them and their differences:


- You have the option "force_outbound_calls_to_peer" you can set in the
proxy
  preferences in the admin interface. If this option is set, any incoming
call
  coming from this peer will be routed to pstn (to another or the same peer)
  based on the peer rules. This will be done even if the callee is a local
  subscriber.


- There's another option in config.yml which is called "allow_peer_relay".
  This is global and will work like this: When a call comes from a peer it
  will terminate the call locally if the calee is local or will send the
call
  to the pstn following the peer rules if not.


The first method has the advantage of being specific for a single peer and
you can enable it per peer. I use it for sbc scenarios or pure peer-peer
scenarios, where you don't have local subscriber or those are "sbc
subscribers with upper registration". 

The second one is global, it will be enabled for all you peers. The
advantage is that it won't send out you local calls. I use this for
migrations, where you still need to send the traffic of legacy class5
systems via the new Sipwise Class5.


In case you want to sell minutes to other peers and change them for it, the
above options have a problem: The rating engine is not prepared atm to bill
and charge peering servers. You'll get the cdrs and you will need to use an
external billing system for the charges. If this is your case you still have
a third option:

- Create your pstn gateways as peers. But other providers, you clients, the
  ones you want to charge create them as *subscribers*. Creating them as
  subscribers will allow you to rate them, apply ncos, prepaid (in case of
  pro) and lots of features not present in peer options. You just need to
  create a permanent contact for them, set them as trusted form the ip
  addresses they use and allow all the clis (if you want).


Please let me know which scenario you're looking for and if any of these
three options match it.

cheers,

Jon

 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130323/2ea74561/
attachment-0001.pgp>

------------------------------

Message: 5
Date: Sat, 23 Mar 2013 20:44:02 +0100
From: Jon Bonilla (Manwe) <jbonilla at sipwise.com>
Subject: Re: [Spce-user] Invoicing and Billing Peers/Trunks
To: spce-user at lists.sipwise.com
Message-ID: <20130323204402.4eeaf025 at quenya>
Content-Type: text/plain; charset="utf-8"

El Fri, 22 Mar 2013 15:16:19 +0200
Deon Vermeulen <vermeulen.deon at gmail.com> escribi?:

> Good Day
> 
> Is there any possibility that one could generate an Invoice to 
> customers with a Trunk to SPCE 2.7?


You can get the cdrs, but not an invoice as PDF is you mean that. Any
customer, has to be created as subscriber and it will be rated if you don't
disable it.

> 
> Is there any possibility to Generate a Bill to a Peering Partner for 
> calls sent to SPCE 2.7. i.e Reverse billing?
> 

Not at the moment. Peers can't be charged in this version of the spce.


cheers,

Jon
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130323/d3643d13/
attachment-0001.pgp>

------------------------------

Message: 6
Date: Sat, 23 Mar 2013 15:21:46 -0500 (CDT)
From: Marc Rys <m.rys at tri-lakes.net>
Subject: [Spce-user] 483 Too Many Hops
To: spce-user at lists.sipwise.com
Message-ID:
	
<c6ac07e3-43b0-4127-8a7d-8f562e0eef4c at usbsnpzimbra01v.servers.rystec.com>
	
Content-Type: text/plain; charset="utf-8"

I've been evaluating SPCE over the last week.  I've already setup a couple
test subscribers and setup a peer with a provider we work with for SIP LD
Term.  All of those test have worked very successful.

We also have our own media gateway which is interconnected with the local
PSTN via TDM trunk, but I send incoming calls from the PSTN through our
MediaGateway to SPCE, SPCE is responding back with 483 "Too Many Hops".
Attached is the wireshark cap.

Any help is appreciated.

Marc Rys
http://www.tri-lakes.net
http://www.rystec.com
-------------- next part --------------
A non-text attachment was scrubbed...
Name: SPCE-INBOUND-Taqua-483TooManyHops.pcap
Type: application/octet-stream
Size: 2220 bytes
Desc: not available
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130323/3e72b68f/
attachment-0001.obj>

------------------------------

Message: 7
Date: Sat, 23 Mar 2013 21:45:10 +0100
From: Jon Bonilla (Manwe) <jbonilla at sipwise.com>
Subject: Re: [Spce-user] 483 Too Many Hops
To: spce-user at lists.sipwise.com
Message-ID: <20130323214510.5498f5dc at quenya>
Content-Type: text/plain; charset="utf-8"

El Sat, 23 Mar 2013 15:21:46 -0500 (CDT) Marc Rys <m.rys at tri-lakes.net>
escribi?:

> En-Respuesta-A: <20130323204402.4eeaf025 at quenya>


Please don't answer a mail to create a new thread. Use the "new mail" button
instead. 


Cheers,

Jon

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130323/df575ddc/
attachment.pgp>

------------------------------

_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com
http://lists.sipwise.com/listinfo/spce-user


End of Spce-user Digest, Vol 28, Issue 52
*****************************************




From wfulton at thirdhatch.com  Sun Mar 24 01:21:55 2013
From: wfulton at thirdhatch.com (William Fulton)
Date: Sat, 23 Mar 2013 22:21:55 -0700
Subject: [Spce-user] Sipwise CE 2.7 SBC config revisited
In-Reply-To: <20130323235710.275098f4@quenya>
References: <0F449623D8AAB145A3278882C750BA5831C713@sea2k3ex06.thirdhatch.corp><20130323111316.773fab5b@quenya><0F449623D8AAB145A3278882C750BA5831C719@sea2k3ex06.thirdhatch.corp>
	<20130323235710.275098f4@quenya>
Message-ID: <0F449623D8AAB145A3278882C750BA5831C71B@sea2k3ex06.thirdhatch.corp>


>Th spce will always use sip_ext as incoming and outgoing interface for the sip traffic. sip_int is only for internal communication of the spce elements.

>If you want the spce to communicate to the asterisk server and register there, it will use x.x.x.x ip to go to y.y.y.x


>Do you have the network and the routes prepared for that? You router should have the routes for the spce to reach the asterisk server in the internal >network.

>I think this is more a networking issue here.
So the SPCE does not know to use eth1 to access servers on that subnet?  Is there a way to designate both eth0 and eth1 for sip_ext?

Thank you,
William

From jbonilla at sipwise.com  Sun Mar 24 05:50:35 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Sun, 24 Mar 2013 10:50:35 +0100
Subject: [Spce-user] Sipwise CE 2.7 SBC config revisited
In-Reply-To: <0F449623D8AAB145A3278882C750BA5831C71B@sea2k3ex06.thirdhatch.corp>
References: <0F449623D8AAB145A3278882C750BA5831C713@sea2k3ex06.thirdhatch.corp>
	<20130323111316.773fab5b@quenya>
	<0F449623D8AAB145A3278882C750BA5831C719@sea2k3ex06.thirdhatch.corp>
	<20130323235710.275098f4@quenya>
	<0F449623D8AAB145A3278882C750BA5831C71B@sea2k3ex06.thirdhatch.corp>
Message-ID: <20130324105035.3ca73296@quenya>

El Sat, 23 Mar 2013 22:21:55 -0700
"William Fulton" <wfulton at thirdhatch.com> escribi?:

> 
> >Th spce will always use sip_ext as incoming and outgoing interface for the
> >sip traffic. sip_int is only for internal communication of the spce elements.
> 
> >If you want the spce to communicate to the asterisk server and register
> >there, it will use x.x.x.x ip to go to y.y.y.x
> 
> 
> >Do you have the network and the routes prepared for that? You router should
> >have the routes for the spce to reach the asterisk server in the internal
> >>network.
> 
> >I think this is more a networking issue here.
> So the SPCE does not know to use eth1 to access servers on that subnet?  Is
> there a way to designate both eth0 and eth1 for sip_ext?


Hi William.

No, the spce will listen only a single ip address. From that address, it's up
to the network to provide routes to the internet and to any internal network
you want to reach.

Can't your router provide that? Can't your internal elements reach the spce's
public ip address?

You can always do it via "route commands" and "ip_forward" enabled on the
system. But the "normal" way of doing this is network routing in a router.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130324/12266e3e/attachment-0001.asc>

From m.rys at tri-lakes.net  Sun Mar 24 11:54:57 2013
From: m.rys at tri-lakes.net (Marc Rys)
Date: Sun, 24 Mar 2013 10:54:57 -0500 (CDT)
Subject: [Spce-user] 483 Too Many Hops
In-Reply-To: <CAKN+pHaUiEuDtj9k=yKUcqSqWbHKCC=iTSoN5ZAOL6d7wF9fgA@mail.gmail.com>
Message-ID: <9dbde9c8-a122-472d-841f-f7b03069dd21@usbsnpzimbra01v.servers.rystec.com>

ok I'll stop changing the subject, even though I regret not naming it SPCE Taqua Integration.. 

Anyways, I've heeded your advice and I normalized the patterns and I'm now routing the calls from my Taqua PSTN gateway to my IP phones.  Now the newest problem to reveal itself is the incoming calls to my Phones fail to setup completely.  It appears all the call routing is working properly now,  I can call my DID from the PSTN, and my IP phone rings, but when I pickup the call, I hear no audio, and it appears the phone never completely sets up the call.  The screen still shows the phone as ringing.

The wireshark cap shows RTP moving between my gateway & SPCE, and RTP moving between SPCE and my IP phone, but the call never appears to setup completely on the IP phone.

Any thoughts?

Marc Rys 
http://www.tri-lakes.net 
http://www.rystec.com 


----- Original Message -----
From: "Lorenzo Mangani" <lorenzo.mangani at gmail.com>
To: "Marc Rys" <m.rys at tri-lakes.net>
Cc: spce-user at lists.sipwise.com
Sent: Saturday, March 23, 2013 4:39:50 PM
Subject: Re: [Spce-user] 483 Too Many Hops

Marc, 


Please don't fork the messages by creating a new thread for each step of the discussion and consult the documentation. 
You need an inbound rewrite rule applied to strip the + from the INVITE in order to match the local user, as well described in the Handbook, actually this is exactly the example shown there: http://www.sipwise.com/doc/2.7/spce/ar01s06.html#dialplans 



Lorenzo Mangani 



HOMER DEV TEAM 
QXIP - Network Engineering 


On Sat, Mar 23, 2013 at 9:55 PM, Lorenzo Mangani < lorenzo.mangani at gmail.com > wrote: 


Marc, 


Your Tarqua invite has Max-Forwards set to 1. 
Try increasing the allowed hops and the call will terminate. 





Lorenzo Mangani 



HOMER DEV TEAM 
QXIP - Network Engineering 



On Sat, Mar 23, 2013 at 9:21 PM, Marc Rys < m.rys at tri-lakes.net > wrote: 




I've been evaluating SPCE over the last week. I've already setup a couple test subscribers and setup a peer with a provider we work with for SIP LD Term. All of those test have worked very successful. 

We also have our own media gateway which is interconnected with the local PSTN via TDM trunk, but I send incoming calls from the PSTN through our MediaGateway to SPCE, SPCE is responding back with 483 "Too Many Hops". Attached is the wireshark cap. 

Any help is appreciated. 

Marc Rys 
http://www.tri-lakes.net 
http://www.rystec.com 
_______________________________________________ 
Spce-user mailing list 
Spce-user at lists.sipwise.com 
http://lists.sipwise.com/listinfo/spce-user 





-- 










-- 








-------------- next part --------------
A non-text attachment was scrubbed...
Name: SPCE-InboundFromTaqua-NoAudio1.pcap
Type: application/octet-stream
Size: 452641 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130324/c866f247/attachment-0001.pcap>

From m.rys at tri-lakes.net  Sun Mar 24 12:19:02 2013
From: m.rys at tri-lakes.net (Marc Rys)
Date: Sun, 24 Mar 2013 11:19:02 -0500 (CDT)
Subject: [Spce-user] 483 Too Many Hops
In-Reply-To: <9dbde9c8-a122-472d-841f-f7b03069dd21@usbsnpzimbra01v.servers.rystec.com>
Message-ID: <1c5d4ad6-1793-41ee-bc29-e46c581f0c55@usbsnpzimbra01v.servers.rystec.com>

Also, just as an FYI,  Both of my test IP Phone subscribers can all each other flawlessly.  The calls between them setup just as expected.  (video worked too!)

Marc Rys 
http://www.tri-lakes.net 
http://www.rystec.com 

----- Original Message -----
From: "Marc Rys" <m.rys at tri-lakes.net>
To: "Marc Rys" <m.rys at zimbra.rystec.com>
Cc: spce-user at lists.sipwise.com
Sent: Sunday, March 24, 2013 10:54:57 AM
Subject: Re: [Spce-user] 483 Too Many Hops

ok I'll stop changing the subject, even though I regret not naming it SPCE Taqua Integration.. 

Anyways, I've heeded your advice and I normalized the patterns and I'm now routing the calls from my Taqua PSTN gateway to my IP phones.  Now the newest problem to reveal itself is the incoming calls to my Phones fail to setup completely.  It appears all the call routing is working properly now,  I can call my DID from the PSTN, and my IP phone rings, but when I pickup the call, I hear no audio, and it appears the phone never completely sets up the call.  The screen still shows the phone as ringing.

The wireshark cap shows RTP moving between my gateway & SPCE, and RTP moving between SPCE and my IP phone, but the call never appears to setup completely on the IP phone.

Any thoughts?

Marc Rys 
http://www.tri-lakes.net 
http://www.rystec.com 


----- Original Message -----
From: "Lorenzo Mangani" <lorenzo.mangani at gmail.com>
To: "Marc Rys" <m.rys at tri-lakes.net>
Cc: spce-user at lists.sipwise.com
Sent: Saturday, March 23, 2013 4:39:50 PM
Subject: Re: [Spce-user] 483 Too Many Hops

Marc, 


Please don't fork the messages by creating a new thread for each step of the discussion and consult the documentation. 
You need an inbound rewrite rule applied to strip the + from the INVITE in order to match the local user, as well described in the Handbook, actually this is exactly the example shown there: http://www.sipwise.com/doc/2.7/spce/ar01s06.html#dialplans 



Lorenzo Mangani 



HOMER DEV TEAM 
QXIP - Network Engineering 


On Sat, Mar 23, 2013 at 9:55 PM, Lorenzo Mangani < lorenzo.mangani at gmail.com > wrote: 


Marc, 


Your Tarqua invite has Max-Forwards set to 1. 
Try increasing the allowed hops and the call will terminate. 





Lorenzo Mangani 



HOMER DEV TEAM 
QXIP - Network Engineering 



On Sat, Mar 23, 2013 at 9:21 PM, Marc Rys < m.rys at tri-lakes.net > wrote: 




I've been evaluating SPCE over the last week. I've already setup a couple test subscribers and setup a peer with a provider we work with for SIP LD Term. All of those test have worked very successful. 

We also have our own media gateway which is interconnected with the local PSTN via TDM trunk, but I send incoming calls from the PSTN through our MediaGateway to SPCE, SPCE is responding back with 483 "Too Many Hops". Attached is the wireshark cap. 

Any help is appreciated. 

Marc Rys 
http://www.tri-lakes.net 
http://www.rystec.com 
_______________________________________________ 
Spce-user mailing list 
Spce-user at lists.sipwise.com 
http://lists.sipwise.com/listinfo/spce-user 





-- 










-- 









_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com
http://lists.sipwise.com/listinfo/spce-user



From lorenzo.mangani at gmail.com  Sun Mar 24 12:25:03 2013
From: lorenzo.mangani at gmail.com (Lorenzo Mangani)
Date: Sun, 24 Mar 2013 17:25:03 +0100
Subject: [Spce-user] 483 Too Many Hops
In-Reply-To: <9dbde9c8-a122-472d-841f-f7b03069dd21@usbsnpzimbra01v.servers.rystec.com>
References: <CAKN+pHaUiEuDtj9k=yKUcqSqWbHKCC=iTSoN5ZAOL6d7wF9fgA@mail.gmail.com>
	<9dbde9c8-a122-472d-841f-f7b03069dd21@usbsnpzimbra01v.servers.rystec.com>
Message-ID: <CAKN+pHYD6tE55RxhbMoTx92cuOoVcn-GDRPpNv=GeUoFifK+dQ@mail.gmail.com>

Hi Marc,

As you can see from your trace, the call is never completely established,
as the 200 OK's ACK never makes it back to the user-agent.
After a brief look this seems to be caused by the gateway always using the
contact header to respond instead of the from header, as for the BYE
message also being rejected for the same reason - just my 2 cents, might be
wrong - maybe someone else on the list will have different input,

Best,

Lorenzo Mangani

HOMER DEV TEAM
QXIP - Network Engineering



On Sun, Mar 24, 2013 at 4:54 PM, Marc Rys <m.rys at tri-lakes.net> wrote:

> ok I'll stop changing the subject, even though I regret not naming it SPCE
> Taqua Integration..
>
> Anyways, I've heeded your advice and I normalized the patterns and I'm now
> routing the calls from my Taqua PSTN gateway to my IP phones.  Now the
> newest problem to reveal itself is the incoming calls to my Phones fail to
> setup completely.  It appears all the call routing is working properly now,
>  I can call my DID from the PSTN, and my IP phone rings, but when I pickup
> the call, I hear no audio, and it appears the phone never completely sets
> up the call.  The screen still shows the phone as ringing.
>
> The wireshark cap shows RTP moving between my gateway & SPCE, and RTP
> moving between SPCE and my IP phone, but the call never appears to setup
> completely on the IP phone.
>
> Any thoughts?
>
> Marc Rys
> http://www.tri-lakes.net
> http://www.rystec.com
>
>
> ----- Original Message -----
> From: "Lorenzo Mangani" <lorenzo.mangani at gmail.com>
> To: "Marc Rys" <m.rys at tri-lakes.net>
> Cc: spce-user at lists.sipwise.com
> Sent: Saturday, March 23, 2013 4:39:50 PM
> Subject: Re: [Spce-user] 483 Too Many Hops
>
> Marc,
>
>
> Please don't fork the messages by creating a new thread for each step of
> the discussion and consult the documentation.
> You need an inbound rewrite rule applied to strip the + from the INVITE in
> order to match the local user, as well described in the Handbook, actually
> this is exactly the example shown there:
> http://www.sipwise.com/doc/2.7/spce/ar01s06.html#dialplans
>
>
>
> Lorenzo Mangani
>
>
>
> HOMER DEV TEAM
> QXIP - Network Engineering
>
>
> On Sat, Mar 23, 2013 at 9:55 PM, Lorenzo Mangani <
> lorenzo.mangani at gmail.com > wrote:
>
>
> Marc,
>
>
> Your Tarqua invite has Max-Forwards set to 1.
> Try increasing the allowed hops and the call will terminate.
>
>
>
>
>
> Lorenzo Mangani
>
>
>
> HOMER DEV TEAM
> QXIP - Network Engineering
>
>
>
> On Sat, Mar 23, 2013 at 9:21 PM, Marc Rys < m.rys at tri-lakes.net > wrote:
>
>
>
>
> I've been evaluating SPCE over the last week. I've already setup a couple
> test subscribers and setup a peer with a provider we work with for SIP LD
> Term. All of those test have worked very successful.
>
> We also have our own media gateway which is interconnected with the local
> PSTN via TDM trunk, but I send incoming calls from the PSTN through our
> MediaGateway to SPCE, SPCE is responding back with 483 "Too Many Hops".
> Attached is the wireshark cap.
>
> Any help is appreciated.
>
> Marc Rys
> http://www.tri-lakes.net
> http://www.rystec.com
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
>
>
>
>
>
> --
>
>
>
>
>
>
>
>
>
>
> --
>
>
>
>
>
>
>
>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
>
>


--
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130324/20afc139/attachment-0001.html>

From agranig at sipwise.com  Sun Mar 24 12:43:18 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Sun, 24 Mar 2013 17:43:18 +0100
Subject: [Spce-user] 483 Too Many Hops
In-Reply-To: <CAKN+pHYD6tE55RxhbMoTx92cuOoVcn-GDRPpNv=GeUoFifK+dQ@mail.gmail.com>
References: <CAKN+pHaUiEuDtj9k=yKUcqSqWbHKCC=iTSoN5ZAOL6d7wF9fgA@mail.gmail.com>
	<9dbde9c8-a122-472d-841f-f7b03069dd21@usbsnpzimbra01v.servers.rystec.com>
	<CAKN+pHYD6tE55RxhbMoTx92cuOoVcn-GDRPpNv=GeUoFifK+dQ@mail.gmail.com>
Message-ID: <514F2D26.3030103@sipwise.com>

Hi,

On 03/24/2013 05:25 PM, Lorenzo Mangani wrote:
> As you can see from your trace, the call is never completely
> established, as the 200 OK's ACK never makes it back to the user-agent.
> After a brief look this seems to be caused by the gateway always using
> the contact header to respond instead of the from header, as for the BYE
> message also being rejected for the same reason - just my 2 cents, might
> be wrong - maybe someone else on the list will have different input,
>

The Contact handling is as expected, but the 208.73.198.12 doesn't 
convert the Record-Route headers added by the SPCE in the 200-OK (packet 
#14) into Route headers of the ACK (packet #15), so the proxy has no 
information where to relay the ACK to (note that ACKs to 200 are relayed 
end-to-end, so all Route information must be preserved).

The sender of the ACK *must* take the Record-Route headers of the 
previous 200-OK, convert them to Routes and add them to the message in 
reverse order, and send the request to the URI found in the top-most Route.

Andreas



From oliver at oliverv.com  Sun Mar 24 18:57:05 2013
From: oliver at oliverv.com (Oliver Vermeulen)
Date: Sun, 24 Mar 2013 23:57:05 +0100
Subject: [Spce-user] Question of features
Message-ID: <02c601ce28e2$e86a8ce0$b93fa6a0$@oliverv.com>

Hello ,

 

I'm testing sipwise , had some questions if anybody can answer me.

 

User CE version

 

1)      Does is support simple sip (messaging)

2)      Xcap 

3)      Presence

4)      Push notifications 

5)      Is possible to use external billing system via Radius ?

 

What clients are supported ? For all features ?

 

Best Regards,

 

Olivier Vermeulen

Description: oliverv.jpg


USA:  +1 (305) 722-1457
BELGIUM: + (32) 9-3955620
SPAIN: + (34) 931-845025
RO: + (40) 21-569-4703
msn:  <mailto:oliver at oliverv.com> oliver at oliverv.com

 <http://www.wvg-tele.com/> http://www.oliverv.com

****************************************************************************
************ 
This E-mail message, including any attachments, is intended only for the
person or entity to which it is addressed, 
and may contain confidential Information. If you are not the intended
recipient, any review, retransmission, disclosure, copying, 
modification or other use of this E-mail message or attachments is strictly
forbidden. If you have received this E-mail message in error,
please contact the author and delete the message and any attachments from
your computer.
P Think before you print.

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130324/78edfec5/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.png
Type: application/octet-stream
Size: 21421 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130324/78edfec5/attachment-0001.png>

From skchopperguy at gmail.com  Mon Mar 25 04:35:36 2013
From: skchopperguy at gmail.com (Skyler)
Date: Mon, 25 Mar 2013 01:35:36 -0700
Subject: [Spce-user] Question of features
In-Reply-To: <02c601ce28e2$e86a8ce0$b93fa6a0$@oliverv.com>
References: <02c601ce28e2$e86a8ce0$b93fa6a0$@oliverv.com>
Message-ID: <51500C58.6080902@gmail.com>

Hi,

  1. yes
  2. yes
  3. yes
  4. yes with work
  5. yes
  6. All that send REGISTER
  7. yes

  --Skyler

On 3/24/2013 3:57 PM, Oliver Vermeulen wrote:
> Hello ,
>
> I?m testing sipwise , had some questions if anybody can answer me.
>
> User CE version
>
> 1)Does is support simple sip (messaging)
>
> 2)Xcap
>
> 3)Presence
>
> 4)Push notifications
>
> 5)Is possible to use external billing system via Radius ?
>
> What clients are supported ? For all features ?
>
> *Best Regards,*
>
> *Olivier Vermeulen**
>
> *Description: oliverv.jpg
>
> *
> USA:*  +1 (305) 722-1457
> *BELGIUM: *+ (32) 9-3955620
> *SPAIN: *+ (34) 931-845025
> *RO:* + (40) 21-569-4703*
> msn:*oliver at oliverv.com <mailto:oliver at oliverv.com>
>
> http://www.oliverv.com <http://www.wvg-tele.com/>
>
> ****************************************************************************************
>
> This E-mail message, including any attachments, is intended only for the
> person or entity to which it is addressed,
> and may contain confidential Information. If you are not the intended
> recipient, any review, retransmission, disclosure, copying,
> modification or other use of this E-mail message or attachments is
> strictly forbidden. If you have received this E-mail message in error,
> please contact the author and delete the message and any attachments
> from your computer.
> P***Think before you print.*
>
>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
>



From mario.contreras at innovasur.es  Mon Mar 25 06:21:19 2013
From: mario.contreras at innovasur.es (Mario Contreras)
Date: Mon, 25 Mar 2013 11:21:19 +0100
Subject: [Spce-user] Outgoing calls in subscriber's cdr (admin panel)
Message-ID: <5150251F.10504@innovasur.es>

Hi all,

I've realized that since I upgraded to 2.7, from certain subscribers(I 
think all subscribers are configured in asterisk), outgoing calls are 
not showing in web admin panel, and that's not happening in csc.

I've seen that the file with this output is 
/usr/share/ngcp-www-admin/lib/admin/root/tt/subscriber_call_data.tt, 
right? But I don't know exactly how provisioning works, so I can't solve 
it. It's not a big deal, really...

Anyway, with the new details link, when the subscriber have many calls, 
it takes very long to load. Can it be done using AJAX? It's only an idea.

Thanks for your great job!



From am at sipalto.com  Mon Mar 25 07:19:20 2013
From: am at sipalto.com (Ahmed Murad)
Date: Mon, 25 Mar 2013 11:19:20 -0000
Subject: [Spce-user] Connect softphone to SIPWISE (Jon Bonilla (Manwe))
Message-ID: <058901ce294a$9919d9d0$cb4d8d70$@sipalto.com>

Hi,

Thanks for the info. I received help on connecting a UA to SIPWISE from
another user who advised to run the below commands, after running them UA's
connected without any issues, is it worth adding this to your documentation?

ngcp-network --set-interface=eth0 --ip=auto --netmask=auto
ngcp-network --move-from=lo --move-to=eth0 --type=web_ext --type=sip_ext
--type=rtp_ext --  -type=ssh_ext
Ngcpfg apply

Many thanks for your quick response anyway.

Best regards,

Ahmed

Ahmed Murad | SiPalto Ltd. 
Tel: 0207 148 7525? |? Fax: 0207 148? 7526? |? Email: am at sipalto.com
6-16 Arbutus St, London, E8 4DT | Website: www.sipalto.com
?



-----Original Message-----
From: spce-user-bounces at lists.sipwise.com
[mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of
spce-user-request at lists.sipwise.com
Sent: 23 March 2013 20:45
To: spce-user at lists.sipwise.com
Subject: Spce-user Digest, Vol 28, Issue 52

Send Spce-user mailing list submissions to
	spce-user at lists.sipwise.com

To subscribe or unsubscribe via the World Wide Web, visit
	http://lists.sipwise.com/listinfo/spce-user
or, via email, send a message with subject or body 'help' to
	spce-user-request at lists.sipwise.com

You can reach the person managing the list at
	spce-user-owner at lists.sipwise.com

When replying, please edit your Subject line so it is more specific than
"Re: Contents of Spce-user digest..."


Today's Topics:

   1. Re: Why does it take 35sec to register (Jon Bonilla (Manwe))
   2. Re: Sipwise over openvpn (Jon Bonilla (Manwe))
   3. Re: Connect softphone to SIPWISE (Jon Bonilla (Manwe))
   4. Re: routing calls between peers (Jon Bonilla (Manwe))
   5. Re: Invoicing and Billing Peers/Trunks (Jon Bonilla (Manwe))
   6. 483 Too Many Hops (Marc Rys)
   7. Re: 483 Too Many Hops (Jon Bonilla (Manwe))


----------------------------------------------------------------------

Message: 1
Date: Sat, 23 Mar 2013 20:21:10 +0100
From: Jon Bonilla (Manwe) <jbonilla at sipwise.com>
Subject: Re: [Spce-user] Why does it take 35sec to register
To: spce-user at lists.sipwise.com
Message-ID: <20130323202110.6f71ae8d at quenya>
Content-Type: text/plain; charset="utf-8"

El Sat, 23 Mar 2013 20:12:39 +0100
"Oliver Vermeulen" <oliver at oliverv.com> escribi?:

> Hello All,
> 
>  
> 
> I tested sipwise yesterday.. install was successful very easy.
> 
>  
> 
> Does anybody have problems with sip registration ?
> 
> Why does it take 35sec to register.
> 
>  
> 
> Also when it's registered and I dial takes long time to process call.
> 
> 

Please make a ngrep-sip capture of a registration and another one of a call.
We will be able to see how long takes.


-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130323/afee13bc/
attachment-0001.pgp>

------------------------------

Message: 2
Date: Sat, 23 Mar 2013 20:24:46 +0100
From: Jon Bonilla (Manwe) <jbonilla at sipwise.com>
Subject: Re: [Spce-user] Sipwise over openvpn
To: spce-user at lists.sipwise.com
Message-ID: <20130323202446.1bc5eeed at quenya>
Content-Type: text/plain; charset="utf-8"

El Sat, 23 Mar 2013 20:03:01 +0100
"Developing Tomorrow" <nh.developer at gmail.com> escribi?:

> Hi.,
> 
>  
> 
> I have "successfully" installed sipwise and am able to register users 
> and make calls using any "normal" providers.
> 
> Now
> 
> In the same box, I have created a tunnel to another network that has a 
> gateway.
> 
> I am able to ping that gateway from the station that has sipwise also 
> able to send traffic to the gateway, but I will get no answer because 
> of the ip used.
> 
> Let me put down the layout:
> 
> All the following ips are fake but will clearly show the layout.
> 
>  
> 
> Sipwise external ip: 200.200.200.200
> 
> Sipwise internal ip: 10.0.0.1
> 
>  
> 
> Tunnel router: 10.0.0.10
> 
> Gateway across tunel: 192.168.10.10
> 
>  
> 
> If I use a standard asterisk box, give it ip 10.0.1 and send calls to 
> gateway, the voice will come back thru tunnel because the gateway is 
> replying to 10.0.0.1
> 
>  
> 
> In my sipwise case, it does not work as sipwise is sending the call 
> with ip
> 200.200.200.200 so the gateway is answering thru the normal internet 
> and not the tunnel.
> 
>  
> 
> How can I configure sipwise to send info to that gateway using ip in 
> the form ttttttttt at 10.0.0.1 <mailto:ttttttttt at 10.0.0.1>  and not
> ttttttttt at 200.200.200.200 <mailto:ttttttttt at 200.200.200.200> .
> 
>  

Why don't you configure in the other side of the tunnel. "Send the traffic
to
20.200.200.200 via 10.0.0.1" ?

You can use the "extra_socket" option in the spce and that will work for you
at sip signaling level but at rtp level it won't work so you'd get signaling
via the tunel and media over internet. If that's what you want it's fine. 

Anyways I'd continue using the external main address and try to fix you
issue at networking level, with routes.


> 
> The media relay is supposed to do that function but I have no clue on 
> how to configure it.
> 
> I could only create a socket with ip 10.0.0.1 and that allows me to 
> send the traffic thru the vpn, but does not change the ip that the gateway
receives.
> 



cheers,

Jon

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130323/d6223bd4/
attachment-0001.pgp>

------------------------------

Message: 3
Date: Sat, 23 Mar 2013 20:26:05 +0100
From: Jon Bonilla (Manwe) <jbonilla at sipwise.com>
Subject: Re: [Spce-user] Connect softphone to SIPWISE
To: spce-user at lists.sipwise.com
Message-ID: <20130323202605.57ed0cdd at quenya>
Content-Type: text/plain; charset="utf-8"

El Sat, 23 Mar 2013 17:36:19 -0000
"Ahmed Murad" <am at sipalto.com> escribi?:

> Hello,
> 
>  
> 
> I have been following your guides and successfully created & connected 
> to a couple of external peers.
> 
>  
> 
> However I am having trouble connecting a softphone to our first
subscriber.
> I have created the domain, account and subscriber.
> 
>  
> 
> But I get a timeout 500 error when I try to connect the softphone to 
> the subscriber, I have ensured there are no firewall rules causing the
timeout.
> 
>  
> 
> Please could you point me in the right direction in regards to 
> troubleshooting, is there anything I've missed?
> 
>  
> 

Please do a ngre-sip capture in the systeem and let us check the
incoming/outgoing traffic.

Also, you can check /var/log/ngcp/kamaiio-proxy.log to check if REGISTER
requests are arriving and what's going on with them.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130323/8d7f84f5/
attachment-0001.pgp>

------------------------------

Message: 4
Date: Sat, 23 Mar 2013 20:40:18 +0100
From: Jon Bonilla (Manwe) <jbonilla at sipwise.com>
Subject: Re: [Spce-user] routing calls between peers
To: spce-user at lists.sipwise.com
Message-ID: <20130323204018.49f4428a at quenya>
Content-Type: text/plain; charset="utf-8"

El Sat, 23 Mar 2013 17:32:11 -0000
"Ahmed Murad" <am at sipalto.com> escribi?:

> Hello,
> 
>  
> 
> I am new to SIPWISE, do you have any guides on how to route calls from 
> 1 peer to another peer via the SIPWISE box?
> 
>  
> 
> Any guidance would be greatly appreciated.
> 
>  
> 
> Best regards,
> 
>  
>

Hi Ahmed

By default, the spce does not allow peer-peer calls. It only allows
peer-subscriber, subscriber-subscriber and subscriber-peer.

But you can indeed send calls from one peer to another. There are two ways.
Let me explain them and their differences:


- You have the option "force_outbound_calls_to_peer" you can set in the
proxy
  preferences in the admin interface. If this option is set, any incoming
call
  coming from this peer will be routed to pstn (to another or the same peer)
  based on the peer rules. This will be done even if the callee is a local
  subscriber.


- There's another option in config.yml which is called "allow_peer_relay".
  This is global and will work like this: When a call comes from a peer it
  will terminate the call locally if the calee is local or will send the
call
  to the pstn following the peer rules if not.


The first method has the advantage of being specific for a single peer and
you can enable it per peer. I use it for sbc scenarios or pure peer-peer
scenarios, where you don't have local subscriber or those are "sbc
subscribers with upper registration". 

The second one is global, it will be enabled for all you peers. The
advantage is that it won't send out you local calls. I use this for
migrations, where you still need to send the traffic of legacy class5
systems via the new Sipwise Class5.


In case you want to sell minutes to other peers and change them for it, the
above options have a problem: The rating engine is not prepared atm to bill
and charge peering servers. You'll get the cdrs and you will need to use an
external billing system for the charges. If this is your case you still have
a third option:

- Create your pstn gateways as peers. But other providers, you clients, the
  ones you want to charge create them as *subscribers*. Creating them as
  subscribers will allow you to rate them, apply ncos, prepaid (in case of
  pro) and lots of features not present in peer options. You just need to
  create a permanent contact for them, set them as trusted form the ip
  addresses they use and allow all the clis (if you want).


Please let me know which scenario you're looking for and if any of these
three options match it.

cheers,

Jon

 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130323/2ea74561/
attachment-0001.pgp>

------------------------------

Message: 5
Date: Sat, 23 Mar 2013 20:44:02 +0100
From: Jon Bonilla (Manwe) <jbonilla at sipwise.com>
Subject: Re: [Spce-user] Invoicing and Billing Peers/Trunks
To: spce-user at lists.sipwise.com
Message-ID: <20130323204402.4eeaf025 at quenya>
Content-Type: text/plain; charset="utf-8"

El Fri, 22 Mar 2013 15:16:19 +0200
Deon Vermeulen <vermeulen.deon at gmail.com> escribi?:

> Good Day
> 
> Is there any possibility that one could generate an Invoice to 
> customers with a Trunk to SPCE 2.7?


You can get the cdrs, but not an invoice as PDF is you mean that. Any
customer, has to be created as subscriber and it will be rated if you don't
disable it.

> 
> Is there any possibility to Generate a Bill to a Peering Partner for 
> calls sent to SPCE 2.7. i.e Reverse billing?
> 

Not at the moment. Peers can't be charged in this version of the spce.


cheers,

Jon
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130323/d3643d13/
attachment-0001.pgp>

------------------------------

Message: 6
Date: Sat, 23 Mar 2013 15:21:46 -0500 (CDT)
From: Marc Rys <m.rys at tri-lakes.net>
Subject: [Spce-user] 483 Too Many Hops
To: spce-user at lists.sipwise.com
Message-ID:
	
<c6ac07e3-43b0-4127-8a7d-8f562e0eef4c at usbsnpzimbra01v.servers.rystec.com>
	
Content-Type: text/plain; charset="utf-8"

I've been evaluating SPCE over the last week.  I've already setup a couple
test subscribers and setup a peer with a provider we work with for SIP LD
Term.  All of those test have worked very successful.

We also have our own media gateway which is interconnected with the local
PSTN via TDM trunk, but I send incoming calls from the PSTN through our
MediaGateway to SPCE, SPCE is responding back with 483 "Too Many Hops".
Attached is the wireshark cap.

Any help is appreciated.

Marc Rys
http://www.tri-lakes.net
http://www.rystec.com
-------------- next part --------------
A non-text attachment was scrubbed...
Name: SPCE-INBOUND-Taqua-483TooManyHops.pcap
Type: application/octet-stream
Size: 2220 bytes
Desc: not available
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130323/3e72b68f/
attachment-0001.obj>

------------------------------

Message: 7
Date: Sat, 23 Mar 2013 21:45:10 +0100
From: Jon Bonilla (Manwe) <jbonilla at sipwise.com>
Subject: Re: [Spce-user] 483 Too Many Hops
To: spce-user at lists.sipwise.com
Message-ID: <20130323214510.5498f5dc at quenya>
Content-Type: text/plain; charset="utf-8"

El Sat, 23 Mar 2013 15:21:46 -0500 (CDT) Marc Rys <m.rys at tri-lakes.net>
escribi?:

> En-Respuesta-A: <20130323204402.4eeaf025 at quenya>


Please don't answer a mail to create a new thread. Use the "new mail" button
instead. 


Cheers,

Jon

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130323/df575ddc/
attachment.pgp>

------------------------------

_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com
http://lists.sipwise.com/listinfo/spce-user


End of Spce-user Digest, Vol 28, Issue 52
*****************************************




From jbonilla at sipwise.com  Mon Mar 25 07:41:46 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Mon, 25 Mar 2013 12:41:46 +0100
Subject: [Spce-user] Connect softphone to SIPWISE (Jon Bonilla (Manwe))
In-Reply-To: <058901ce294a$9919d9d0$cb4d8d70$@sipalto.com>
References: <058901ce294a$9919d9d0$cb4d8d70$@sipalto.com>
Message-ID: <20130325124146.7c2f0804@quenya>

El Mon, 25 Mar 2013 11:19:20 -0000
"Ahmed Murad" <am at sipalto.com> escribi?:

> Hi,
> 
> Thanks for the info. I received help on connecting a UA to SIPWISE from
> another user who advised to run the below commands, after running them UA's
> connected without any issues, is it worth adding this to your documentation?
> 
> ngcp-network --set-interface=eth0 --ip=auto --netmask=auto
> ngcp-network --move-from=lo --move-to=eth0 --type=web_ext --type=sip_ext
> --type=rtp_ext --  -type=ssh_ext
> Ngcpfg apply
> 
> 


Those commands are already present in our documentation.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130325/9d2ba0e7/attachment-0001.asc>

From am at sipalto.com  Mon Mar 25 07:45:23 2013
From: am at sipalto.com (Ahmed Murad)
Date: Mon, 25 Mar 2013 11:45:23 -0000
Subject: [Spce-user] routing calls between peers (Jon Bonilla (Manwe))
Message-ID: <058a01ce294e$3cd0fc40$b672f4c0$@sipalto.com>

Hi Jon,

Many thanks for this info. For peering option 1 appears to be the one we
need. Billing is not an issue in this scenario and this spce is being used
as an SBC. We have 4 peers.

Peer 1 (inbound only), Peer 2(inbound only), Peer 3 & Peer 4.

Peer 1 sends inbound calls to the spce, we want to relay all Peer 1 calls to
Peer 4 which will then handle the call.
Peer 2 also sends inbound calls to the spce, we want to relay all calls
again to Peer 4.
Peer 3 and Peer 4 are our outbound Peers. Which subscriber calls will be
routed (inbound & outbound).

I can see the option ' force_outbound_calls_to_peer', how do I go about
setting the peer I want to force these calls to go to?

Again many thanks for your quick response, highly appreciated.

Best regards,

Ahmed

Ahmed Murad | SiPalto Ltd. 
Tel: 0207 148 7525? |? Fax: 0207 148? 7526? |? Email: am at sipalto.com
6-16 Arbutus St, London, E8 4DT | Website: www.sipalto.com
?



-----Original Message-----
From: spce-user-bounces at lists.sipwise.com
[mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of
spce-user-request at lists.sipwise.com
Sent: 23 March 2013 20:45
To: spce-user at lists.sipwise.com
Subject: Spce-user Digest, Vol 28, Issue 52

Send Spce-user mailing list submissions to
	spce-user at lists.sipwise.com

To subscribe or unsubscribe via the World Wide Web, visit
	http://lists.sipwise.com/listinfo/spce-user
or, via email, send a message with subject or body 'help' to
	spce-user-request at lists.sipwise.com

You can reach the person managing the list at
	spce-user-owner at lists.sipwise.com

When replying, please edit your Subject line so it is more specific than
"Re: Contents of Spce-user digest..."


Today's Topics:

   1. Re: Why does it take 35sec to register (Jon Bonilla (Manwe))
   2. Re: Sipwise over openvpn (Jon Bonilla (Manwe))
   3. Re: Connect softphone to SIPWISE (Jon Bonilla (Manwe))
   4. Re: routing calls between peers (Jon Bonilla (Manwe))
   5. Re: Invoicing and Billing Peers/Trunks (Jon Bonilla (Manwe))
   6. 483 Too Many Hops (Marc Rys)
   7. Re: 483 Too Many Hops (Jon Bonilla (Manwe))


----------------------------------------------------------------------

Message: 1
Date: Sat, 23 Mar 2013 20:21:10 +0100
From: Jon Bonilla (Manwe) <jbonilla at sipwise.com>
Subject: Re: [Spce-user] Why does it take 35sec to register
To: spce-user at lists.sipwise.com
Message-ID: <20130323202110.6f71ae8d at quenya>
Content-Type: text/plain; charset="utf-8"

El Sat, 23 Mar 2013 20:12:39 +0100
"Oliver Vermeulen" <oliver at oliverv.com> escribi?:

> Hello All,
> 
>  
> 
> I tested sipwise yesterday.. install was successful very easy.
> 
>  
> 
> Does anybody have problems with sip registration ?
> 
> Why does it take 35sec to register.
> 
>  
> 
> Also when it's registered and I dial takes long time to process call.
> 
> 

Please make a ngrep-sip capture of a registration and another one of a call.
We will be able to see how long takes.


-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130323/afee13bc/
attachment-0001.pgp>

------------------------------

Message: 2
Date: Sat, 23 Mar 2013 20:24:46 +0100
From: Jon Bonilla (Manwe) <jbonilla at sipwise.com>
Subject: Re: [Spce-user] Sipwise over openvpn
To: spce-user at lists.sipwise.com
Message-ID: <20130323202446.1bc5eeed at quenya>
Content-Type: text/plain; charset="utf-8"

El Sat, 23 Mar 2013 20:03:01 +0100
"Developing Tomorrow" <nh.developer at gmail.com> escribi?:

> Hi.,
> 
>  
> 
> I have "successfully" installed sipwise and am able to register users 
> and make calls using any "normal" providers.
> 
> Now
> 
> In the same box, I have created a tunnel to another network that has a 
> gateway.
> 
> I am able to ping that gateway from the station that has sipwise also 
> able to send traffic to the gateway, but I will get no answer because 
> of the ip used.
> 
> Let me put down the layout:
> 
> All the following ips are fake but will clearly show the layout.
> 
>  
> 
> Sipwise external ip: 200.200.200.200
> 
> Sipwise internal ip: 10.0.0.1
> 
>  
> 
> Tunnel router: 10.0.0.10
> 
> Gateway across tunel: 192.168.10.10
> 
>  
> 
> If I use a standard asterisk box, give it ip 10.0.1 and send calls to 
> gateway, the voice will come back thru tunnel because the gateway is 
> replying to 10.0.0.1
> 
>  
> 
> In my sipwise case, it does not work as sipwise is sending the call 
> with ip
> 200.200.200.200 so the gateway is answering thru the normal internet 
> and not the tunnel.
> 
>  
> 
> How can I configure sipwise to send info to that gateway using ip in 
> the form ttttttttt at 10.0.0.1 <mailto:ttttttttt at 10.0.0.1>  and not
> ttttttttt at 200.200.200.200 <mailto:ttttttttt at 200.200.200.200> .
> 
>  

Why don't you configure in the other side of the tunnel. "Send the traffic
to
20.200.200.200 via 10.0.0.1" ?

You can use the "extra_socket" option in the spce and that will work for you
at sip signaling level but at rtp level it won't work so you'd get signaling
via the tunel and media over internet. If that's what you want it's fine. 

Anyways I'd continue using the external main address and try to fix you
issue at networking level, with routes.


> 
> The media relay is supposed to do that function but I have no clue on 
> how to configure it.
> 
> I could only create a socket with ip 10.0.0.1 and that allows me to 
> send the traffic thru the vpn, but does not change the ip that the gateway
receives.
> 



cheers,

Jon

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130323/d6223bd4/
attachment-0001.pgp>

------------------------------

Message: 3
Date: Sat, 23 Mar 2013 20:26:05 +0100
From: Jon Bonilla (Manwe) <jbonilla at sipwise.com>
Subject: Re: [Spce-user] Connect softphone to SIPWISE
To: spce-user at lists.sipwise.com
Message-ID: <20130323202605.57ed0cdd at quenya>
Content-Type: text/plain; charset="utf-8"

El Sat, 23 Mar 2013 17:36:19 -0000
"Ahmed Murad" <am at sipalto.com> escribi?:

> Hello,
> 
>  
> 
> I have been following your guides and successfully created & connected 
> to a couple of external peers.
> 
>  
> 
> However I am having trouble connecting a softphone to our first
subscriber.
> I have created the domain, account and subscriber.
> 
>  
> 
> But I get a timeout 500 error when I try to connect the softphone to 
> the subscriber, I have ensured there are no firewall rules causing the
timeout.
> 
>  
> 
> Please could you point me in the right direction in regards to 
> troubleshooting, is there anything I've missed?
> 
>  
> 

Please do a ngre-sip capture in the systeem and let us check the
incoming/outgoing traffic.

Also, you can check /var/log/ngcp/kamaiio-proxy.log to check if REGISTER
requests are arriving and what's going on with them.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130323/8d7f84f5/
attachment-0001.pgp>

------------------------------

Message: 4
Date: Sat, 23 Mar 2013 20:40:18 +0100
From: Jon Bonilla (Manwe) <jbonilla at sipwise.com>
Subject: Re: [Spce-user] routing calls between peers
To: spce-user at lists.sipwise.com
Message-ID: <20130323204018.49f4428a at quenya>
Content-Type: text/plain; charset="utf-8"

El Sat, 23 Mar 2013 17:32:11 -0000
"Ahmed Murad" <am at sipalto.com> escribi?:

> Hello,
> 
>  
> 
> I am new to SIPWISE, do you have any guides on how to route calls from 
> 1 peer to another peer via the SIPWISE box?
> 
>  
> 
> Any guidance would be greatly appreciated.
> 
>  
> 
> Best regards,
> 
>  
>

Hi Ahmed

By default, the spce does not allow peer-peer calls. It only allows
peer-subscriber, subscriber-subscriber and subscriber-peer.

But you can indeed send calls from one peer to another. There are two ways.
Let me explain them and their differences:


- You have the option "force_outbound_calls_to_peer" you can set in the
proxy
  preferences in the admin interface. If this option is set, any incoming
call
  coming from this peer will be routed to pstn (to another or the same peer)
  based on the peer rules. This will be done even if the callee is a local
  subscriber.


- There's another option in config.yml which is called "allow_peer_relay".
  This is global and will work like this: When a call comes from a peer it
  will terminate the call locally if the calee is local or will send the
call
  to the pstn following the peer rules if not.


The first method has the advantage of being specific for a single peer and
you can enable it per peer. I use it for sbc scenarios or pure peer-peer
scenarios, where you don't have local subscriber or those are "sbc
subscribers with upper registration". 

The second one is global, it will be enabled for all you peers. The
advantage is that it won't send out you local calls. I use this for
migrations, where you still need to send the traffic of legacy class5
systems via the new Sipwise Class5.


In case you want to sell minutes to other peers and change them for it, the
above options have a problem: The rating engine is not prepared atm to bill
and charge peering servers. You'll get the cdrs and you will need to use an
external billing system for the charges. If this is your case you still have
a third option:

- Create your pstn gateways as peers. But other providers, you clients, the
  ones you want to charge create them as *subscribers*. Creating them as
  subscribers will allow you to rate them, apply ncos, prepaid (in case of
  pro) and lots of features not present in peer options. You just need to
  create a permanent contact for them, set them as trusted form the ip
  addresses they use and allow all the clis (if you want).


Please let me know which scenario you're looking for and if any of these
three options match it.

cheers,

Jon

 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130323/2ea74561/
attachment-0001.pgp>

------------------------------

Message: 5
Date: Sat, 23 Mar 2013 20:44:02 +0100
From: Jon Bonilla (Manwe) <jbonilla at sipwise.com>
Subject: Re: [Spce-user] Invoicing and Billing Peers/Trunks
To: spce-user at lists.sipwise.com
Message-ID: <20130323204402.4eeaf025 at quenya>
Content-Type: text/plain; charset="utf-8"

El Fri, 22 Mar 2013 15:16:19 +0200
Deon Vermeulen <vermeulen.deon at gmail.com> escribi?:

> Good Day
> 
> Is there any possibility that one could generate an Invoice to 
> customers with a Trunk to SPCE 2.7?


You can get the cdrs, but not an invoice as PDF is you mean that. Any
customer, has to be created as subscriber and it will be rated if you don't
disable it.

> 
> Is there any possibility to Generate a Bill to a Peering Partner for 
> calls sent to SPCE 2.7. i.e Reverse billing?
> 

Not at the moment. Peers can't be charged in this version of the spce.


cheers,

Jon
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130323/d3643d13/
attachment-0001.pgp>

------------------------------

Message: 6
Date: Sat, 23 Mar 2013 15:21:46 -0500 (CDT)
From: Marc Rys <m.rys at tri-lakes.net>
Subject: [Spce-user] 483 Too Many Hops
To: spce-user at lists.sipwise.com
Message-ID:
	
<c6ac07e3-43b0-4127-8a7d-8f562e0eef4c at usbsnpzimbra01v.servers.rystec.com>
	
Content-Type: text/plain; charset="utf-8"

I've been evaluating SPCE over the last week.  I've already setup a couple
test subscribers and setup a peer with a provider we work with for SIP LD
Term.  All of those test have worked very successful.

We also have our own media gateway which is interconnected with the local
PSTN via TDM trunk, but I send incoming calls from the PSTN through our
MediaGateway to SPCE, SPCE is responding back with 483 "Too Many Hops".
Attached is the wireshark cap.

Any help is appreciated.

Marc Rys
http://www.tri-lakes.net
http://www.rystec.com
-------------- next part --------------
A non-text attachment was scrubbed...
Name: SPCE-INBOUND-Taqua-483TooManyHops.pcap
Type: application/octet-stream
Size: 2220 bytes
Desc: not available
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130323/3e72b68f/
attachment-0001.obj>

------------------------------

Message: 7
Date: Sat, 23 Mar 2013 21:45:10 +0100
From: Jon Bonilla (Manwe) <jbonilla at sipwise.com>
Subject: Re: [Spce-user] 483 Too Many Hops
To: spce-user at lists.sipwise.com
Message-ID: <20130323214510.5498f5dc at quenya>
Content-Type: text/plain; charset="utf-8"

El Sat, 23 Mar 2013 15:21:46 -0500 (CDT) Marc Rys <m.rys at tri-lakes.net>
escribi?:

> En-Respuesta-A: <20130323204402.4eeaf025 at quenya>


Please don't answer a mail to create a new thread. Use the "new mail" button
instead. 


Cheers,

Jon

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL:
<http://lists.sipwise.com/pipermail/spce-user/attachments/20130323/df575ddc/
attachment.pgp>

------------------------------

_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com
http://lists.sipwise.com/listinfo/spce-user


End of Spce-user Digest, Vol 28, Issue 52
*****************************************




From jbonilla at sipwise.com  Mon Mar 25 07:56:49 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Mon, 25 Mar 2013 12:56:49 +0100
Subject: [Spce-user] routing calls between peers (Jon Bonilla (Manwe))
In-Reply-To: <058a01ce294e$3cd0fc40$b672f4c0$@sipalto.com>
References: <058a01ce294e$3cd0fc40$b672f4c0$@sipalto.com>
Message-ID: <20130325125649.5e328acf@quenya>

El Mon, 25 Mar 2013 11:45:23 -0000
"Ahmed Murad" <am at sipalto.com> escribi?:

> Hi Jon,
> 
> Many thanks for this info. For peering option 1 appears to be the one we
> need. Billing is not an issue in this scenario and this spce is being used
> as an SBC. We have 4 peers.
> 
> Peer 1 (inbound only), Peer 2(inbound only), Peer 3 & Peer 4.
> 
> Peer 1 sends inbound calls to the spce, we want to relay all Peer 1 calls to
> Peer 4 which will then handle the call.
> Peer 2 also sends inbound calls to the spce, we want to relay all calls
> again to Peer 4.
> Peer 3 and Peer 4 are our outbound Peers. Which subscriber calls will be
> routed (inbound & outbound).
> 
> I can see the option ' force_outbound_calls_to_peer', how do I go about
> setting the peer I want to force these calls to go to?
> 
> Again many thanks for your quick response, highly appreciated.
> 
>

If a call comes from a subscriber which has upper registration (sbc mode) the
call will go to the upper registration peer.

For any other call, peering rules will be used. 

In this particular case, for
calls coming form peers 1 and 2 I guess they will always have the same domain
(domain1 and domain2). If you set a peering rule in peer_group4 which matches
all the destinations but only domain1 and domain2 from domains you'll get that.
It's just using the peering rules, nothing out of the ordinary.


Sample:
Caller_Pattern: ^.*@domain1\.com$

cheers,

Jon
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130325/b0eb295e/attachment-0001.asc>

From sn at fest.dk  Mon Mar 25 08:17:39 2013
From: sn at fest.dk (Sylvester Nielsen)
Date: Mon, 25 Mar 2013 12:17:39 +0000
Subject: [Spce-user] Incoming calls from provider does not work
Message-ID: <E50575DD48D9B743925AC3BCE6310A4B243687F2@DB3PRD0710MB355.eurprd07.prod.outlook.com>

Hi,

I just installed and configured a new server. I have setup the routing to my provider.
Outbound call works as expected but inbound does not work. The trunk is without registration.

At first I got a message that the domain wasen't supported. I added to IP to the domain list, but then the server send a 401 Unauthorized message.
I tried to trace the kamailio logs and see this in the proxy log.
New request - M=INVITE R=sip:+4578788471 at 87.116.27.246 F=sip:+4522771051 at 212.98.66.168;user=phone T=sip:+4578788471 at 87.116.27.246 IP=212.98.66.168:11054 (127.0.0.1:5060) ID=BW131042756250313-108955888 at 212.98.65.15<mailto:ID=BW131042756250313-108955888 at 212.98.65.15>
Authentication failed, no credentials - R=sip:+4578788471 at 87.116.27.246 ID=BW131042756250313-108955888 at 212.98.65.15

The invite I get from the peer look like this:

T 212.98.66.168:11054 -> 87.116.27.246:5060 [AP]
INVITE sip:+4578788471 at 87.116.27.246 SIP/2.0.
Via: SIP/2.0/TCP 212.98.66.168:5060;branch=z9hG4bKvea1ve006o0hvpov25t0.1.
From: "+4522771051"<sip:+4522771051 at 212.98.66.168;user=phone>;tag=659644293-1364213720121-.
To: "4578788471 4578788471"<sip:+4578788471 at 87.116.27.246>.
Call-ID: BW131520121250313-842633845 at 212.98.65.15.
CSeq: 280801821 INVITE.
Contact: <sip:+4522771051 at 212.98.66.168:5060;transport=tcp>.
P-Asserted-Identity: "+4522771051"<sip:+4522771051 at 212.98.65.15;user=phone>.
Privacy: none.
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE.
Accept: application/dtmf,application/dtmf-relay,application/media_control+xml,application/sdp,multipart/mixed.
Supported: .
Max-Forwards: 29.
Content-Type: application/sdp.
Content-Length: 348.
.
v=0.
o=BroadWorks 5699654 1 IN IP4 212.98.66.168.
s=-.
c=IN IP4 212.98.66.168.
t=0 0.
m=audio 23772 RTP/AVP 8 0 18 99 100.
a=rtpmap:99 telephone-event/8000.
a=rtpmap:100 X-NSE/8000.
a=fmtp:99 0-15.
a=fmtp:100 200-202.
a=X-sqn:0.
a=X-cap:1 audio RTP/AVP 100.
a=X-cpar:a=rtpmap:100 X-NSE/8000.
a=X-cpar:a=fmtp:100 200-202.
a=X-cap:2 image udptl t38.

Med venlig hilsen

   Sylvester Nielsen

---------------------------------------------------------------

Fest Data A/S
Generatorvej 8D, 1.th
2730 Herlev

e-mail : sn at fest.dk<mailto:sn at fest.dk>

Tlf:     +45 4582 7389
Fax:   +45 4582 7387
Dir:    +45 6996 2011

http://www.festdata.dk

[Beskrivelse: Description: Description: Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse: cid:image003.jpg at 01CBCE06.BAABD4A0]




-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130325/e92f6b42/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.jpg
Type: image/jpeg
Size: 11834 bytes
Desc: image001.jpg
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130325/e92f6b42/attachment-0001.jpg>

From apogrebennyk at sipwise.com  Mon Mar 25 08:55:14 2013
From: apogrebennyk at sipwise.com (Andrew Pogrebennyk)
Date: Mon, 25 Mar 2013 13:55:14 +0100
Subject: [Spce-user] Incoming calls from provider does not work
In-Reply-To: <E50575DD48D9B743925AC3BCE6310A4B243687F2@DB3PRD0710MB355.eurprd07.prod.outlook.com>
References: <E50575DD48D9B743925AC3BCE6310A4B243687F2@DB3PRD0710MB355.eurprd07.prod.outlook.com>
Message-ID: <51504932.8030108@sipwise.com>

Hello Sylvester,
it could be that peer is using different IP addresses for origination
and termination. The calls are originated from this IP address:
212.98.66.168. Do you have a peer with this IP address?

On 03/25/2013 01:17 PM, Sylvester Nielsen wrote:
> Hi,
> 
>  
> 
> I just installed and configured a new server. I have setup the routing
> to my provider.
> 
> Outbound call works as expected but inbound does not work. The trunk is
> without registration.
> 
>  
> 
> At first I got a message that the domain wasen?t supported. I added to
> IP to the domain list, but then the server send a 401 Unauthorized message.
> 
> I tried to trace the kamailio logs and see this in the proxy log.
> 
> New request - M=INVITE R=sip:+4578788471 at 87.116.27.246
> F=sip:+4522771051 at 212.98.66.168;user=phone
> T=sip:+4578788471 at 87.116.27.246 IP=212.98.66.168:11054 (127.0.0.1:5060)
> ID=BW131042756250313-108955888 at 212.98.65.15
> <mailto:ID=BW131042756250313-108955888 at 212.98.65.15>
> 
> Authentication failed, no credentials - R=sip:+4578788471 at 87.116.27.246
> ID=BW131042756250313-108955888 at 212.98.65.15
> 
>  
> 
> The invite I get from the peer look like this:
> 
>  
> 
> T 212.98.66.168:11054 -> 87.116.27.246:5060 [AP]
> 
> INVITE sip:+4578788471 at 87.116.27.246 SIP/2.0.
> 
> Via: SIP/2.0/TCP 212.98.66.168:5060;branch=z9hG4bKvea1ve006o0hvpov25t0.1.
> 
> From:
> "+4522771051"<sip:+4522771051 at 212.98.66.168;user=phone>;tag=659644293-1364213720121-.
> 
> To: "4578788471 4578788471"<sip:+4578788471 at 87.116.27.246>.
> 
> Call-ID: BW131520121250313-842633845 at 212.98.65.15.
> 
> CSeq: 280801821 INVITE.
> 
> Contact: <sip:+4522771051 at 212.98.66.168:5060;transport=tcp>.
> 
> P-Asserted-Identity: "+4522771051"<sip:+4522771051 at 212.98.65.15;user=phone>.
> 
> Privacy: none.
> 
> Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE.
> 
> Accept:
> application/dtmf,application/dtmf-relay,application/media_control+xml,application/sdp,multipart/mixed.
> 
> Supported: .
> 
> Max-Forwards: 29.
> 
> Content-Type: application/sdp.
> 
> Content-Length: 348.
> 
> .
> 
> v=0.
> 
> o=BroadWorks 5699654 1 IN IP4 212.98.66.168.
> 
> s=-.
> 
> c=IN IP4 212.98.66.168.
> 
> t=0 0.
> 
> m=audio 23772 RTP/AVP 8 0 18 99 100.
> 
> a=rtpmap:99 telephone-event/8000.
> 
> a=rtpmap:100 X-NSE/8000.
> 
> a=fmtp:99 0-15.
> 
> a=fmtp:100 200-202.
> 
> a=X-sqn:0.
> 
> a=X-cap:1 audio RTP/AVP 100.
> 
> a=X-cpar:a=rtpmap:100 X-NSE/8000.
> 
> a=X-cpar:a=fmtp:100 200-202.
> 
> a=X-cap:2 image udptl t38.
> 
>  
> 
> Med venlig hilsen
> 
>  
> 
>    Sylvester Nielsen
> 
>  
> 
> ---------------------------------------------------------------
> 
> 	
> 
> Fest Data A/S
> Generatorvej 8D, 1.th
> 2730 Herlev
> 
> e-mail : sn at fest.dk <mailto:sn at fest.dk>
> 
> 	
> 
> Tlf:     +45 4582 7389
> Fax:   +45 4582 7387
> Dir:    +45 6996 2011
> 
> http://www.festdata.dk
> 
> 	
> 
> Beskrivelse: Description: Description: Beskrivelse: Beskrivelse:
> Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse:
> cid:image003.jpg at 01CBCE06.BAABD4A0
> 
> 		
> 
>  
> 
>  
> 
> 
> 
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
> 




From m.rys at tri-lakes.net  Mon Mar 25 20:09:49 2013
From: m.rys at tri-lakes.net (Marc Rys)
Date: Mon, 25 Mar 2013 19:09:49 -0500 (CDT)
Subject: [Spce-user] 483 Too Many Hops
In-Reply-To: <514F2D26.3030103@sipwise.com>
Message-ID: <ef498e89-2477-4559-bd77-a272859a326e@usbsnpzimbra01v.servers.rystec.com>

Hi Everybody,  Well I have news.. I switched my test phone from a grandstream to a yeahlink and now i can receive calls from my gateway..  Hooray!!!

Now the only problem I see is that when I try to send a call from my UA to my gateway i get a 403 forbidden from my gateway.  Not sure what its seeing and not liking.  Perhaps is this the same problem the AudioCodes have?

Attached is the pcap.

I truely appreciate all your help.

Marc Rys
http://www.tri-lakes.net
http://www.rystec.com



----- Original Message -----
From: "Andreas Granig" <agranig at sipwise.com>
To: spce-user at lists.sipwise.com
Sent: Sunday, March 24, 2013 11:43:18 AM
Subject: Re: [Spce-user] 483 Too Many Hops

Hi,

On 03/24/2013 05:25 PM, Lorenzo Mangani wrote:
> As you can see from your trace, the call is never completely
> established, as the 200 OK's ACK never makes it back to the user-agent.
> After a brief look this seems to be caused by the gateway always using
> the contact header to respond instead of the from header, as for the BYE
> message also being rejected for the same reason - just my 2 cents, might
> be wrong - maybe someone else on the list will have different input,
>

The Contact handling is as expected, but the 208.73.198.12 doesn't 
convert the Record-Route headers added by the SPCE in the 200-OK (packet 
#14) into Route headers of the ACK (packet #15), so the proxy has no 
information where to relay the ACK to (note that ACKs to 200 are relayed 
end-to-end, so all Route information must be preserved).

The sender of the ACK *must* take the Record-Route headers of the 
previous 200-OK, convert them to Routes and add them to the message in 
reverse order, and send the request to the URI found in the top-most Route.

Andreas

_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com
http://lists.sipwise.com/listinfo/spce-user
-------------- next part --------------
A non-text attachment was scrubbed...
Name: SPCE-OUTBOUND-Taqua-403Forbidden2.pcap
Type: application/octet-stream
Size: 10137 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130325/0cf0e1ce/attachment-0001.pcap>

From m.rys at tri-lakes.net  Mon Mar 25 20:15:06 2013
From: m.rys at tri-lakes.net (Marc Rys)
Date: Mon, 25 Mar 2013 19:15:06 -0500 (CDT)
Subject: [Spce-user] 483 Too Many Hops
In-Reply-To: <514F2D26.3030103@sipwise.com>
Message-ID: <eeb0a182-2a8c-4330-b200-a1bdd1b873cd@usbsnpzimbra01v.servers.rystec.com>

Hi Everybody,  Well I have news.. I switched my test phone from a grandstream to a yeahlink and now i can receive calls from my gateway..  Hooray!!!

Now the only problem I see is that when I try to send a call from my UA to my gateway i get a 403 forbidden from my gateway.
Attached is the pcap.

I truely appreciate all your help.

Marc Rys
http://www.tri-lakes.net
http://www.rystec.com



----- Original Message -----
From: "Andreas Granig" <agranig at sipwise.com>
To: spce-user at lists.sipwise.com
Sent: Sunday, March 24, 2013 11:43:18 AM
Subject: Re: [Spce-user] 483 Too Many Hops

Hi,

On 03/24/2013 05:25 PM, Lorenzo Mangani wrote:
> As you can see from your trace, the call is never completely
> established, as the 200 OK's ACK never makes it back to the user-agent.
> After a brief look this seems to be caused by the gateway always using
> the contact header to respond instead of the from header, as for the BYE
> message also being rejected for the same reason - just my 2 cents, might
> be wrong - maybe someone else on the list will have different input,
>

The Contact handling is as expected, but the 208.73.198.12 doesn't 
convert the Record-Route headers added by the SPCE in the 200-OK (packet 
#14) into Route headers of the ACK (packet #15), so the proxy has no 
information where to relay the ACK to (note that ACKs to 200 are relayed 
end-to-end, so all Route information must be preserved).

The sender of the ACK *must* take the Record-Route headers of the 
previous 200-OK, convert them to Routes and add them to the message in 
reverse order, and send the request to the URI found in the top-most Route.

Andreas

_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com
http://lists.sipwise.com/listinfo/spce-user



From jbonilla at sipwise.com  Tue Mar 26 05:05:29 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Tue, 26 Mar 2013 10:05:29 +0100
Subject: [Spce-user] Mailing list rule changes: Daily digest
Message-ID: <20130326100529.22401a25@quenya>

Hi all! This is for our beloved community

We've been using the spce-user mailing list for free, best effort support
during the last years. It has been more successful than expected in terms of
questions, patches and quality in general. Thanks to all of you for that! We
(Sipwise) find that the mailing list archive is a useful place to find answers
to common questions or find people that had the same issues in the past. Many
bug reports have been reported here too. That's why the mailing list is our
primary support method for our community.

To preserve the mailing list archive as a resource where people can search in
the future, we try to keep track of all threads and finish them for good or
bad. In that goal, we find that using the "daily digest" option to answer the
list for a particular thread breaks the thread and breaks our workflow where
every thread should be followed and closed via answers to a single mail. We
find that the spce-user mailing list is not a high traffic list. Receiving
every mail shouldn't be painful at all. For that reason, we plan to remove the
"receive a daily digest" option form the mailing list and make users to receive
a every single mail. This change would be effective in the following days.


As side notes, and for this same goal of preserving the list clean and useful,
I'd like to add some tips regarding the mailing list usage:

- Please *do not* use the "answer" button to create a new thread. Even if you
  delete the body and the subject of the mail, the mail you send will contain a
  "in-reply-to" header and will be ordered with the old thread in many
  mail-clients.

- Please use the "answer" button to answer an open thread. Don't use the
  forward button or "new mail" or any other buttons.

- Please consider that some people will read emails in plain text and not html.
  "Find my answer in blue" is not a good idea :)

- Please, try to keep the mailing list in CC instead of only answering the
  person. That makes the mailing list archive to have the full information. If
  there's sensible data and it can be replaced by another string do so.

In general, we're not too restrictive with the rules. RFC1855, the old
"netiquette" rules are more or less valid even nowadays.

Tip: To search our mailing list you can just google for 

$KEYWORD site:http://lists.sipwise.com/pipermail/spce-user/


As always, any comments are more than welcome. This is *your* mailing list and
we try to keep it useful for you.


cheers,

Jon

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130326/aba99bef/attachment-0001.asc>

From apogrebennyk at sipwise.com  Tue Mar 26 06:09:06 2013
From: apogrebennyk at sipwise.com (Andrew Pogrebennyk)
Date: Tue, 26 Mar 2013 11:09:06 +0100
Subject: [Spce-user] 483 Too Many Hops
In-Reply-To: <eeb0a182-2a8c-4330-b200-a1bdd1b873cd@usbsnpzimbra01v.servers.rystec.com>
References: <eeb0a182-2a8c-4330-b200-a1bdd1b873cd@usbsnpzimbra01v.servers.rystec.com>
Message-ID: <515173C2.1020506@sipwise.com>

Hi Marc,
well, the reason is outside of sip:provider. In your pcap the called
gateway/UA is sending 403 back to sip:provider. I can only guess that
maybe it needs a specific number in From or PAI or the given number
format is wrong.

On 03/26/2013 01:15 AM, Marc Rys wrote:
> Hi Everybody,  Well I have news.. I switched my test phone from a grandstream to a yeahlink and now i can receive calls from my gateway..  Hooray!!!
> 
> Now the only problem I see is that when I try to send a call from my UA to my gateway i get a 403 forbidden from my gateway.
> Attached is the pcap.
> 
> I truely appreciate all your help.
> 
> Marc Rys
> http://www.tri-lakes.net
> http://www.rystec.com




From rea1ity at kaist.ac.kr  Tue Mar 26 06:52:23 2013
From: rea1ity at kaist.ac.kr (=?UTF-8?B?7ZeI7Jyk?=)
Date: Tue, 26 Mar 2013 19:52:23 +0900 (KST)
Subject: [Spce-user] =?utf-8?q?PSTN_GW_can_register_SIPWISE_server_but_GW_?=
	=?utf-8?q?peer_status_always_busy=2E?=
Message-ID: <51517eaf3fe8_@_imoxion.com>

Hi,
I succeed register with SIP phones and PSTN Gateway on SIPWISE.

So it is working well each SIP phones it located in my private network.

But, When I tried to call telephone number as PSTN using by SIP phone, it returned busy signal as sound.

So now I trying to solve it could communicate with PSTN.

If you were me, how to find the problem that couldn't communicate with PSTN?

If you were me, where to configure in SIPWISE?

I'm also saw the log files it located /var/log/ngcp/ but still I don't know what is problem.

And please let me know what needed the file if you could check what is exactly problem of point.

David Ho
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130326/82abb898/attachment-0001.html>

From jbonilla at sipwise.com  Tue Mar 26 07:05:43 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Tue, 26 Mar 2013 12:05:43 +0100
Subject: [Spce-user] PSTN GW can register SIPWISE server but GW peer
 status always busy.
In-Reply-To: <51517eaf3fe8_@_imoxion.com>
References: <51517eaf3fe8_@_imoxion.com>
Message-ID: <20130326120543.1d753acd@quenya>

El Tue, 26 Mar 2013 19:52:23 +0900 (KST)
??<rea1ity at kaist.ac.kr> escribi?:

> Hi,
> I succeed register with SIP phones and PSTN Gateway on SIPWISE.

Why would you rgister a pstn gw to the spce? The Peers do not register on the
spce.

> 
> So it is working well each SIP phones it located in my private network.
> 
> But, When I tried to call telephone number as PSTN using by SIP phone, it
> returned busy signal as sound.
> 

That's not informational. What's the nwtwork capture? ngrep-sip? kamailio-proxy
logs?


> So now I trying to solve it could communicate with PSTN.
> 
> If you were me, how to find the problem that couldn't communicate with PSTN?
> 
> If you were me, where to configure in SIPWISE?
> 

Please read the handbook carefully and follow the steps.


> I'm also saw the log files it located /var/log/ngcp/ but still I don't know
> what is problem.
> 
> And please let me know what needed the file if you could check what is
> exactly problem of point.
> 


cheers,

Jon
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130326/7a992dd6/attachment-0001.asc>

From agranig at sipwise.com  Tue Mar 26 09:09:50 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Tue, 26 Mar 2013 14:09:50 +0100
Subject: [Spce-user] PSTN GW can register SIPWISE server but GW peer
 status always busy.
In-Reply-To: <51517eaf3fe8_@_imoxion.com>
References: <51517eaf3fe8_@_imoxion.com>
Message-ID: <51519E1E.3070300@sipwise.com>

Hi,

On 03/26/2013 11:52 AM, ?? wrote:
> So now I trying to solve it could communicate with PSTN.
>
> If you were me, how to find the problem that couldn't communicate with PSTN?
>
> If you were me, where to configure in SIPWISE?

Well, do you already have a SIP trunk to some peering provider? If yes, 
did you configure it in the "SIP Peering" section of the admin panel 
(check the handbook for the how-to)? If yes, what did you put there and 
where, and what does the kamailio-proxy.log say for a call towards this 
peer?

Andreas



From jbonilla at sipwise.com  Tue Mar 26 09:16:26 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Tue, 26 Mar 2013 14:16:26 +0100
Subject: [Spce-user] Mailing list rule changes: Daily digest
In-Reply-To: <20130326100529.22401a25@quenya>
References: <20130326100529.22401a25@quenya>
Message-ID: <20130326141626.5436c526@quenya>

El Tue, 26 Mar 2013 10:05:29 +0100
Jon Bonilla (Manwe) <jbonilla at sipwise.com> escribi?:
 
> For that reason, we plan to remove the
> "receive a daily digest" option form the mailing list and make users to
> receive a every single mail. This change would be effective in the following
> days.


Actually, I have been told that the digest option has already been
disabled. Prepare/adjust your inbox filters. 

:)



-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130326/a7438c8b/attachment-0001.asc>

From rea1ity at kaist.ac.kr  Tue Mar 26 21:15:08 2013
From: rea1ity at kaist.ac.kr (=?UTF-8?B?RGF2aWQgSG8=?=)
Date: Wed, 27 Mar 2013 10:15:08 +0900 (KST)
Subject: [Spce-user]
	=?utf-8?q?PSTN_GW_can_register_SIPWISE_server_but_GW_?=
	=?utf-8?q?peerstatus_always_busy=2E?=
Message-ID: <515248863fe3_@_imoxion.com>

I have already a PSTN Gateway it is not a SIP trunk but it can convert SIP to PSTN.
And I the PSTN Gateway IP set private network. i.e. 192.168.102.254
In "SIP Peering" section of admin panel, I made "test group" and there have "test-gw-1" server it configured same like following;
-------------------
Name : test-gw-1
IP Address : 192.168.102.254 ->it is same IP address with PSTN Gateway because I want to use this PSTN.
Hostname : " "(empty)
Port : UDP
Weight : 1
-------------------
Also Peering Rules values it configured all empty. Callee Prefix, Callee Pattern, Caller Pattern, Description.
And as I opened the log file Kamailio-proxy.log, there are so many logs made at just one call.
So I attached two log files one is a call from SIP phone to SIP phone as locally and the other one is a call from SIP Phone to PSTN Gateway.
I don't know exactly what is point of problem in the log files why PSTN status is always busy.
But as I had inference, it is point why didn't call from SIP phone to PSTN Gateway and it status is always busy.
------------------
Mar 27 01:21:25 spce /usr/sbin/kamailio[1994]: INFO: <script>: No matching rewrite rules for '0112796171' found - R=sip:0112796171 at 192.168.102.254:5060;transport=udp ID=d8bb68a825c52d88dc2813b4e77ace43 at 192.168.102.60
Mar 27 01:21:25 spce /usr/sbin/kamailio[1994]: INFO: <script>: Setting P-Called-Party-ID '<sip:0112796171 at 192.168.102.254>' - R=sip:0112796171 at 192.168.102.254:5060;transport=udp ID=d8bb68a825c52d88dc2813b4e77ace43 at 192.168.102.60
Mar 27 01:21:25 spce /usr/sbin/kamailio[1994]: INFO: <script>: Writing sbc parameters  ;aleg_sst_enable=no;sst_enable=no - R=sip:0112796171 at 192.168.102.254:5060;transport=udp ID=d8bb68a825c52d88dc2813b4e77ace43 at 192.168.102.60
Mar 27 01:21:25 spce /usr/sbin/kamailio[1994]: INFO: <script>: Setting 'sip:192.168.102.254:5060' taken from R-URI as next hop after lb for PSTN call - R=sip:0112796171 at 192.168.102.254:5060;transport=udp ID=d8bb68a825c52d88dc2813b4e77ace43 at 192.168.102.60
Mar 27 01:21:25 spce /usr/sbin/kamailio[1994]: INFO: <script>: Appending P-D-URI 'sip:127.0.0.1:5060;received='sip:192.168.102.254:5060;transport=udp'' - R=sip:0112796171 at 192.168.102.254:5060;transport=udp ID=d8bb68a825c52d88dc2813b4e77ace43 at 192.168.102.60
Mar 27 01:21:25 spce /usr/sbin/kamailio[1994]: INFO: <script>: Forcing request via B2BUA 'sip:127.0.0.1:5080' - R=sip:0112796171 at 192.168.102.254:5060;transport=udp ID=d8bb68a825c52d88dc2813b4e77ace43 at 192.168.102.60
Mar 27 01:21:25 spce /usr/sbin/kamailio[1994]: INFO: <script>: Setting P-Caller-UUID to '0cbfd478-f7eb-445d-9f3c-6061b0f96eb2' - R=sip:0112796171 at 192.168.102.254:5060;transport=udp ID=d8bb68a825c52d88dc2813b4e77ace43 at 192.168.102.60
Mar 27 01:21:25 spce /usr/sbin/kamailio[1994]: INFO: <script>: Setting P-Callee-UUID to '0' - R=sip:0112796171 at 192.168.102.254:5060;transport=udp ID=d8bb68a825c52d88dc2813b4e77ace43 at 192.168.102.60
Mar 27 01:21:25 spce /usr/sbin/kamailio[1994]: INFO: <script>: Request leaving server, D-URI='sip:127.0.0.1:5080' - R=sip:0112796171 at 192.168.102.254:5060;transport=udp ID=d8bb68a825c52d88dc2813b4e77ace43 at 192.168.102.60
Mar 27 01:21:25 spce /usr/sbin/kamailio[1995]: INFO: <script>: NAT-Reply - S=100 - Connecting M=INVITE IP=192.168.102.60:5060 (127.0.0.1:5080) ID=d8bb68a825c52d88dc2813b4e77ace43 at 192.168.102.60
Mar 27 01:21:25 spce /usr/sbin/kamailio[1992]: INFO: <script>: NAT-Reply - S=486 - Busy Here M=INVITE IP=192.168.102.60:5060 (127.0.0.1:5080) ID=d8bb68a825c52d88dc2813b4e77ace43 at 192.168.102.60
Mar 27 01:21:25 spce /usr/sbin/kamailio[1992]: INFO: <script>: Failure route for PSTN call - R=sip:0112796171 at sip.baekdoo.local:5060;transport=udp ID=d8bb68a825c52d88dc2813b4e77ace43 at 192.168.102.60
------------------------


So I don't know exactly but the SIP Server tried to call the PSTN Gateway as the SIPWISE server IP 127.0.0.1:5080 as local SIP it referenced following a log in upper logs.
-------------------------------
Forcing request via B2BUA 'sip:127.0.0.1:5080' - R=sip:0112796171 at 192.168.102.254:5060;transport=udp
-------------------------------

But SIPWISE server couldn't reached to PSTN gateway because IP networks are different each others.
So SIPWISE can't reached the network 192.168.102.254:5060 that is only my opinion.


Please let me know if you find what was problem from connect with from SIP phone to PSTN network.


David Ho




----------------------????-----------------------
???? : Andreas Granig <agranig at sipwise.com>
????: spce-user at lists.sipwise.com
????: 2013-03-26 22:09:50 GMT +0900 (Asia/Seoul)
??: Re: [Spce-user] PSTN GW can register SIPWISE server but GW peerstatus always busy.

Hi,

On 03/26/2013 11:52 AM, ?? wrote:
> So now I trying to solve it could communicate with PSTN.
>
> If you were me, how to find the problem that couldn't communicate with PSTN?
>
> If you were me, where to configure in SIPWISE?

Well, do you already have a SIP trunk to some peering provider? If yes, 
did you configure it in the "SIP Peering" section of the admin panel 
(check the handbook for the how-to)? If yes, what did you put there and 
where, and what does the kamailio-proxy.log say for a call towards this 
peer?

Andreas

_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com
http://lists.sipwise.com/listinfo/spce-user


-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130327/1ca17207/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: SIP_to_SIP_Locally.txt
Type: application/octet-stream
Size: 17754 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130327/1ca17207/attachment-0002.txt>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: SIP_to_PSTN_GW.txt
Type: application/octet-stream
Size: 16571 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130327/1ca17207/attachment-0003.txt>

From agranig at sipwise.com  Wed Mar 27 04:34:36 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Wed, 27 Mar 2013 09:34:36 +0100
Subject: [Spce-user] PSTN GW can register SIPWISE server but GW
 peerstatus always busy.
In-Reply-To: <515248833fda_@_imoxion.com>
References: <515248833fda_@_imoxion.com>
Message-ID: <5152AF1C.5000601@mail.sipwise.com>

Hi,

On 03/27/2013 02:15 AM, David Ho wrote:
> But SIPWISE server couldn't reached to PSTN gateway because IP
> networks are different each others.
>
> So SIPWISE can't reached the network 192.168.102.254:5060 that is only
> my opinion.

Well, since you get this line in the logs:

NAT-Reply - S=486 - Busy Here M=INVITE IP=192.168.102.60:5060

it clearly indicates that your GW is in fact getting the request, but as 
logged, answers with Busy. You need to check the logs of your peer/gw to 
see why it does so.

Maybe it doesn't like some headers, number formats or similar, but we 
can just guess without knowing what the GW says in its logs.

Andreas



From sn at fest.dk  Wed Mar 27 10:56:37 2013
From: sn at fest.dk (Sylvester Nielsen)
Date: Wed, 27 Mar 2013 14:56:37 +0000
Subject: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
Message-ID: <E50575DD48D9B743925AC3BCE6310A4B2436E488@DB3PRD0710MB355.eurprd07.prod.outlook.com>

Hi,

When we send an invite to the peer we have ngcpct='sip:127.0.0.1:5080' in the contact header.
Can someone tell me what this is and if we can remove it? For some reason the peer think it should send BYE and CANCEL to this address which makes calls hang.

Med venlig hilsen

   Sylvester Nielsen

---------------------------------------------------------------

Fest Data A/S
Generatorvej 8D, 1.th
2730 Herlev

e-mail : sn at fest.dk<mailto:sn at fest.dk>

Tlf:     +45 4582 7389
Fax:   +45 4582 7387
Dir:    +45 6996 2011

http://www.festdata.dk

[Beskrivelse: Description: Description: Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse: cid:image003.jpg at 01CBCE06.BAABD4A0]




-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130327/7e2a023b/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.jpg
Type: image/jpeg
Size: 11834 bytes
Desc: image001.jpg
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130327/7e2a023b/attachment-0001.jpg>

From agranig at sipwise.com  Wed Mar 27 11:01:23 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Wed, 27 Mar 2013 16:01:23 +0100
Subject: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
In-Reply-To: <E50575DD48D9B743925AC3BCE6310A4B2436E488@DB3PRD0710MB355.eurprd07.prod.outlook.com>
References: <E50575DD48D9B743925AC3BCE6310A4B2436E488@DB3PRD0710MB355.eurprd07.prod.outlook.com>
Message-ID: <515309C3.1080601@mail.sipwise.com>

Hi,

Any URI params in the Contact header must to be preserved for correct 
routing, otherwise you're opening a huge can of worms.

Andreas

On 03/27/2013 03:56 PM, Sylvester Nielsen wrote:
> Hi,
>
> When we send an invite to the peer we have ngcpct='sip:127.0.0.1:5080'
> in the contact header.
>
> Can someone tell me what this is and if we can remove it? For some
> reason the peer think it should send BYE and CANCEL to this address
> which makes calls hang.
>
> Med venlig hilsen
>
>     Sylvester Nielsen
>
> ---------------------------------------------------------------
>
> 	
>
> Fest Data A/S
> Generatorvej 8D, 1.th
> 2730 Herlev
>
> e-mail : sn at fest.dk <mailto:sn at fest.dk>
>
> 	
>
> Tlf:     +45 4582 7389
> Fax:   +45 4582 7387
> Dir:    +45 6996 2011
>
> http://www.festdata.dk
>
> 	
>
> Beskrivelse: Description: Description: Beskrivelse: Beskrivelse:
> Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse:
> cid:image003.jpg at 01CBCE06.BAABD4A0
>
> 		
>
>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
>



From john.murray at skyracktelecom.com  Wed Mar 27 11:11:48 2013
From: john.murray at skyracktelecom.com (John Murray)
Date: Wed, 27 Mar 2013 15:11:48 -0000
Subject: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
In-Reply-To: <515309C3.1080601@mail.sipwise.com>
References: <E50575DD48D9B743925AC3BCE6310A4B2436E488@DB3PRD0710MB355.eurprd07.prod.outlook.com>
	<515309C3.1080601@mail.sipwise.com>
Message-ID: <51530c35.a266b40a.3a4f.ffffb1e3@mx.google.com>

Andreas,

With a peer you are typically in some sort of relationship and can negotiate
any fixes (not always the case of course). But with a subscriber you are
dependent on the code in the device.
In my experience with PBXs ( naming no names (but NEC is terrible)) are very
bad with this header so it would be very helpful if there were some switch
against the subscriber to remove this as a workaround.
Getting the vendor to do something about this has always proven very
difficult/painful/slow.

Regards

John 

-----Original Message-----
From: spce-user-bounces at lists.sipwise.com
[mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of Andreas Granig
Sent: 27 March 2013 15:01
To: spce-user at lists.sipwise.com
Subject: Re: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header

Hi,

Any URI params in the Contact header must to be preserved for correct
routing, otherwise you're opening a huge can of worms.

Andreas

On 03/27/2013 03:56 PM, Sylvester Nielsen wrote:
> Hi,
>
> When we send an invite to the peer we have ngcpct='sip:127.0.0.1:5080'
> in the contact header.
>
> Can someone tell me what this is and if we can remove it? For some 
> reason the peer think it should send BYE and CANCEL to this address 
> which makes calls hang.
>
> Med venlig hilsen
>
>     Sylvester Nielsen
>
> ---------------------------------------------------------------
>
> 	
>
> Fest Data A/S
> Generatorvej 8D, 1.th
> 2730 Herlev
>
> e-mail : sn at fest.dk <mailto:sn at fest.dk>
>
> 	
>
> Tlf:     +45 4582 7389
> Fax:   +45 4582 7387
> Dir:    +45 6996 2011
>
> http://www.festdata.dk
>
> 	
>
> Beskrivelse: Description: Description: Beskrivelse: Beskrivelse:
> Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse:
> cid:image003.jpg at 01CBCE06.BAABD4A0
>
> 		
>
>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
>

_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com
http://lists.sipwise.com/listinfo/spce-user




From gary at netdiverse.com  Wed Mar 27 11:39:20 2013
From: gary at netdiverse.com (Gary Nieboer)
Date: Wed, 27 Mar 2013 09:39:20 -0600
Subject: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
In-Reply-To: <515309C3.1080601@mail.sipwise.com>
References: <E50575DD48D9B743925AC3BCE6310A4B2436E488@DB3PRD0710MB355.eurprd07.prod.outlook.com>
	<515309C3.1080601@mail.sipwise.com>
Message-ID: <CAMXUi2TGRE3b8FaRZSi_nmtAfKg_ii7Q-JYsPebsvauC0iM_Xg@mail.gmail.com>

Andreas,

We have the same issue with Level 3 Communications / Global Crossing on
their Enterprise SIP Platform.   As background, they have a Acme Packet SBC
in front of their Broadsoft App Server.

When the party on the Peer/Level3 side of the network sends a BYE, it never
reaches the subscriber on the SPCE side and hangs.  If the SPCE subscriber
sends the BYE, then it appears to work properly.

>From everything we can see and with feedback from Level , their SBC or App
Server appears to be getting confused by the address SPCE puts in the
contact ID field.   Is there a setting that will put an additional or
different IP address in the Contact ID field?    I likely can't get a
packet capture until Thursday night (US-EDT), but would be happy to provide
one privately later in the week.

-Gary


--
Gary Nieboer


877-763-1014 Toll Free  |  801-657-5484 Utah  |  615-800-6946 Tennessee  |
801-694-6106 Mobile
gary at netdiverse.com  |   www.netdiverse.com


On Wed, Mar 27, 2013 at 9:01 AM, Andreas Granig <agranig at sipwise.com> wrote:

> Hi,
>
> Any URI params in the Contact header must to be preserved for correct
> routing, otherwise you're opening a huge can of worms.
>
> Andreas
>
>
> On 03/27/2013 03:56 PM, Sylvester Nielsen wrote:
>
>> Hi,
>>
>> When we send an invite to the peer we have ngcpct='sip:127.0.0.1:5080'
>> in the contact header.
>>
>> Can someone tell me what this is and if we can remove it? For some
>> reason the peer think it should send BYE and CANCEL to this address
>> which makes calls hang.
>>
>> Med venlig hilsen
>>
>>     Sylvester Nielsen
>>
>> ------------------------------**------------------------------**---
>>
>>
>>
>> Fest Data A/S
>> Generatorvej 8D, 1.th
>> 2730 Herlev
>>
>> e-mail : sn at fest.dk <mailto:sn at fest.dk>
>>
>>
>>
>>
>> Tlf:     +45 4582 7389
>> Fax:   +45 4582 7387
>> Dir:    +45 6996 2011
>>
>> http://www.festdata.dk
>>
>>
>>
>> Beskrivelse: Description: Description: Beskrivelse: Beskrivelse:
>> Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse:
>> cid:image003.jpg at 01CBCE06.**BAABD4A0
>>
>>
>>
>>
>>
>> ______________________________**_________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/**listinfo/spce-user<http://lists.sipwise.com/listinfo/spce-user>
>>
>>
> ______________________________**_________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/**listinfo/spce-user<http://lists.sipwise.com/listinfo/spce-user>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130327/ae023c36/attachment-0001.html>

From agranig at sipwise.com  Wed Mar 27 11:41:14 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Wed, 27 Mar 2013 16:41:14 +0100
Subject: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
In-Reply-To: <51530c35.a266b40a.3a4f.ffffb1e3@mx.google.com>
References: <E50575DD48D9B743925AC3BCE6310A4B2436E488@DB3PRD0710MB355.eurprd07.prod.outlook.com>
	<515309C3.1080601@mail.sipwise.com>
	<51530c35.a266b40a.3a4f.ffffb1e3@mx.google.com>
Message-ID: <5153131A.7060002@mail.sipwise.com>

Hi,

On 03/27/2013 04:11 PM, John Murray wrote:
> With a peer you are typically in some sort of relationship and can negotiate
> any fixes (not always the case of course). But with a subscriber you are
> dependent on the code in the device.
> In my experience with PBXs ( naming no names (but NEC is terrible)) are very
> bad with this header so it would be very helpful if there were some switch
> against the subscriber to remove this as a workaround.
> Getting the vendor to do something about this has always proven very
> difficult/painful/slow.

Well, the reason this has been added was to mask the local sems contact 
of 127.0.0.1:5080, because some peers are doing far-end nat traversal 
(like the SPCE itself), and if some peer tries to fix this, the SIP flow 
is again broken.

So in one way or another, this information needs to be coded into the 
message. I'm open for any better suggestions of course :)

Andreas



From john.murray at skyracktelecom.com  Wed Mar 27 11:52:40 2013
From: john.murray at skyracktelecom.com (John Murray)
Date: Wed, 27 Mar 2013 15:52:40 -0000
Subject: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
In-Reply-To: <CAMXUi2TGRE3b8FaRZSi_nmtAfKg_ii7Q-JYsPebsvauC0iM_Xg@mail.gmail.com>
References: <E50575DD48D9B743925AC3BCE6310A4B2436E488@DB3PRD0710MB355.eurprd07.prod.outlook.com>
	<515309C3.1080601@mail.sipwise.com>
	<CAMXUi2TGRE3b8FaRZSi_nmtAfKg_ii7Q-JYsPebsvauC0iM_Xg@mail.gmail.com>
Message-ID: <515315ca.2560b40a.32bb.ffffc590@mx.google.com>

Gary,

 

Check out the Audiocodes workaround in the handbook section 5.5.1 if the
problem in the address itself being 'loopback'

We use our external address with the first octet replaced with '10' for
internal communications. It makes debugging easier.

This fixes a number of systems getting confused.

 

Regards

 

John

 

From: spce-user-bounces at lists.sipwise.com
[mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of Gary Nieboer
Sent: 27 March 2013 15:39
To: spce-user at lists.sipwise.com
Subject: Re: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header

 

Andreas, 

 

We have the same issue with Level 3 Communications / Global Crossing on
their Enterprise SIP Platform.   As background, they have a Acme Packet SBC
in front of their Broadsoft App Server.

 

When the party on the Peer/Level3 side of the network sends a BYE, it never
reaches the subscriber on the SPCE side and hangs.  If the SPCE subscriber
sends the BYE, then it appears to work properly. 

 

>From everything we can see and with feedback from Level , their SBC or App
Server appears to be getting confused by the address SPCE puts in the
contact ID field.   Is there a setting that will put an additional or
different IP address in the Contact ID field?    I likely can't get a packet
capture until Thursday night (US-EDT), but would be happy to provide one
privately later in the week.

 

-Gary

 

 

--
Gary Nieboer

  <http://www.netdiverse.com/content/images/NewNDLogo180wide.png> 


877-763-1014 Toll Free  |  801-657-5484 Utah  |  615-800-6946 Tennessee  |
801-694-6106 Mobile
gary at netdiverse.com  |   www.netdiverse.com <http://www.netdiverse.com/> 

 

On Wed, Mar 27, 2013 at 9:01 AM, Andreas Granig <agranig at sipwise.com> wrote:

Hi,

Any URI params in the Contact header must to be preserved for correct
routing, otherwise you're opening a huge can of worms.

Andreas



On 03/27/2013 03:56 PM, Sylvester Nielsen wrote:

Hi,

When we send an invite to the peer we have ngcpct='sip:127.0.0.1:5080'
in the contact header.

Can someone tell me what this is and if we can remove it? For some
reason the peer think it should send BYE and CANCEL to this address
which makes calls hang.

Med venlig hilsen

    Sylvester Nielsen

---------------------------------------------------------------

        

Fest Data A/S
Generatorvej 8D, 1.th
2730 Herlev

e-mail : sn at fest.dk <mailto:sn at fest.dk>



        

Tlf:     +45 4582 7389 <tel:%2B45%204582%207389> 
Fax:   +45 4582 7387 <tel:%2B45%204582%207387> 
Dir:    +45 6996 2011 <tel:%2B45%206996%202011> 

http://www.festdata.dk

        

Beskrivelse: Description: Description: Beskrivelse: Beskrivelse:
Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse: Beskrivelse:
cid:image003.jpg at 01CBCE06.BAABD4A0

                




_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com
http://lists.sipwise.com/listinfo/spce-user


_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com
http://lists.sipwise.com/listinfo/spce-user

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130327/26d8e552/attachment-0001.html>

From skchopperguy at gmail.com  Wed Mar 27 11:55:20 2013
From: skchopperguy at gmail.com (Skyler)
Date: Wed, 27 Mar 2013 08:55:20 -0700
Subject: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
In-Reply-To: <5153131A.7060002@mail.sipwise.com>
References: <E50575DD48D9B743925AC3BCE6310A4B2436E488@DB3PRD0710MB355.eurprd07.prod.outlook.com>
	<515309C3.1080601@mail.sipwise.com>
	<51530c35.a266b40a.3a4f.ffffb1e3@mx.google.com>
	<5153131A.7060002@mail.sipwise.com>
Message-ID: <51531668.2090106@gmail.com>

Hi,

On 3/27/2013 8:41 AM, Andreas Granig wrote:
> Hi,
>
> On 03/27/2013 04:11 PM, John Murray wrote:
>> With a peer you are typically in some sort of relationship and can
>> negotiate
>> any fixes (not always the case of course). But with a subscriber you are
>> dependent on the code in the device.
>> In my experience with PBXs ( naming no names (but NEC is terrible))
>> are very
>> bad with this header so it would be very helpful if there were some
>> switch
>> against the subscriber to remove this as a workaround.
>> Getting the vendor to do something about this has always proven very
>> difficult/painful/slow.
>
> Well, the reason this has been added was to mask the local sems contact
> of 127.0.0.1:5080, because some peers are doing far-end nat traversal
> (like the SPCE itself), and if some peer tries to fix this, the SIP flow
> is again broken.
>
> So in one way or another, this information needs to be coded into the
> message. I'm open for any better suggestions of course :)
>
> Andreas
>

  I created an lo:0 with a private IP address (10.10.xxx.xxx) for lb > 
b2buas: & proxies > sip:. That may help in this case as its likely the 
127.0.0.1 which is confusing the upstream proxy.

Skyler



From john.murray at skyracktelecom.com  Wed Mar 27 12:03:21 2013
From: john.murray at skyracktelecom.com (John Murray)
Date: Wed, 27 Mar 2013 16:03:21 -0000
Subject: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
In-Reply-To: <51531668.2090106@gmail.com>
References: <E50575DD48D9B743925AC3BCE6310A4B2436E488@DB3PRD0710MB355.eurprd07.prod.outlook.com>
	<515309C3.1080601@mail.sipwise.com>
	<51530c35.a266b40a.3a4f.ffffb1e3@mx.google.com>
	<5153131A.7060002@mail.sipwise.com> <51531668.2090106@gmail.com>
Message-ID: <5153184b.8a0db40a.5bf5.ffffc4e1@mx.google.com>

Yes one problem with some vendors is the loopback address of 127.0.0.1 which
as you say can be fixed by changing the internal address as described in
handbook section 5.5.1.
The other problem some vendors have is with the 'ngcpct=' section of the
Contact: header as:

Contact: <sip:ngcp-lb at 10.23.11.18:5060;ngcpct='sip:10.23.11.18:5080'>'

Some vendors 'forget' this and reply to sip:ngcp-lb at 10.23.11.18:5060 and
some get stuck and don't reply at all when they see this.

Regards

John

-----Original Message-----
From: Skyler [mailto:skchopperguy at gmail.com] 
Sent: 27 March 2013 15:55
To: Andreas Granig
Cc: John Murray; spce-user at lists.sipwise.com
Subject: Re: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header

Hi,

On 3/27/2013 8:41 AM, Andreas Granig wrote:
> Hi,
>
> On 03/27/2013 04:11 PM, John Murray wrote:
>> With a peer you are typically in some sort of relationship and can 
>> negotiate any fixes (not always the case of course). But with a 
>> subscriber you are dependent on the code in the device.
>> In my experience with PBXs ( naming no names (but NEC is terrible)) 
>> are very bad with this header so it would be very helpful if there 
>> were some switch against the subscriber to remove this as a 
>> workaround.
>> Getting the vendor to do something about this has always proven very 
>> difficult/painful/slow.
>
> Well, the reason this has been added was to mask the local sems 
> contact of 127.0.0.1:5080, because some peers are doing far-end nat 
> traversal (like the SPCE itself), and if some peer tries to fix this, 
> the SIP flow is again broken.
>
> So in one way or another, this information needs to be coded into the 
> message. I'm open for any better suggestions of course :)
>
> Andreas
>

  I created an lo:0 with a private IP address (10.10.xxx.xxx) for lb >
b2buas: & proxies > sip:. That may help in this case as its likely the
127.0.0.1 which is confusing the upstream proxy.

Skyler




From agranig at sipwise.com  Wed Mar 27 12:07:11 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Wed, 27 Mar 2013 17:07:11 +0100
Subject: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
In-Reply-To: <51531668.2090106@gmail.com>
References: <E50575DD48D9B743925AC3BCE6310A4B2436E488@DB3PRD0710MB355.eurprd07.prod.outlook.com>
	<515309C3.1080601@mail.sipwise.com>
	<51530c35.a266b40a.3a4f.ffffb1e3@mx.google.com>
	<5153131A.7060002@mail.sipwise.com> <51531668.2090106@gmail.com>
Message-ID: <5153192F.4050106@mail.sipwise.com>

Hi,

On 03/27/2013 04:55 PM, Skyler wrote:
>   I created an lo:0 with a private IP address (10.10.xxx.xxx) for lb >
> b2buas: & proxies > sip:. That may help in this case as its likely the
> 127.0.0.1 which is confusing the upstream proxy.

I've heard of endpoints disliking the SIP uri with the quotes... so one 
option could be to convert the "ngcpct='sip:127.0.0.1:5080'" and 
base64-encode the "sip:127.0.0.1:5080", so it gets something like 
"ngcpct=c2lwOjEyNy4wLjAuMTo1MDgwCg==" instead.

Andreas



From skchopperguy at gmail.com  Wed Mar 27 12:09:03 2013
From: skchopperguy at gmail.com (Skyler)
Date: Wed, 27 Mar 2013 09:09:03 -0700
Subject: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
In-Reply-To: <5153184b.8a0db40a.5bf5.ffffc4e1@mx.google.com>
References: <E50575DD48D9B743925AC3BCE6310A4B2436E488@DB3PRD0710MB355.eurprd07.prod.outlook.com>
	<515309C3.1080601@mail.sipwise.com>
	<51530c35.a266b40a.3a4f.ffffb1e3@mx.google.com>
	<5153131A.7060002@mail.sipwise.com> <51531668.2090106@gmail.com>
	<5153184b.8a0db40a.5bf5.ffffc4e1@mx.google.com>
Message-ID: <5153199F.2020608@gmail.com>

Ah good to know, I haven't seen that be a problem 'yet' (knock wood). 
I'll step away :)

Skyler

On 3/27/2013 9:03 AM, John Murray wrote:
> Yes one problem with some vendors is the loopback address of 127.0.0.1 which
> as you say can be fixed by changing the internal address as described in
> handbook section 5.5.1.
> The other problem some vendors have is with the 'ngcpct=' section of the
> Contact: header as:
>
> Contact: <sip:ngcp-lb at 10.23.11.18:5060;ngcpct='sip:10.23.11.18:5080'>'
>
> Some vendors 'forget' this and reply to sip:ngcp-lb at 10.23.11.18:5060 and
> some get stuck and don't reply at all when they see this.
>
> Regards
>
> John
>
> -----Original Message-----
> From: Skyler [mailto:skchopperguy at gmail.com]
> Sent: 27 March 2013 15:55
> To: Andreas Granig
> Cc: John Murray; spce-user at lists.sipwise.com
> Subject: Re: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
>
> Hi,
>
> On 3/27/2013 8:41 AM, Andreas Granig wrote:
>> Hi,
>>
>> On 03/27/2013 04:11 PM, John Murray wrote:
>>> With a peer you are typically in some sort of relationship and can
>>> negotiate any fixes (not always the case of course). But with a
>>> subscriber you are dependent on the code in the device.
>>> In my experience with PBXs ( naming no names (but NEC is terrible))
>>> are very bad with this header so it would be very helpful if there
>>> were some switch against the subscriber to remove this as a
>>> workaround.
>>> Getting the vendor to do something about this has always proven very
>>> difficult/painful/slow.
>>
>> Well, the reason this has been added was to mask the local sems
>> contact of 127.0.0.1:5080, because some peers are doing far-end nat
>> traversal (like the SPCE itself), and if some peer tries to fix this,
>> the SIP flow is again broken.
>>
>> So in one way or another, this information needs to be coded into the
>> message. I'm open for any better suggestions of course :)
>>
>> Andreas
>>
>
>    I created an lo:0 with a private IP address (10.10.xxx.xxx) for lb >
> b2buas: & proxies > sip:. That may help in this case as its likely the
> 127.0.0.1 which is confusing the upstream proxy.
>
> Skyler
>



From skchopperguy at gmail.com  Wed Mar 27 12:31:33 2013
From: skchopperguy at gmail.com (Skyler)
Date: Wed, 27 Mar 2013 09:31:33 -0700
Subject: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
In-Reply-To: <5153192F.4050106@mail.sipwise.com>
References: <E50575DD48D9B743925AC3BCE6310A4B2436E488@DB3PRD0710MB355.eurprd07.prod.outlook.com>
	<515309C3.1080601@mail.sipwise.com>
	<51530c35.a266b40a.3a4f.ffffb1e3@mx.google.com>
	<5153131A.7060002@mail.sipwise.com> <51531668.2090106@gmail.com>
	<5153192F.4050106@mail.sipwise.com>
Message-ID: <51531EE5.1010802@gmail.com>

That makes sense, also might just be that the ngcpct sip uri is the last 
uri read in the contact.

  Maybe Sylvester and Gary could test this out as they have the OP.

What would a good test look like to base64-encode that ngcpct in a 
customtt.tt2?

Skyler

On 3/27/2013 9:07 AM, Andreas Granig wrote:
> Hi,
>
> On 03/27/2013 04:55 PM, Skyler wrote:
>>   I created an lo:0 with a private IP address (10.10.xxx.xxx) for lb >
>> b2buas: & proxies > sip:. That may help in this case as its likely the
>> 127.0.0.1 which is confusing the upstream proxy.
>
> I've heard of endpoints disliking the SIP uri with the quotes... so one
> option could be to convert the "ngcpct='sip:127.0.0.1:5080'" and
> base64-encode the "sip:127.0.0.1:5080", so it gets something like
> "ngcpct=c2lwOjEyNy4wLjAuMTo1MDgwCg==" instead.
>
> Andreas



From gary at netdiverse.com  Wed Mar 27 12:42:12 2013
From: gary at netdiverse.com (Gary Nieboer)
Date: Wed, 27 Mar 2013 10:42:12 -0600
Subject: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
In-Reply-To: <51531EE5.1010802@gmail.com>
References: <E50575DD48D9B743925AC3BCE6310A4B2436E488@DB3PRD0710MB355.eurprd07.prod.outlook.com>
	<515309C3.1080601@mail.sipwise.com>
	<51530c35.a266b40a.3a4f.ffffb1e3@mx.google.com>
	<5153131A.7060002@mail.sipwise.com> <51531668.2090106@gmail.com>
	<5153192F.4050106@mail.sipwise.com> <51531EE5.1010802@gmail.com>
Message-ID: <CAMXUi2QV2vOAtpSW_CJU73oy=iScpKV0cx+YdShSk6Gi64_pnA@mail.gmail.com>

I will give a couple of the items advised a try.

Should there be more than one uri in the Contact field?  If so, that may be
the quick fix.  The contact field I captured a few weeks ago only has one,
and it is the loopback/proxy address of 127.0.0.1:5080

-Gary
--
Gary Nieboer


877-763-1014 Toll Free  |  801-657-5484 Utah  |  615-800-6946 Tennessee  |
801-694-6106 Mobile
gary at netdiverse.com  |   www.netdiverse.com


On Wed, Mar 27, 2013 at 10:31 AM, Skyler <skchopperguy at gmail.com> wrote:

> That makes sense, also might just be that the ngcpct sip uri is the last
> uri read in the contact.
>
>  Maybe Sylvester and Gary could test this out as they have the OP.
>
> What would a good test look like to base64-encode that ngcpct in a
> customtt.tt2?
>
> Skyler
>
>
> On 3/27/2013 9:07 AM, Andreas Granig wrote:
>
>> Hi,
>>
>> On 03/27/2013 04:55 PM, Skyler wrote:
>>
>>>   I created an lo:0 with a private IP address (10.10.xxx.xxx) for lb >
>>> b2buas: & proxies > sip:. That may help in this case as its likely the
>>> 127.0.0.1 which is confusing the upstream proxy.
>>>
>>
>> I've heard of endpoints disliking the SIP uri with the quotes... so one
>> option could be to convert the "ngcpct='sip:127.0.0.1:5080'" and
>> base64-encode the "sip:127.0.0.1:5080", so it gets something like
>> "ngcpct=**c2lwOjEyNy4wLjAuMTo1MDgwCg==" instead.
>>
>> Andreas
>>
>
> ______________________________**_________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/**listinfo/spce-user<http://lists.sipwise.com/listinfo/spce-user>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130327/8e2bce26/attachment-0001.html>

From skchopperguy at gmail.com  Wed Mar 27 12:48:44 2013
From: skchopperguy at gmail.com (Skyler)
Date: Wed, 27 Mar 2013 09:48:44 -0700
Subject: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
In-Reply-To: <CAMXUi2QV2vOAtpSW_CJU73oy=iScpKV0cx+YdShSk6Gi64_pnA@mail.gmail.com>
References: <E50575DD48D9B743925AC3BCE6310A4B2436E488@DB3PRD0710MB355.eurprd07.prod.outlook.com>
	<515309C3.1080601@mail.sipwise.com>
	<51530c35.a266b40a.3a4f.ffffb1e3@mx.google.com>
	<5153131A.7060002@mail.sipwise.com> <51531668.2090106@gmail.com>
	<5153192F.4050106@mail.sipwise.com> <51531EE5.1010802@gmail.com>
	<CAMXUi2QV2vOAtpSW_CJU73oy=iScpKV0cx+YdShSk6Gi64_pnA@mail.gmail.com>
Message-ID: <515322EC.5080908@gmail.com>

My guess (as I've not actually had this issue) is that your contact 
would look something like:

Contact: <sip:ngcp-lb at your.ext.ip.address:5060;ngcpct='sip:127.0.0.1:5080'>

Skyler

On 3/27/2013 9:42 AM, Gary Nieboer wrote:
> I will give a couple of the items advised a try.
>
> Should there be more than one uri in the Contact field?  If so, that may
> be the quick fix.  The contact field I captured a few weeks ago only has
> one, and it is the loopback/proxy address of 127.0.0.1:5080
> <http://127.0.0.1:5080>
>
> -Gary
> --
> Gary Nieboer
>
>
> 877-763-1014 Toll Free  |  801-657-5484 Utah  |  615-800-6946 Tennessee
> | 801-694-6106 Mobile
> gary at netdiverse.com <mailto:gary at netdiverse.com>  | www.netdiverse.com
> <http://www.netdiverse.com/>
>
>
> On Wed, Mar 27, 2013 at 10:31 AM, Skyler <skchopperguy at gmail.com
> <mailto:skchopperguy at gmail.com>> wrote:
>
>     That makes sense, also might just be that the ngcpct sip uri is the
>     last uri read in the contact.
>
>       Maybe Sylvester and Gary could test this out as they have the OP.
>
>     What would a good test look like to base64-encode that ngcpct in a
>     customtt.tt2?
>
>     Skyler
>
>
>     On 3/27/2013 9:07 AM, Andreas Granig wrote:
>
>         Hi,
>
>         On 03/27/2013 04:55 PM, Skyler wrote:
>
>                I created an lo:0 with a private IP address
>             (10.10.xxx.xxx) for lb >
>             b2buas: & proxies > sip:. That may help in this case as its
>             likely the
>             127.0.0.1 which is confusing the upstream proxy.
>
>
>         I've heard of endpoints disliking the SIP uri with the quotes...
>         so one
>         option could be to convert the "ngcpct='sip:127.0.0.1:5080
>         <http://127.0.0.1:5080>'" and
>         base64-encode the "sip:127.0.0.1:5080 <http://127.0.0.1:5080>",
>         so it gets something like
>         "ngcpct=__c2lwOjEyNy4wLjAuMTo1MDgwCg==" instead.
>
>         Andreas
>
>
>     _________________________________________________
>     Spce-user mailing list
>     Spce-user at lists.sipwise.com <mailto:Spce-user at lists.sipwise.com>
>     http://lists.sipwise.com/__listinfo/spce-user
>     <http://lists.sipwise.com/listinfo/spce-user>
>
>



From gary at netdiverse.com  Wed Mar 27 13:17:23 2013
From: gary at netdiverse.com (Gary Nieboer)
Date: Wed, 27 Mar 2013 11:17:23 -0600
Subject: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
In-Reply-To: <515322EC.5080908@gmail.com>
References: <E50575DD48D9B743925AC3BCE6310A4B2436E488@DB3PRD0710MB355.eurprd07.prod.outlook.com>
	<515309C3.1080601@mail.sipwise.com>
	<51530c35.a266b40a.3a4f.ffffb1e3@mx.google.com>
	<5153131A.7060002@mail.sipwise.com> <51531668.2090106@gmail.com>
	<5153192F.4050106@mail.sipwise.com> <51531EE5.1010802@gmail.com>
	<CAMXUi2QV2vOAtpSW_CJU73oy=iScpKV0cx+YdShSk6Gi64_pnA@mail.gmail.com>
	<515322EC.5080908@gmail.com>
Message-ID: <CAMXUi2QrywYL5wEcXk-A2hDx_4tcDgtAcx9wpdxU1LREVwYFtw@mail.gmail.com>

I found an old pcap with an invite showing the Contact.

That could be the issue.   On the SPCE Invite to the Peer, the only thing
in the capture is:

Contact: <sip:127.0.0.1:5080>


and nothing in the Contact field about our ip:  x.x.x.x:5060 or with
 "ngcp-lb" or "ngcpct"




-Gary
--
Gary Nieboer


877-763-1014 Toll Free  |  801-657-5484 Utah  |  615-800-6946 Tennessee  |
801-694-6106 Mobile
gary at netdiverse.com  |   www.netdiverse.com


On Wed, Mar 27, 2013 at 10:48 AM, Skyler <skchopperguy at gmail.com> wrote:

> My guess (as I've not actually had this issue) is that your contact would
> look something like:
>
> Contact: <sip:ngcp-lb at your.ext.ip.**address:5060;ngcpct='sip:127.**
> 0.0.1:5080 <http://127.0.0.1:5080>'>
>
> Skyler
>
>
> On 3/27/2013 9:42 AM, Gary Nieboer wrote:
>
>> I will give a couple of the items advised a try.
>>
>> Should there be more than one uri in the Contact field?  If so, that may
>> be the quick fix.  The contact field I captured a few weeks ago only has
>> one, and it is the loopback/proxy address of 127.0.0.1:5080
>> <http://127.0.0.1:5080>
>>
>> -Gary
>> --
>> Gary Nieboer
>>
>>
>> 877-763-1014 Toll Free  |  801-657-5484 Utah  |  615-800-6946 Tennessee
>> | 801-694-6106 Mobile
>> gary at netdiverse.com <mailto:gary at netdiverse.com>  | www.netdiverse.com
>> <http://www.netdiverse.com/>
>>
>>
>>
>> On Wed, Mar 27, 2013 at 10:31 AM, Skyler <skchopperguy at gmail.com
>> <mailto:skchopperguy at gmail.com**>> wrote:
>>
>>     That makes sense, also might just be that the ngcpct sip uri is the
>>     last uri read in the contact.
>>
>>       Maybe Sylvester and Gary could test this out as they have the OP.
>>
>>     What would a good test look like to base64-encode that ngcpct in a
>>     customtt.tt2?
>>
>>     Skyler
>>
>>
>>     On 3/27/2013 9:07 AM, Andreas Granig wrote:
>>
>>         Hi,
>>
>>         On 03/27/2013 04:55 PM, Skyler wrote:
>>
>>                I created an lo:0 with a private IP address
>>             (10.10.xxx.xxx) for lb >
>>             b2buas: & proxies > sip:. That may help in this case as its
>>             likely the
>>             127.0.0.1 which is confusing the upstream proxy.
>>
>>
>>         I've heard of endpoints disliking the SIP uri with the quotes...
>>         so one
>>         option could be to convert the "ngcpct='sip:127.0.0.1:5080
>>         <http://127.0.0.1:5080>'" and
>>         base64-encode the "sip:127.0.0.1:5080 <http://127.0.0.1:5080>",
>>
>>         so it gets something like
>>         "ngcpct=__**c2lwOjEyNy4wLjAuMTo1MDgwCg==" instead.
>>
>>         Andreas
>>
>>
>>     ______________________________**___________________
>>     Spce-user mailing list
>>     Spce-user at lists.sipwise.com <mailto:Spce-user at lists.**sipwise.com<Spce-user at lists.sipwise.com>
>> >
>>     http://lists.sipwise.com/__**listinfo/spce-user<http://lists.sipwise.com/__listinfo/spce-user>
>>     <http://lists.sipwise.com/**listinfo/spce-user<http://lists.sipwise.com/listinfo/spce-user>
>> >
>>
>>
>>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130327/29e262db/attachment-0001.html>

From m.rys at tri-lakes.net  Wed Mar 27 13:30:08 2013
From: m.rys at tri-lakes.net (Marc Rys)
Date: Wed, 27 Mar 2013 12:30:08 -0500 (CDT)
Subject: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
In-Reply-To: <515322EC.5080908@gmail.com>
Message-ID: <c44cce95-50a6-41bf-9ca3-931723259972@usbsnpzimbra01v.servers.rystec.com>

Ok, Im not sure if this is related, but I've been having struggles with our Taqua Gateway peering to SPCE.  OUr Gateway kept replying back with a 403 Forbidden whenever SPCE tried to send a call to it.

After seeing your dialog about the contact headers, i looked at one our wireshark captures and our Invites were showing the same thing...
sip:ngcp-lb at your.ext.ip.address:5060;ngcpct='sip:127.0.0.1:5080'

I followed the instructions for the audiocodes fix and changed our dummy address to 10.10.10.1  On our Taqua Gateway I had to add the dummy IP to the IP control list.. (10.10.10.1:5080) and now the Taqua accepts calls from SPCE.

I'm not sure if this is solution for you to use with the other providers listed in the thread.

In my attached wireshark cap, it appears that SPCE is sending an ACK back from my dummy address 10.10.10.1 to my gateway.  Thats not right is it?  at this point now, my calls to our gateway dont stay up for longer than 32Seconds then the audio drops out of the call and the person on the PSTN side of the call is disconnected completely, and my SPCE UA appears to have the call up still with no audio until I end the call.



Marc Rys
http://www.tri-lakes.net
http://www.rystec.com

----- Original Message -----
From: "Skyler" <skchopperguy at gmail.com>
To: "Gary Nieboer" <gary at netdiverse.com>
Cc: spce-user at lists.sipwise.com
Sent: Wednesday, March 27, 2013 11:48:44 AM
Subject: Re: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header

My guess (as I've not actually had this issue) is that your contact 
would look something like:

Contact: <sip:ngcp-lb at your.ext.ip.address:5060;ngcpct='sip:127.0.0.1:5080'>

Skyler

On 3/27/2013 9:42 AM, Gary Nieboer wrote:
> I will give a couple of the items advised a try.
>
> Should there be more than one uri in the Contact field?  If so, that may
> be the quick fix.  The contact field I captured a few weeks ago only has
> one, and it is the loopback/proxy address of 127.0.0.1:5080
> <http://127.0.0.1:5080>
>
> -Gary
> --
> Gary Nieboer
>
>
> 877-763-1014 Toll Free  |  801-657-5484 Utah  |  615-800-6946 Tennessee
> | 801-694-6106 Mobile
> gary at netdiverse.com <mailto:gary at netdiverse.com>  | www.netdiverse.com
> <http://www.netdiverse.com/>
>
>
> On Wed, Mar 27, 2013 at 10:31 AM, Skyler <skchopperguy at gmail.com
> <mailto:skchopperguy at gmail.com>> wrote:
>
>     That makes sense, also might just be that the ngcpct sip uri is the
>     last uri read in the contact.
>
>       Maybe Sylvester and Gary could test this out as they have the OP.
>
>     What would a good test look like to base64-encode that ngcpct in a
>     customtt.tt2?
>
>     Skyler
>
>
>     On 3/27/2013 9:07 AM, Andreas Granig wrote:
>
>         Hi,
>
>         On 03/27/2013 04:55 PM, Skyler wrote:
>
>                I created an lo:0 with a private IP address
>             (10.10.xxx.xxx) for lb >
>             b2buas: & proxies > sip:. That may help in this case as its
>             likely the
>             127.0.0.1 which is confusing the upstream proxy.
>
>
>         I've heard of endpoints disliking the SIP uri with the quotes...
>         so one
>         option could be to convert the "ngcpct='sip:127.0.0.1:5080
>         <http://127.0.0.1:5080>'" and
>         base64-encode the "sip:127.0.0.1:5080 <http://127.0.0.1:5080>",
>         so it gets something like
>         "ngcpct=__c2lwOjEyNy4wLjAuMTo1MDgwCg==" instead.
>
>         Andreas
>
>
>     _________________________________________________
>     Spce-user mailing list
>     Spce-user at lists.sipwise.com <mailto:Spce-user at lists.sipwise.com>
>     http://lists.sipwise.com/__listinfo/spce-user
>     <http://lists.sipwise.com/listinfo/spce-user>
>
>

_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com
http://lists.sipwise.com/listinfo/spce-user
-------------- next part --------------
A non-text attachment was scrubbed...
Name: SPCE-OUTBOUND-Taqua-30SecDisc.pcap
Type: application/octet-stream
Size: 1654240 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130327/d0b68338/attachment-0001.pcap>

From skchopperguy at gmail.com  Wed Mar 27 15:53:13 2013
From: skchopperguy at gmail.com (Skyler)
Date: Wed, 27 Mar 2013 12:53:13 -0700
Subject: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
In-Reply-To: <CAMXUi2QrywYL5wEcXk-A2hDx_4tcDgtAcx9wpdxU1LREVwYFtw@mail.gmail.com>
References: <E50575DD48D9B743925AC3BCE6310A4B2436E488@DB3PRD0710MB355.eurprd07.prod.outlook.com>
	<515309C3.1080601@mail.sipwise.com>
	<51530c35.a266b40a.3a4f.ffffb1e3@mx.google.com>
	<5153131A.7060002@mail.sipwise.com> <51531668.2090106@gmail.com>
	<5153192F.4050106@mail.sipwise.com> <51531EE5.1010802@gmail.com>
	<CAMXUi2QV2vOAtpSW_CJU73oy=iScpKV0cx+YdShSk6Gi64_pnA@mail.gmail.com>
	<515322EC.5080908@gmail.com>
	<CAMXUi2QrywYL5wEcXk-A2hDx_4tcDgtAcx9wpdxU1LREVwYFtw@mail.gmail.com>
Message-ID: <51534E29.6090006@gmail.com>

I would say that the invite you are looking at was internal between the 
proxy and lb. its easy to mix them. Would be impossible for any outside 
peer to contact localhost so SPCE would not insert 127.0.0.1 as a 
contact to an outside peer. Are you sure the invite was from your 
external IP to the external peer IP?

Skyler

On 3/27/2013 10:17 AM, Gary Nieboer wrote:
> I found an old pcap with an invite showing the Contact.
>
> That could be the issue.   On the SPCE Invite to the Peer, the only
> thing in the capture is:
>
>     Contact: <sip:127.0.0.1:5080 <http://127.0.0.1:5080>>
>
>
> and nothing in the Contact field about our ip:  x.x.x.x:5060 or with
>   "ngcp-lb" or "ngcpct"
>
>
>
>
> -Gary
> --
> Gary Nieboer
>
>
> 877-763-1014 Toll Free  |  801-657-5484 Utah  |  615-800-6946 Tennessee
> | 801-694-6106 Mobile
> gary at netdiverse.com <mailto:gary at netdiverse.com>  | www.netdiverse.com
> <http://www.netdiverse.com/>
>
>
> On Wed, Mar 27, 2013 at 10:48 AM, Skyler <skchopperguy at gmail.com
> <mailto:skchopperguy at gmail.com>> wrote:
>
>     My guess (as I've not actually had this issue) is that your contact
>     would look something like:
>
>     Contact:
>     <sip:ngcp-lb at your.ext.ip.__address:5060;ngcpct='sip:127.__0.0.1:5080
>     <http://127.0.0.1:5080>'>
>
>     Skyler
>
>
>     On 3/27/2013 9:42 AM, Gary Nieboer wrote:
>
>         I will give a couple of the items advised a try.
>
>         Should there be more than one uri in the Contact field?  If so,
>         that may
>         be the quick fix.  The contact field I captured a few weeks ago
>         only has
>         one, and it is the loopback/proxy address of 127.0.0.1:5080
>         <http://127.0.0.1:5080>
>         <http://127.0.0.1:5080>
>
>         -Gary
>         --
>         Gary Nieboer
>
>
>         877-763-1014 <tel:877-763-1014> Toll Free  | 801-657-5484
>         <tel:801-657-5484> Utah  | 615-800-6946 <tel:615-800-6946> Tennessee
>         | 801-694-6106 <tel:801-694-6106> Mobile
>         gary at netdiverse.com <mailto:gary at netdiverse.com>
>         <mailto:gary at netdiverse.com <mailto:gary at netdiverse.com>>  |
>         www.netdiverse.com <http://www.netdiverse.com>
>         <http://www.netdiverse.com/>
>
>
>
>         On Wed, Mar 27, 2013 at 10:31 AM, Skyler <skchopperguy at gmail.com
>         <mailto:skchopperguy at gmail.com>
>         <mailto:skchopperguy at gmail.com
>         <mailto:skchopperguy at gmail.com>__>> wrote:
>
>              That makes sense, also might just be that the ngcpct sip
>         uri is the
>              last uri read in the contact.
>
>                Maybe Sylvester and Gary could test this out as they have
>         the OP.
>
>              What would a good test look like to base64-encode that
>         ngcpct in a
>              customtt.tt2?
>
>              Skyler
>
>
>              On 3/27/2013 9:07 AM, Andreas Granig wrote:
>
>                  Hi,
>
>                  On 03/27/2013 04:55 PM, Skyler wrote:
>
>                         I created an lo:0 with a private IP address
>                      (10.10.xxx.xxx) for lb >
>                      b2buas: & proxies > sip:. That may help in this
>         case as its
>                      likely the
>                      127.0.0.1 which is confusing the upstream proxy.
>
>
>                  I've heard of endpoints disliking the SIP uri with the
>         quotes...
>                  so one
>                  option could be to convert the
>         "ngcpct='sip:127.0.0.1:5080 <http://127.0.0.1:5080>
>                  <http://127.0.0.1:5080>'" and
>                  base64-encode the "sip:127.0.0.1:5080
>         <http://127.0.0.1:5080> <http://127.0.0.1:5080>",
>
>                  so it gets something like
>                  "ngcpct=____c2lwOjEyNy4wLjAuMTo1MDgwCg==" instead.
>
>                  Andreas
>
>
>              ___________________________________________________
>              Spce-user mailing list
>         Spce-user at lists.sipwise.com <mailto:Spce-user at lists.sipwise.com>
>         <mailto:Spce-user at lists.__sipwise.com
>         <mailto:Spce-user at lists.sipwise.com>>
>         http://lists.sipwise.com/____listinfo/spce-user
>         <http://lists.sipwise.com/__listinfo/spce-user>
>              <http://lists.sipwise.com/__listinfo/spce-user
>         <http://lists.sipwise.com/listinfo/spce-user>>
>
>
>



From skchopperguy at gmail.com  Wed Mar 27 16:05:30 2013
From: skchopperguy at gmail.com (Skyler)
Date: Wed, 27 Mar 2013 13:05:30 -0700
Subject: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
In-Reply-To: <c44cce95-50a6-41bf-9ca3-931723259972@usbsnpzimbra01v.servers.rystec.com>
References: <c44cce95-50a6-41bf-9ca3-931723259972@usbsnpzimbra01v.servers.rystec.com>
Message-ID: <5153510A.8000400@gmail.com>

I would run..

ngrep-sip b -d any -qt -W byline port 5060 >/tmp/ngcpct_check.txt

and make a test call. Then attach the txt to the list.

Skyler

On 3/27/2013 10:30 AM, Marc Rys wrote:
> Ok, Im not sure if this is related, but I've been having struggles with our Taqua Gateway peering to SPCE.  OUr Gateway kept replying back with a 403 Forbidden whenever SPCE tried to send a call to it.
>
> After seeing your dialog about the contact headers, i looked at one our wireshark captures and our Invites were showing the same thing...
> sip:ngcp-lb at your.ext.ip.address:5060;ngcpct='sip:127.0.0.1:5080'
>
> I followed the instructions for the audiocodes fix and changed our dummy address to 10.10.10.1  On our Taqua Gateway I had to add the dummy IP to the IP control list.. (10.10.10.1:5080) and now the Taqua accepts calls from SPCE.
>
> I'm not sure if this is solution for you to use with the other providers listed in the thread.
>
> In my attached wireshark cap, it appears that SPCE is sending an ACK back from my dummy address 10.10.10.1 to my gateway.  Thats not right is it?  at this point now, my calls to our gateway dont stay up for longer than 32Seconds then the audio drops out of the call and the person on the PSTN side of the call is disconnected completely, and my SPCE UA appears to have the call up still with no audio until I end the call.
>
>
>
> Marc Rys
> http://www.tri-lakes.net
> http://www.rystec.com
>
> ----- Original Message -----
> From: "Skyler" <skchopperguy at gmail.com>
> To: "Gary Nieboer" <gary at netdiverse.com>
> Cc: spce-user at lists.sipwise.com
> Sent: Wednesday, March 27, 2013 11:48:44 AM
> Subject: Re: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
>
> My guess (as I've not actually had this issue) is that your contact
> would look something like:
>
> Contact: <sip:ngcp-lb at your.ext.ip.address:5060;ngcpct='sip:127.0.0.1:5080'>
>
> Skyler
>
> On 3/27/2013 9:42 AM, Gary Nieboer wrote:
>> I will give a couple of the items advised a try.
>>
>> Should there be more than one uri in the Contact field?  If so, that may
>> be the quick fix.  The contact field I captured a few weeks ago only has
>> one, and it is the loopback/proxy address of 127.0.0.1:5080
>> <http://127.0.0.1:5080>
>>
>> -Gary
>> --
>> Gary Nieboer
>>
>>
>> 877-763-1014 Toll Free  |  801-657-5484 Utah  |  615-800-6946 Tennessee
>> | 801-694-6106 Mobile
>> gary at netdiverse.com <mailto:gary at netdiverse.com>  | www.netdiverse.com
>> <http://www.netdiverse.com/>
>>
>>
>> On Wed, Mar 27, 2013 at 10:31 AM, Skyler <skchopperguy at gmail.com
>> <mailto:skchopperguy at gmail.com>> wrote:
>>
>>      That makes sense, also might just be that the ngcpct sip uri is the
>>      last uri read in the contact.
>>
>>        Maybe Sylvester and Gary could test this out as they have the OP.
>>
>>      What would a good test look like to base64-encode that ngcpct in a
>>      customtt.tt2?
>>
>>      Skyler
>>
>>
>>      On 3/27/2013 9:07 AM, Andreas Granig wrote:
>>
>>          Hi,
>>
>>          On 03/27/2013 04:55 PM, Skyler wrote:
>>
>>                 I created an lo:0 with a private IP address
>>              (10.10.xxx.xxx) for lb >
>>              b2buas: & proxies > sip:. That may help in this case as its
>>              likely the
>>              127.0.0.1 which is confusing the upstream proxy.
>>
>>
>>          I've heard of endpoints disliking the SIP uri with the quotes...
>>          so one
>>          option could be to convert the "ngcpct='sip:127.0.0.1:5080
>>          <http://127.0.0.1:5080>'" and
>>          base64-encode the "sip:127.0.0.1:5080 <http://127.0.0.1:5080>",
>>          so it gets something like
>>          "ngcpct=__c2lwOjEyNy4wLjAuMTo1MDgwCg==" instead.
>>
>>          Andreas
>>
>>
>>      _________________________________________________
>>      Spce-user mailing list
>>      Spce-user at lists.sipwise.com <mailto:Spce-user at lists.sipwise.com>
>>      http://lists.sipwise.com/__listinfo/spce-user
>>      <http://lists.sipwise.com/listinfo/spce-user>
>>
>>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
>
>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
>



From jchism2 at gmail.com  Wed Mar 27 17:39:41 2013
From: jchism2 at gmail.com (Jeremie Chism)
Date: Wed, 27 Mar 2013 16:39:41 -0500
Subject: [Spce-user] Bug report for 2.7
Message-ID: <82D1110D-5433-4AB7-A974-9CB8B97E2692@gmail.com>

We have a subscriber that provides a sip trunk to a freepbx instance. We have around 12-14 dids pointed at this instance through alias. When you make any change to the alias the cli which was cleared out before to allow different dids to show populates back with the e164 number. I do not remember this happening on 2.6. Is there a way to correct this. 

Sent from my iPhone


From ljallow at utg.edu.gm  Thu Mar 28 03:24:36 2013
From: ljallow at utg.edu.gm (lamarana jallow)
Date: Thu, 28 Mar 2013 15:24:36 +0800
Subject: [Spce-user]  Packet Routing by SIP server
Message-ID: <CADRbaC6+N+zkruw6YcO29k8USdNbtNvxkU50XDBhZa+myRGsxA@mail.gmail.com>

hello guys,
first of all sorry for the title, i dint know the right title to give it.

i read some papers about SIP but i have a question which i cant seem to
find the answer to,
and i am hoping the gurus in here will help me with it, or direct me to a
resource.

i am actually concerned about the individual packets that is sent from one
UA to another.
Lets say  after the connection has been established by the SIP provider,
and the ACK has been
sent to the callee and the media is now between the two UA's   what happens
to the packets, do
they go via the SIP server or not? So in this case when the call has been
established  and the server is down
does it mean that the call will drop?

Thank you for the usual help.





-- 

-- 
Lamarana Jallow
Graduate Assistant
University of The Gambia
Administration Building
Brikama Campus
P.O. Box 3530
Mob:(+220)7019072
The Gambia. West Africa
E-mail:ljallow at utg.edu.gm <http://goog_1591787375/>
/lamjals<http://goog_1591787375/>
@ <http://goog_1591787375/>yahoo.com
Web: www.utg.edu.gm
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130328/c5ab7422/attachment-0001.html>

From jbonilla at sipwise.com  Thu Mar 28 04:37:27 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Thu, 28 Mar 2013 09:37:27 +0100
Subject: [Spce-user] Packet Routing by SIP server
In-Reply-To: <CADRbaC6+N+zkruw6YcO29k8USdNbtNvxkU50XDBhZa+myRGsxA@mail.gmail.com>
References: <CADRbaC6+N+zkruw6YcO29k8USdNbtNvxkU50XDBhZa+myRGsxA@mail.gmail.com>
Message-ID: <20130328093727.27e82ee5@quenya>

El Thu, 28 Mar 2013 15:24:36 +0800
lamarana jallow <ljallow at utg.edu.gm> escribi?:

> hello guys,
> first of all sorry for the title, i dint know the right title to give it.
> 
> i read some papers about SIP but i have a question which i cant seem to
> find the answer to,
> and i am hoping the gurus in here will help me with it, or direct me to a
> resource.
> 
> i am actually concerned about the individual packets that is sent from one
> UA to another.
> Lets say  after the connection has been established by the SIP provider,
> and the ACK has been
> sent to the callee and the media is now between the two UA's   what happens
> to the packets, do
> they go via the SIP server or not?

Yes, they do. You need them lto keep track of the session and close the
accounting when the call is finished.

> So in this case when the call has been
> established  and the server is down
> does it mean that the call will drop?
> 

Yes, it will.


> Thank you for the usual help.
> 

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130328/eb35a380/attachment-0001.asc>

From mpower at entelit.ca  Thu Mar 28 14:25:19 2013
From: mpower at entelit.ca (Maxwell Power)
Date: Thu, 28 Mar 2013 12:25:19 -0600
Subject: [Spce-user] Passing dialed number to subscriber
Message-ID: <005301ce2be1$9abc49f0$d034ddd0$@entelit.ca>

Hello,

 

I have been trying to wrap my head around how to accomplish this for a
little while now. 

 

A subscriber has a their own PBX. We have multiple number aliases setup for
a subscriber. How do I pass the dialled number to the subscribers PBX so it
can route the call appropriately? It seems no matter what I change and mess
around with the dialled number does not show up in any of the headers.

 

I have messed around in the gui but cannot find an easy way to accomplish
this. Am I missing something? Do I have to set each number up as a
subscriber separately?

Cheers,

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130328/95d9b18e/attachment-0001.html>

From jbonilla at sipwise.com  Thu Mar 28 14:45:31 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Thu, 28 Mar 2013 19:45:31 +0100
Subject: [Spce-user] Passing dialed number to subscriber
In-Reply-To: <005301ce2be1$9abc49f0$d034ddd0$@entelit.ca>
References: <005301ce2be1$9abc49f0$d034ddd0$@entelit.ca>
Message-ID: <20130328194531.5cbfc35c@quenya>

El Thu, 28 Mar 2013 12:25:19 -0600
"Maxwell Power" <mpower at entelit.ca> escribi?:

> Hello,
> 
>  
> 
> I have been trying to wrap my head around how to accomplish this for a
> little while now. 
> 
>  
> 
> A subscriber has a their own PBX. We have multiple number aliases setup for
> a subscriber. How do I pass the dialled number to the subscribers PBX so it
> can route the call appropriately? It seems no matter what I change and mess
> around with the dialled number does not show up in any of the headers.
> 
>  
> 
> I have messed around in the gui but cannot find an easy way to accomplish
> this. Am I missing something? Do I have to set each number up as a
> subscriber separately?
> 
> Cheers,
> 

You have an option for that: "e164_to_ruri"

You can enable it at subscriber level and this is the recommended way of
dealing with multialiased pbx systems. With this option, the system will send
the alias number (normalized via rewrite rules of course) and it will fallback
to the sip username in case a 404 is returned by the subscriber.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130328/86438eb0/attachment-0001.asc>

From sn at fest.dk  Thu Mar 28 14:46:45 2013
From: sn at fest.dk (Sylvester Nielsen)
Date: Thu, 28 Mar 2013 18:46:45 +0000
Subject: [Spce-user] Passing dialed number to subscriber
In-Reply-To: <005301ce2be1$9abc49f0$d034ddd0$@entelit.ca>
References: <005301ce2be1$9abc49f0$d034ddd0$@entelit.ca>
Message-ID: <E50575DD48D9B743925AC3BCE6310A4B2436F2BD@DB3PRD0710MB355.eurprd07.prod.outlook.com>

Hi,

You need to check the e164_to_ruri for that subscriber.
By doing this the invite to the PBX will contain the DID number for you to route the inbound calls.

// Sylvester

Fra: spce-user-bounces at lists.sipwise.com [mailto:spce-user-bounces at lists.sipwise.com] P? vegne af Maxwell Power
Sendt: 28. marts 2013 19:25
Til: spce-user at lists.sipwise.com
Emne: [Spce-user] Passing dialed number to subscriber

Hello,

I have been trying to wrap my head around how to accomplish this for a little while now.

A subscriber has a their own PBX. We have multiple number aliases setup for a subscriber. How do I pass the dialled number to the subscribers PBX so it can route the call appropriately? It seems no matter what I change and mess around with the dialled number does not show up in any of the headers.

I have messed around in the gui but cannot find an easy way to accomplish this. Am I missing something? Do I have to set each number up as a subscriber separately?
Cheers,

________________________________
Denne mail er blevet scannet af fest.dk ApS<http://www.fest.dk/> og indeholder ikke virus!
________________________________
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130328/b18fed7b/attachment-0001.html>

From mpower at entelit.ca  Thu Mar 28 14:59:24 2013
From: mpower at entelit.ca (Maxwell Power)
Date: Thu, 28 Mar 2013 12:59:24 -0600
Subject: [Spce-user] Passing dialed number to subscriber
In-Reply-To: <E50575DD48D9B743925AC3BCE6310A4B2436F2BD@DB3PRD0710MB355.eurprd07.prod.outlook.com>
References: <005301ce2be1$9abc49f0$d034ddd0$@entelit.ca>
	<E50575DD48D9B743925AC3BCE6310A4B2436F2BD@DB3PRD0710MB355.eurprd07.prod.outlook.com>
Message-ID: <007201ce2be6$5e1574e0$1a405ea0$@entelit.ca>

Thanks Jon and Sylvester. I knew it was something really simple I was
overlooking. Worked like a charm!

Cheers,



From: Sylvester Nielsen [mailto:sn at fest.dk] 
Sent: March-28-13 12:47
To: Maxwell Power; spce-user at lists.sipwise.com
Subject: SV: [Spce-user] Passing dialed number to subscriber

 

Hi,

 

You need to check the e164_to_ruri for that subscriber.

By doing this the invite to the PBX will contain the DID number for you to
route the inbound calls.

 

// Sylvester

 

Fra: spce-user-bounces at lists.sipwise.com
[mailto:spce-user-bounces at lists.sipwise.com] P? vegne af Maxwell Power
Sendt: 28. marts 2013 19:25
Til: spce-user at lists.sipwise.com
Emne: [Spce-user] Passing dialed number to subscriber

 

Hello,

 

I have been trying to wrap my head around how to accomplish this for a
little while now. 

 

A subscriber has a their own PBX. We have multiple number aliases setup for
a subscriber. How do I pass the dialled number to the subscribers PBX so it
can route the call appropriately? It seems no matter what I change and mess
around with the dialled number does not show up in any of the headers.

 

I have messed around in the gui but cannot find an easy way to accomplish
this. Am I missing something? Do I have to set each number up as a
subscriber separately?

Cheers,

 

  _____  

Denne mail er blevet scannet af  <http://www.fest.dk/> fest.dk ApS og
indeholder ikke virus! 

  _____  

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130328/a6f2eb0c/attachment-0001.html>

From agranig at sipwise.com  Thu Mar 28 15:39:15 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Thu, 28 Mar 2013 20:39:15 +0100
Subject: [Spce-user] Packet Routing by SIP server
In-Reply-To: <CADRbaC6+N+zkruw6YcO29k8USdNbtNvxkU50XDBhZa+myRGsxA@mail.gmail.com>
References: <CADRbaC6+N+zkruw6YcO29k8USdNbtNvxkU50XDBhZa+myRGsxA@mail.gmail.com>
Message-ID: <51549C63.2000308@mail.sipwise.com>

Hi,

On 03/28/2013 08:24 AM, lamarana jallow wrote:
> i am actually concerned about the individual packets that is sent from
> one UA to another.
> Lets say  after the connection has been established by the SIP provider,
> and the ACK has been
> sent to the callee and the media is now between the two UA's   what
> happens to the packets, do
> they go via the SIP server or not? So in this case when the call has
> been established  and the server is down
> does it mean that the call will drop?

Well, depends :)

If you mean the media streams, then on the SPCE there are user 
preferences "always_use_rtpproxy" and "never_use_rtpproxy" for 
subscriber/domain/peer to control whether the media relay on the SPCE 
will be engaged or not. If not, your media goes end-to-end (if you're 
not behind nat), so it won't matter if the SPCE goes down for a short 
time during a call.

However, UAs also (can) do session keep-alives using session timers on 
SIP level during a call, and if the SPCE is down, the UAs will end the 
call if they are not successful.

And finally, if you want to rely on correct billing, you want your SPCE 
to be always up to write an accounting record for the BYE, otherwise you 
won't get a CDR from these calls and can't bill your users.

Andreas



From agranig at sipwise.com  Fri Mar 29 11:36:27 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Fri, 29 Mar 2013 16:36:27 +0100
Subject: [Spce-user] sip:provider CE v2.8 released
Message-ID: <5155B4FB.5090700@mail.sipwise.com>

Hi guys,

I'm excited to announce the new SPCE v2.8. Please check the official 
announcement at

	http://www.sipwise.com/news/announcements/spce-v2_8-release/

for a list of changes.

This version will be our first LTS version, so we will keep back-porting 
fixes and security updates to this version for the next three years. If 
you want something rock-stable for the years to come, you should go with 
that!

After v2.8 we'll switch to v3.0 (due in ~3 months as always) with a 
completely reworked web panel, and we'll switch to a REST interface for 
3rd party provisioning clients, and no more new features will be exposed 
via the SOAP and XMLRPC (and we'll phase it out completely by end of 
this year).

I'd like to thank you all for your valuable feedback and discussions, 
let's keep it going for the next v3.0 with some awesome changes ahead!

Thank you,
Andreas

PS: Please let us know if there goes something wrong in your upgrade or 
installation. It has been tested extensively via automated jobs and also 
manually a lot of times, but if you find something nonetheless, we'll be 
happy to fix it.



From skchopperguy at gmail.com  Fri Mar 29 19:45:24 2013
From: skchopperguy at gmail.com (Skyler)
Date: Fri, 29 Mar 2013 16:45:24 -0700
Subject: [Spce-user] sip:provider CE v2.8 released
In-Reply-To: <5155B4FB.5090700@mail.sipwise.com>
References: <5155B4FB.5090700@mail.sipwise.com>
Message-ID: <51562794.3070006@gmail.com>

Hi,

  Thank you for the LTS and many great updates this release!

  Wondering if Kamailio 4.0 is slated for SPCE 3.0? Will it also be 
back-ported for 2.8-LTS at that time too?

  Very excited on the move to REST interface in provisioning!

Skyler

On 3/29/2013 8:36 AM, Andreas Granig wrote:
> Hi guys,
>
> I'm excited to announce the new SPCE v2.8. Please check the official
> announcement at
>
>      http://www.sipwise.com/news/announcements/spce-v2_8-release/
>
> for a list of changes.
>
> This version will be our first LTS version, so we will keep back-porting
> fixes and security updates to this version for the next three years. If
> you want something rock-stable for the years to come, you should go with
> that!
>
> After v2.8 we'll switch to v3.0 (due in ~3 months as always) with a
> completely reworked web panel, and we'll switch to a REST interface for
> 3rd party provisioning clients, and no more new features will be exposed
> via the SOAP and XMLRPC (and we'll phase it out completely by end of
> this year).
>
> I'd like to thank you all for your valuable feedback and discussions,
> let's keep it going for the next v3.0 with some awesome changes ahead!
>
> Thank you,
> Andreas
>
> PS: Please let us know if there goes something wrong in your upgrade or
> installation. It has been tested extensively via automated jobs and also
> manually a lot of times, but if you find something nonetheless, we'll be
> happy to fix it.
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user



From sipwise at ics-il.net  Sat Mar 30 11:47:59 2013
From: sipwise at ics-il.net (sipwise at ics-il.net)
Date: Sat, 30 Mar 2013 10:47:59 -0500 (CDT)
Subject: [Spce-user] Intro
In-Reply-To: <18585488.5264.1364657228791.JavaMail.mhammett@ThunderFuck>
Message-ID: <26490533.5272.1364658487949.JavaMail.mhammett@ThunderFuck>

I just thought I'd introduce myself to the list. I am looking to migrate to the SIPWise platform. I'll mostly lurk, but I'll surely ask some questions before I move over. 



----- 
Mike Hammett 
Intelligent Computing Solutions 
http://www.ics-il.com 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130330/7642b32b/attachment-0001.html>

From jbonilla at sipwise.com  Sat Mar 30 15:26:23 2013
From: jbonilla at sipwise.com (Jon Bonilla (Manwe))
Date: Sat, 30 Mar 2013 20:26:23 +0100
Subject: [Spce-user] Intro
In-Reply-To: <26490533.5272.1364658487949.JavaMail.mhammett@ThunderFuck>
References: <18585488.5264.1364657228791.JavaMail.mhammett@ThunderFuck>
	<26490533.5272.1364658487949.JavaMail.mhammett@ThunderFuck>
Message-ID: <20130330202623.2374e1e1@quenya>

El Sat, 30 Mar 2013 10:47:59 -0500 (CDT)
sipwise at ics-il.net escribi?:

> I just thought I'd introduce myself to the list. I am looking to migrate to
> the SIPWise platform. I'll mostly lurk, but I'll surely ask some questions
> before I move over. 
> 
> 


Welcome to the list!

Happy to see that the family is growing :)

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 836 bytes
Desc: not available
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130330/be7bc1ab/attachment-0001.asc>

From agranig at sipwise.com  Sat Mar 30 15:28:26 2013
From: agranig at sipwise.com (Andreas Granig)
Date: Sat, 30 Mar 2013 20:28:26 +0100
Subject: [Spce-user] Intro
In-Reply-To: <26490533.5272.1364658487949.JavaMail.mhammett@ThunderFuck>
References: <26490533.5272.1364658487949.JavaMail.mhammett@ThunderFuck>
Message-ID: <51573CDA.8060604@sipwise.com>

Hi Mike, welcome!

On 03/30/2013 04:47 PM, sipwise at ics-il.net wrote:
> I just thought I'd introduce myself to the list. I am looking to migrate
> to the SIPWise platform. I'll mostly lurk, but I'll surely ask some
> questions before I move over.

Nice! If anything is unclear from the handbook, ask here. Also, if you 
find there is something missing in the handbook, let us know. We're 
trying to constantly improve it, but if you work with it on a day-to-day 
basis, lots of things become obvious... for beginners though, not so much.

We're also welcoming contributions to the handbook, just post new 
sections/chapters to the spce-dev list. We're also working on moving it 
to github, so it gets easy to send pull requests.

Andreas



From sipwise at ics-il.net  Sat Mar 30 15:35:25 2013
From: sipwise at ics-il.net (sipwise at ics-il.net)
Date: Sat, 30 Mar 2013 14:35:25 -0500 (CDT)
Subject: [Spce-user] Provisioning
Message-ID: <8414863.5386.1364672121268.JavaMail.mhammett@ThunderFuck>

The only references I saw in Google about provisioning in SIPWise were to the raw information the end-user could provision their device with. 

Is there any provisioning system built-in? 

If not, what are people using for this? 

What I'm talking about is the end-user ATA fetches a config from the server that is propagated with username\password, server address, etc. 




----- 
Mike Hammett 
Intelligent Computing Solutions 
http://www.ics-il.com 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130330/a1783dc4/attachment-0001.html>

From jchism2 at gmail.com  Sat Mar 30 15:58:24 2013
From: jchism2 at gmail.com (Jeremie Chism)
Date: Sat, 30 Mar 2013 14:58:24 -0500
Subject: [Spce-user] Provisioning
In-Reply-To: <8414863.5386.1364672121268.JavaMail.mhammett@ThunderFuck>
References: <8414863.5386.1364672121268.JavaMail.mhammett@ThunderFuck>
Message-ID: <3B6E8EC8-9CC7-4BD8-83B1-B9BD44442432@gmail.com>

Ata provisioning is not built into sipwise. You would need to develop your own system. We are looking at several options but have not settled on one. 

Sent from my iPhone

On Mar 30, 2013, at 2:35 PM, sipwise at ics-il.net wrote:

> The only references I saw in Google about provisioning in SIPWise were to the raw information the end-user could provision their device with.
> 
> Is there any provisioning system built-in?
> 
> If not, what are people using for this?
> 
> What I'm talking about is the end-user ATA fetches a config from the server that is propagated with username\password, server address, etc.
> 
> 
> 
> -----
> Mike Hammett
> Intelligent Computing Solutions
> http://www.ics-il.com
> 
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20130330/b6b3a08d/attachment-0001.html>

From m.rys at tri-lakes.net  Sat Mar 30 17:22:00 2013
From: m.rys at tri-lakes.net (Marc Rys)
Date: Sat, 30 Mar 2013 16:22:00 -0500 (CDT)
Subject: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
In-Reply-To: <5153510A.8000400@gmail.com>
Message-ID: <3eb26bda-cede-43bc-a8f0-20c907977c71@usbsnpzimbra01v.servers.rystec.com>

Sorry for the delay.  Attached is the file as requested.

Marc

----- Original Message -----
From: "Skyler" <skchopperguy at gmail.com>
To: spce-user at lists.sipwise.com
Sent: Wednesday, March 27, 2013 3:05:30 PM
Subject: Re: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header

I would run..

ngrep-sip b -d any -qt -W byline port 5060 >/tmp/ngcpct_check.txt

and make a test call. Then attach the txt to the list.

Skyler

On 3/27/2013 10:30 AM, Marc Rys wrote:
> Ok, Im not sure if this is related, but I've been having struggles with our Taqua Gateway peering to SPCE.  OUr Gateway kept replying back with a 403 Forbidden whenever SPCE tried to send a call to it.
>
> After seeing your dialog about the contact headers, i looked at one our wireshark captures and our Invites were showing the same thing...
> sip:ngcp-lb at your.ext.ip.address:5060;ngcpct='sip:127.0.0.1:5080'
>
> I followed the instructions for the audiocodes fix and changed our dummy address to 10.10.10.1  On our Taqua Gateway I had to add the dummy IP to the IP control list.. (10.10.10.1:5080) and now the Taqua accepts calls from SPCE.
>
> I'm not sure if this is solution for you to use with the other providers listed in the thread.
>
> In my attached wireshark cap, it appears that SPCE is sending an ACK back from my dummy address 10.10.10.1 to my gateway.  Thats not right is it?  at this point now, my calls to our gateway dont stay up for longer than 32Seconds then the audio drops out of the call and the person on the PSTN side of the call is disconnected completely, and my SPCE UA appears to have the call up still with no audio until I end the call.
>
>
>
> Marc Rys
> http://www.tri-lakes.net
> http://www.rystec.com
>
> ----- Original Message -----
> From: "Skyler" <skchopperguy at gmail.com>
> To: "Gary Nieboer" <gary at netdiverse.com>
> Cc: spce-user at lists.sipwise.com
> Sent: Wednesday, March 27, 2013 11:48:44 AM
> Subject: Re: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
>
> My guess (as I've not actually had this issue) is that your contact
> would look something like:
>
> Contact: <sip:ngcp-lb at your.ext.ip.address:5060;ngcpct='sip:127.0.0.1:5080'>
>
> Skyler
>
> On 3/27/2013 9:42 AM, Gary Nieboer wrote:
>> I will give a couple of the items advised a try.
>>
>> Should there be more than one uri in the Contact field?  If so, that may
>> be the quick fix.  The contact field I captured a few weeks ago only has
>> one, and it is the loopback/proxy address of 127.0.0.1:5080
>> <http://127.0.0.1:5080>
>>
>> -Gary
>> --
>> Gary Nieboer
>>
>>
>> 877-763-1014 Toll Free  |  801-657-5484 Utah  |  615-800-6946 Tennessee
>> | 801-694-6106 Mobile
>> gary at netdiverse.com <mailto:gary at netdiverse.com>  | www.netdiverse.com
>> <http://www.netdiverse.com/>
>>
>>
>> On Wed, Mar 27, 2013 at 10:31 AM, Skyler <skchopperguy at gmail.com
>> <mailto:skchopperguy at gmail.com>> wrote:
>>
>>      That makes sense, also might just be that the ngcpct sip uri is the
>>      last uri read in the contact.
>>
>>        Maybe Sylvester and Gary could test this out as they have the OP.
>>
>>      What would a good test look like to base64-encode that ngcpct in a
>>      customtt.tt2?
>>
>>      Skyler
>>
>>
>>      On 3/27/2013 9:07 AM, Andreas Granig wrote:
>>
>>          Hi,
>>
>>          On 03/27/2013 04:55 PM, Skyler wrote:
>>
>>                 I created an lo:0 with a private IP address
>>              (10.10.xxx.xxx) for lb >
>>              b2buas: & proxies > sip:. That may help in this case as its
>>              likely the
>>              127.0.0.1 which is confusing the upstream proxy.
>>
>>
>>          I've heard of endpoints disliking the SIP uri with the quotes...
>>          so one
>>          option could be to convert the "ngcpct='sip:127.0.0.1:5080
>>          <http://127.0.0.1:5080>'" and
>>          base64-encode the "sip:127.0.0.1:5080 <http://127.0.0.1:5080>",
>>          so it gets something like
>>          "ngcpct=__c2lwOjEyNy4wLjAuMTo1MDgwCg==" instead.
>>
>>          Andreas
>>
>>
>>      _________________________________________________
>>      Spce-user mailing list
>>      Spce-user at lists.sipwise.com <mailto:Spce-user at lists.sipwise.com>
>>      http://lists.sipwise.com/__listinfo/spce-user
>>      <http://lists.sipwise.com/listinfo/spce-user>
>>
>>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
>
>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
>

_______________________________________________
Spce-user mailing list
Spce-user at lists.sipwise.com
http://lists.sipwise.com/listinfo/spce-user
-------------- next part --------------
interface: any
filter: (ip or ip6) and ( port 5060 )
match: (^|CSeq:\s?\d* )(INVITE|ACK|CANCEL|BYE|MESSAGE|REFER|PRACK|INFO|UPDATE)

U 2013/03/30 16:13:10.763369 173.218.47.226:5060 -> 208.73.199.250:5060
INVITE sip:4172302354 at 208.73.199.250 SIP/2.0'
Via: SIP/2.0/UDP 192.168.1.15:5060;branch=z9hG4bK383849770;rport'
From: "4179730088" <sip:4179730088 at 208.73.199.250>;tag=381199400'
To: <sip:4172302354 at 208.73.199.250>'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF'
CSeq: 60 INVITE'
Contact: "4179730088" <sip:4179730088 at 192.168.1.15:5060>'
Max-Forwards: 70'
User-Agent: Grandstream GXV3140 1.0.7.50'
Privacy: none'
P-Preferred-Identity: "4179730088" <sip:4179730088 at 208.73.199.250>'
Supported: replaces, path, timer, eventlist'
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE'
Content-Type: application/sdp'
Accept: application/sdp, application/dtmf-relay'
Content-Length:   359'
'
v=0'
o=4179730088 8000 8000 IN IP4 192.168.1.15'
s=SIP Call'
c=IN IP4 192.168.1.15'
t=0 0'
m=audio 5004 RTP/AVP 0 8 4 18 9 3 101'
a=sendrecv'
a=rtpmap:0 PCMU/8000'
a=ptime:20'
a=rtpmap:8 PCMA/8000'
a=rtpmap:4 G723/8000'
a=rtpmap:18 G729/8000'
a=fmtp:18 annexb=no'
a=rtpmap:9 G722/8000'
a=rtpmap:3 GSM/8000'
a=rtpmap:101 telephone-event/8000'
a=fmtp:101 0-15'


U 2013/03/30 16:13:10.763882 10.10.10.1:5060 -> 10.10.10.1:5062
INVITE sip:4172302354 at 208.73.199.250 SIP/2.0'
Record-Route: <sip:10.10.10.1;r2=on;lr=on;ftag=381199400;nat=yes;ngcplb=yes>'
Record-Route: <sip:208.73.199.250;r2=on;lr=on;ftag=381199400;nat=yes;ngcplb=yes>'
Via: SIP/2.0/UDP 10.10.10.1;branch=z9hG4bK3b2b.dafbd44a4785d7d9d19316ba64b6277a.0'
Via: SIP/2.0/UDP 192.168.1.15:5060;received=173.218.47.226;branch=z9hG4bK383849770;rport=5060'
From: "4179730088" <sip:4179730088 at 208.73.199.250>;tag=381199400'
To: <sip:4172302354 at 208.73.199.250>'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF'
CSeq: 60 INVITE'
Contact: "4179730088" <sip:4179730088 at 192.168.1.15:5060;alias=173.218.47.226~5060~1>'
Max-Forwards: 69'
User-Agent: Grandstream GXV3140 1.0.7.50'
Privacy: none'
P-Preferred-Identity: "4179730088" <sip:4179730088 at 208.73.199.250>'
Supported: replaces, path, timer, eventlist'
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE'
Content-Type: application/sdp'
Accept: application/sdp, application/dtmf-relay'
Content-Length:   408'
P-NGCP-Src-Ip: 173.218.47.226'
P-NGCP-Src-Port: 5060'
P-NGCP-Src-Proto: udp'
P-NGCP-Src-Af: 4'
P-NGCP-Src-Nat: 1'
'
v=0'
o=4179730088 8000 8000 IN IP4 192.168.1.15'
s=SIP Call'
c=IN IP4 173.218.47.226'
t=0 0'
m=audio 5004 RTP/AVP 0 8 4 18 9 3 101'
a=sendrecv'
a=rtpmap:0 PCMU/8000'
a=ptime:20'
a=rtpmap:8 PCMA/8000'
a=rtpmap:4 G723/8000'
a=rtpmap:18 G729/8000'
a=fmtp:18 annexb=no'
a=rtpmap:9 G722/8000'
a=rtpmap:3 GSM/8000'
a=rtpmap:101 telephone-event/8000'
a=fmtp:101 0-15'
a=direction:active'
a=oldmediaip:192.168.1.15'


U 2013/03/30 16:13:10.763997 10.10.10.1:5062 -> 10.10.10.1:5060
SIP/2.0 100 Trying'
Via: SIP/2.0/UDP 10.10.10.1;branch=z9hG4bK3b2b.dafbd44a4785d7d9d19316ba64b6277a.0'
Via: SIP/2.0/UDP 192.168.1.15:5060;received=173.218.47.226;branch=z9hG4bK383849770;rport=5060'
From: "4179730088" <sip:4179730088 at 208.73.199.250>;tag=381199400'
To: <sip:4172302354 at 208.73.199.250>'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF'
CSeq: 60 INVITE'
Server: Sipwise NGCP Proxy 2.X'
Content-Length: 0'
'


U 2013/03/30 16:13:10.764180 208.73.199.250:5060 -> 173.218.47.226:5060
SIP/2.0 100 Trying'
Via: SIP/2.0/UDP 192.168.1.15:5060;received=173.218.47.226;branch=z9hG4bK383849770;rport=5060'
From: "4179730088" <sip:4179730088 at 208.73.199.250>;tag=381199400'
To: <sip:4172302354 at 208.73.199.250>'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF'
CSeq: 60 INVITE'
Server: Sipwise NGCP Proxy 2.X'
Content-Length: 0'
'


U 2013/03/30 16:13:10.765351 10.10.10.1:5062 -> 10.10.10.1:5060
SIP/2.0 407 Proxy Authentication Required'
Via: SIP/2.0/UDP 10.10.10.1;branch=z9hG4bK3b2b.dafbd44a4785d7d9d19316ba64b6277a.0'
Via: SIP/2.0/UDP 192.168.1.15:5060;received=173.218.47.226;branch=z9hG4bK383849770;rport=5060'
From: "4179730088" <sip:4179730088 at 208.73.199.250>;tag=381199400'
To: <sip:4172302354 at 208.73.199.250>;tag=b5fb22e342502bf1fc1cee71cf6cdd66.1fdd'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF'
CSeq: 60 INVITE'
Proxy-Authenticate: Digest realm="208.73.199.250", nonce="UVdWkVFXVWX1N74TGFNGQLVlLv5984ca"'
Server: Sipwise NGCP Proxy 2.X'
Content-Length: 0'
'


U 2013/03/30 16:13:10.765473 208.73.199.250:5060 -> 173.218.47.226:5060
SIP/2.0 407 Proxy Authentication Required'
Via: SIP/2.0/UDP 192.168.1.15:5060;received=173.218.47.226;branch=z9hG4bK383849770;rport=5060'
From: "4179730088" <sip:4179730088 at 208.73.199.250>;tag=381199400'
To: <sip:4172302354 at 208.73.199.250>;tag=b5fb22e342502bf1fc1cee71cf6cdd66.1fdd'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF'
CSeq: 60 INVITE'
Proxy-Authenticate: Digest realm="208.73.199.250", nonce="UVdWkVFXVWX1N74TGFNGQLVlLv5984ca"'
Server: Sipwise NGCP Proxy 2.X'
Content-Length: 0'
'


U 2013/03/30 16:13:10.830746 173.218.47.226:5060 -> 208.73.199.250:5060
ACK sip:4172302354 at 208.73.199.250 SIP/2.0'
Via: SIP/2.0/UDP 192.168.1.15:5060;branch=z9hG4bK383849770;rport'
From: "4179730088" <sip:4179730088 at 208.73.199.250>;tag=381199400'
To: <sip:4172302354 at 208.73.199.250>;tag=b5fb22e342502bf1fc1cee71cf6cdd66.1fdd'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF'
CSeq: 60 ACK'
Content-Length: 0'
'


U 2013/03/30 16:13:10.830910 10.10.10.1:5060 -> 10.10.10.1:5062
ACK sip:4172302354 at 208.73.199.250 SIP/2.0'
Max-Forwards: 10'
Record-Route: <sip:10.10.10.1;r2=on;lr=on;ftag=381199400;ngcplb=yes>'
Record-Route: <sip:208.73.199.250;r2=on;lr=on;ftag=381199400;ngcplb=yes>'
Via: SIP/2.0/UDP 10.10.10.1;branch=z9hG4bK3b2b.dafbd44a4785d7d9d19316ba64b6277a.0'
Via: SIP/2.0/UDP 192.168.1.15:5060;received=173.218.47.226;branch=z9hG4bK383849770;rport=5060'
From: "4179730088" <sip:4179730088 at 208.73.199.250>;tag=381199400'
To: <sip:4172302354 at 208.73.199.250>;tag=b5fb22e342502bf1fc1cee71cf6cdd66.1fdd'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF'
CSeq: 60 ACK'
Content-Length: 0'
P-NGCP-Src-Ip: 173.218.47.226'
P-NGCP-Src-Port: 5060'
P-NGCP-Src-Proto: udp'
P-NGCP-Src-Af: 4'
'


U 2013/03/30 16:13:10.836163 173.218.47.226:5060 -> 208.73.199.250:5060
INVITE sip:4172302354 at 208.73.199.250 SIP/2.0'
Via: SIP/2.0/UDP 192.168.1.15:5060;branch=z9hG4bK1893375477;rport'
From: "4179730088" <sip:4179730088 at 208.73.199.250>;tag=381199400'
To: <sip:4172302354 at 208.73.199.250>'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF'
CSeq: 61 INVITE'
Contact: "4179730088" <sip:4179730088 at 192.168.1.15:5060>'
Proxy-Authorization: Digest username="4179730088", realm="208.73.199.250", nonce="UVdWkVFXVWX1N74TGFNGQLVlLv5984ca", uri="sip:4172302354 at 208.73.199.250", response="70ce8a797b576a44b5161134b0f25f30", algorithm=MD5'
Max-Forwards: 70'
User-Agent: Grandstream GXV3140 1.0.7.50'
Privacy: none'
P-Preferred-Identity: "4179730088" <sip:4179730088 at 208.73.199.250>'
Supported: replaces, path, timer, eventlist'
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE'
Content-Type: application/sdp'
Accept: application/sdp, application/dtmf-relay'
Content-Length:   359'
'
v=0'
o=4179730088 8000 8000 IN IP4 192.168.1.15'
s=SIP Call'
c=IN IP4 192.168.1.15'
t=0 0'
m=audio 5004 RTP/AVP 0 8 4 18 9 3 101'
a=sendrecv'
a=rtpmap:0 PCMU/8000'
a=ptime:20'
a=rtpmap:8 PCMA/8000'
a=rtpmap:4 G723/8000'
a=rtpmap:18 G729/8000'
a=fmtp:18 annexb=no'
a=rtpmap:9 G722/8000'
a=rtpmap:3 GSM/8000'
a=rtpmap:101 telephone-event/8000'
a=fmtp:101 0-15'


U 2013/03/30 16:13:10.836390 10.10.10.1:5060 -> 10.10.10.1:5062
INVITE sip:4172302354 at 208.73.199.250 SIP/2.0'
Record-Route: <sip:10.10.10.1;r2=on;lr=on;ftag=381199400;nat=yes;ngcplb=yes>'
Record-Route: <sip:208.73.199.250;r2=on;lr=on;ftag=381199400;nat=yes;ngcplb=yes>'
Via: SIP/2.0/UDP 10.10.10.1;branch=z9hG4bK4b2b.8ca2ffe7be2989c4827a439f3845b7f1.0'
Via: SIP/2.0/UDP 192.168.1.15:5060;received=173.218.47.226;branch=z9hG4bK1893375477;rport=5060'
From: "4179730088" <sip:4179730088 at 208.73.199.250>;tag=381199400'
To: <sip:4172302354 at 208.73.199.250>'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF'
CSeq: 61 INVITE'
Contact: "4179730088" <sip:4179730088 at 192.168.1.15:5060;alias=173.218.47.226~5060~1>'
Proxy-Authorization: Digest username="4179730088", realm="208.73.199.250", nonce="UVdWkVFXVWX1N74TGFNGQLVlLv5984ca", uri="sip:4172302354 at 208.73.199.250", response="70ce8a797b576a44b5161134b0f25f30", algorithm=MD5'
Max-Forwards: 69'
User-Agent: Grandstream GXV3140 1.0.7.50'
Privacy: none'
P-Preferred-Identity: "4179730088" <sip:4179730088 at 208.73.199.250>'
Supported: replaces, path, timer, eventlist'
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE'
Content-Type: application/sdp'
Accept: application/sdp, application/dtmf-relay'
Content-Length:   408'
P-NGCP-Src-Ip: 173.218.47.226'
P-NGCP-Src-Port: 5060'
P-NGCP-Src-Proto: udp'
P-NGCP-Src-Af: 4'
P-NGCP-Src-Nat: 1'
'
v=0'
o=4179730088 8000 8000 IN IP4 192.168.1.15'
s=SIP Call'
c=IN IP4 173.218.47.226'
t=0 0'
m=audio 5004 RTP/AVP 0 8 4 18 9 3 101'
a=sendrecv'
a=rtpmap:0 PCMU/8000'
a=ptime:20'
a=rtpmap:8 PCMA/8000'
a=rtpmap:4 G723/8000'
a=rtpmap:18 G729/8000'
a=fmtp:18 annexb=no'
a=rtpmap:9 G722/8000'
a=rtpmap:3 GSM/8000'
a=rtpmap:101 telephone-event/8000'
a=fmtp:101 0-15'
a=direction:active'
a=oldmediaip:192.168.1.15'


U 2013/03/30 16:13:10.836486 10.10.10.1:5062 -> 10.10.10.1:5060
SIP/2.0 100 Trying'
Via: SIP/2.0/UDP 10.10.10.1;branch=z9hG4bK4b2b.8ca2ffe7be2989c4827a439f3845b7f1.0'
Via: SIP/2.0/UDP 192.168.1.15:5060;received=173.218.47.226;branch=z9hG4bK1893375477;rport=5060'
From: "4179730088" <sip:4179730088 at 208.73.199.250>;tag=381199400'
To: <sip:4172302354 at 208.73.199.250>'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF'
CSeq: 61 INVITE'
Server: Sipwise NGCP Proxy 2.X'
Content-Length: 0'
'


U 2013/03/30 16:13:10.836723 208.73.199.250:5060 -> 173.218.47.226:5060
SIP/2.0 100 Trying'
Via: SIP/2.0/UDP 192.168.1.15:5060;received=173.218.47.226;branch=z9hG4bK1893375477;rport=5060'
From: "4179730088" <sip:4179730088 at 208.73.199.250>;tag=381199400'
To: <sip:4172302354 at 208.73.199.250>'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF'
CSeq: 61 INVITE'
Server: Sipwise NGCP Proxy 2.X'
Content-Length: 0'
'


U 2013/03/30 16:13:10.837896 10.10.10.1:5062 -> 10.10.10.1:5060
SIP/2.0 101 Connecting'
Via: SIP/2.0/UDP 10.10.10.1;branch=z9hG4bK4b2b.8ca2ffe7be2989c4827a439f3845b7f1.0'
Via: SIP/2.0/UDP 192.168.1.15:5060;received=173.218.47.226;branch=z9hG4bK1893375477;rport=5060'
From: "4179730088" <sip:4179730088 at 208.73.199.250>;tag=381199400'
To: <sip:4172302354 at 208.73.199.250>'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF'
CSeq: 61 INVITE'
P-NGCP-Authorization: 4179730088 at 208.73.199.250'
P-NGCP-Authorized: 1'
P-Caller-UUID: fdb671df-abd4-4191-9cda-1c580f01bc27'
Server: Sipwise NGCP Proxy 2.X'
Content-Length: 0'
'


U 2013/03/30 16:13:10.841653 10.10.10.1:5080 -> 10.10.10.1:5060
INVITE sip:4172302354 at 208.73.198.12:5060;transport=udp SIP/2.0'
Via: SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKPfekkaGS;rport'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>'
CSeq: 10 INVITE'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
Contact: <sip:10.10.10.1:5080>'
Route: <sip:10.10.10.1:5060;received='sip:208.73.198.12:5060;transport=udp';lr>'
Supported: replaces, path, timer, eventlist'
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE'
Accept: application/sdp, application/dtmf-relay'
P-Asserted-Identity: <sip:4179730088 at 208.73.199.250>'
P-D-Uri: sip:10.10.10.1:5060;received='sip:208.73.198.12:5060;transport=udp''
Content-Type: application/sdp'
Content-Length: 429'
'
v=0'
o=4179730088 8000 8000 IN IP4 208.73.199.250'
s=SIP Call'
c=IN IP4 208.73.199.250'
t=0 0'
m=audio 30092 RTP/AVP 0 8 4 18 9 3 101'
a=sendrecv'
a=rtpmap:0 PCMU/8000'
a=ptime:20'
a=rtpmap:8 PCMA/8000'
a=rtpmap:4 G723/8000'
a=rtpmap:18 G729/8000'
a=fmtp:18 annexb=no'
a=rtpmap:9 G722/8000'
a=rtpmap:3 GSM/8000'
a=rtpmap:101 telephone-event/8000'
a=fmtp:101 0-15'
a=direction:active'
a=oldmediaip:192.168.1.15'
a=nortpproxy:yes'


U 2013/03/30 16:13:10.841904 208.73.199.250:5060 -> 208.73.198.12:5060
INVITE sip:4172302354 at 208.73.198.12:5060;transport=udp SIP/2.0'
Max-Forwards: 10'
Record-Route: <sip:208.73.199.250;r2=on;lr=on;ftag=57FE8F4B-51575566000CD752-90AB5700;ngcplb=yes>'
Record-Route: <sip:10.10.10.1;r2=on;lr=on;ftag=57FE8F4B-51575566000CD752-90AB5700;ngcplb=yes>'
Via: SIP/2.0/UDP 208.73.199.250;branch=z9hG4bK3642.4b4263c869b06a079cb58096dc4f3039.0'
Via: SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKPfekkaGS;rport=5080'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>'
CSeq: 10 INVITE'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
Supported: replaces, path, timer, eventlist'
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE'
Accept: application/sdp, application/dtmf-relay'
P-Asserted-Identity: <sip:4179730088 at 208.73.199.250>'
Content-Type: application/sdp'
Content-Length: 429'
Contact: <sip:ngcp-lb at 208.73.199.250:5060;ngcpct='sip:10.10.10.1:5080'>'
'
v=0'
o=4179730088 8000 8000 IN IP4 208.73.199.250'
s=SIP Call'
c=IN IP4 208.73.199.250'
t=0 0'
m=audio 30092 RTP/AVP 0 8 4 18 9 3 101'
a=sendrecv'
a=rtpmap:0 PCMU/8000'
a=ptime:20'
a=rtpmap:8 PCMA/8000'
a=rtpmap:4 G723/8000'
a=rtpmap:18 G729/8000'
a=fmtp:18 annexb=no'
a=rtpmap:9 G722/8000'
a=rtpmap:3 GSM/8000'
a=rtpmap:101 telephone-event/8000'
a=fmtp:101 0-15'
a=direction:active'
a=oldmediaip:192.168.1.15'
a=nortpproxy:yes'


U 2013/03/30 16:13:10.848816 208.73.198.12:5060 -> 208.73.199.250:5060
SIP/2.0 100 Trying'
Via: SIP/2.0/UDP 208.73.199.250;branch=z9hG4bK3642.4b4263c869b06a079cb58096dc4f3039.0,SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKPfekkaGS;rport=5080'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
CSeq: 10 INVITE'
Content-Length: 0'
'


U 2013/03/30 16:13:10.848916 10.10.10.1:5060 -> 10.10.10.1:5080
SIP/2.0 100 Trying'
Via: SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKPfekkaGS;rport=5080'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
CSeq: 10 INVITE'
Content-Length: 0'
P-NGCP-Src-Ip: 208.73.198.12'
P-NGCP-Src-Port: 5060'
P-NGCP-Src-Proto: udp'
P-NGCP-Src-Af: 4'
'


U 2013/03/30 16:13:12.406763 208.73.198.12:5060 -> 208.73.199.250:5060
SIP/2.0 180 Ringing'
Via: SIP/2.0/UDP 208.73.199.250;branch=z9hG4bK3642.4b4263c869b06a079cb58096dc4f3039.0,SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKPfekkaGS;rport=5080'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
CSeq: 10 INVITE'
Allow: UPDATE,INVITE,CANCEL,BYE,ACK'
Content-Length: 0'
'


U 2013/03/30 16:13:12.407078 10.10.10.1:5060 -> 10.10.10.1:5080
SIP/2.0 180 Ringing'
Via: SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKPfekkaGS;rport=5080'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
CSeq: 10 INVITE'
Allow: UPDATE,INVITE,CANCEL,BYE,ACK'
Content-Length: 0'
P-NGCP-Src-Ip: 208.73.198.12'
P-NGCP-Src-Port: 5060'
P-NGCP-Src-Proto: udp'
P-NGCP-Src-Af: 4'
'


U 2013/03/30 16:13:12.407447 10.10.10.1:5062 -> 10.10.10.1:5060
SIP/2.0 180 Ringing'
Record-Route: <sip:10.10.10.1:5062;lr=on;ftag=381199400;did=5b1.2f3;mpd=ii;rtpprx=yes;vsf=Yk1rc2RyRTlyYmdOdGFBa2diWXhiTWtzZHJFOXI->'
Record-Route: <sip:10.10.10.1;r2=on;lr=on;ftag=381199400;nat=yes;ngcplb=yes>'
Record-Route: <sip:208.73.199.250;r2=on;lr=on;ftag=381199400;nat=yes;ngcplb=yes>'
Via: SIP/2.0/UDP 10.10.10.1;branch=z9hG4bK4b2b.8ca2ffe7be2989c4827a439f3845b7f1.0'
Via: SIP/2.0/UDP 192.168.1.15:5060;received=173.218.47.226;branch=z9hG4bK1893375477;rport=5060'
From: "4179730088" <sip:4179730088 at 208.73.199.250>;tag=381199400'
To: <sip:4172302354 at 208.73.199.250>;tag=747E2E0E-51575566000CD69C-903AE700'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF'
CSeq: 61 INVITE'
Allow: UPDATE,INVITE,CANCEL,BYE,ACK'
Contact: <sip:4172302354 at 10.10.10.1:5080>'
Content-Length: 0'
'


U 2013/03/30 16:13:12.407664 208.73.199.250:5060 -> 173.218.47.226:5060
SIP/2.0 180 Ringing'
Record-Route: <sip:10.10.10.1:5062;lr=on;ftag=381199400;did=5b1.2f3;mpd=ii;rtpprx=yes;vsf=Yk1rc2RyRTlyYmdOdGFBa2diWXhiTWtzZHJFOXI->'
Record-Route: <sip:10.10.10.1;r2=on;lr=on;ftag=381199400;nat=yes;ngcplb=yes>'
Record-Route: <sip:208.73.199.250;r2=on;lr=on;ftag=381199400;nat=yes;ngcplb=yes>'
Via: SIP/2.0/UDP 192.168.1.15:5060;received=173.218.47.226;branch=z9hG4bK1893375477;rport=5060'
From: "4179730088" <sip:4179730088 at 208.73.199.250>;tag=381199400'
To: <sip:4172302354 at 208.73.199.250>;tag=747E2E0E-51575566000CD69C-903AE700'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF'
CSeq: 61 INVITE'
Allow: UPDATE,INVITE,CANCEL,BYE,ACK'
Content-Length: 0'
Contact: <sip:ngcp-lb at 208.73.199.250:5060;ngcpct='sip:10.10.10.1:5080'>'
'


U 2013/03/30 16:13:28.768836 208.73.198.12:5060 -> 208.73.199.250:5060
SIP/2.0 200 OK'
Via: SIP/2.0/UDP 208.73.199.250;branch=z9hG4bK3642.4b4263c869b06a079cb58096dc4f3039.0,SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKPfekkaGS;rport=5080'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
CSeq: 10 INVITE'
Contact: sip:208.73.198.12:5060'
Content-Type: application/sdp'
Content-Length: 193'
'
v=0'
o=- 1 1 IN IP4 208.73.198.12'
s=SIP Call'
c=IN IP4 208.73.198.13'
t=0 0'
m=audio 49166 RTP/AVP 0 101'
a=ptime:20'
a=rtpmap:0 PCMU/8000'
a=rtpmap:101 telephone-event/8000'
a=fmtp:101 0-15'


U 2013/03/30 16:13:28.769231 10.10.10.1:5060 -> 10.10.10.1:5080
SIP/2.0 200 OK'
Via: SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKPfekkaGS;rport=5080'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
CSeq: 10 INVITE'
Contact: sip:208.73.198.12:5060'
Content-Type: application/sdp'
Content-Length: 193'
P-NGCP-Src-Ip: 208.73.198.12'
P-NGCP-Src-Port: 5060'
P-NGCP-Src-Proto: udp'
P-NGCP-Src-Af: 4'
'
v=0'
o=- 1 1 IN IP4 208.73.198.12'
s=SIP Call'
c=IN IP4 208.73.198.13'
t=0 0'
m=audio 49166 RTP/AVP 0 101'
a=ptime:20'
a=rtpmap:0 PCMU/8000'
a=rtpmap:101 telephone-event/8000'
a=fmtp:101 0-15'


U 2013/03/30 16:13:28.769893 10.10.10.1:5062 -> 10.10.10.1:5060
SIP/2.0 200 OK'
Record-Route: <sip:10.10.10.1:5062;lr=on;ftag=381199400;did=5b1.2f3;mpd=ii;rtpprx=yes;vsf=Yk1rc2RyRTlyYmdOdGFBa2diWXhiTWtzZHJFOXI->'
Record-Route: <sip:10.10.10.1;r2=on;lr=on;ftag=381199400;nat=yes;ngcplb=yes>'
Record-Route: <sip:208.73.199.250;r2=on;lr=on;ftag=381199400;nat=yes;ngcplb=yes>'
Via: SIP/2.0/UDP 10.10.10.1;branch=z9hG4bK4b2b.8ca2ffe7be2989c4827a439f3845b7f1.0'
Via: SIP/2.0/UDP 192.168.1.15:5060;received=173.218.47.226;branch=z9hG4bK1893375477;rport=5060'
From: "4179730088" <sip:4179730088 at 208.73.199.250>;tag=381199400'
To: <sip:4172302354 at 208.73.199.250>;tag=747E2E0E-51575566000CD69C-903AE700'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF'
CSeq: 61 INVITE'
Contact: <sip:4172302354 at 10.10.10.1:5080>'
Content-Type: application/sdp'
Content-Length: 213'
'
v=0'
o=- 1 1 IN IP4 208.73.199.250'
s=SIP Call'
c=IN IP4 208.73.199.250'
t=0 0'
m=audio 30094 RTP/AVP 0 101'
a=ptime:20'
a=rtpmap:0 PCMU/8000'
a=rtpmap:101 telephone-event/8000'
a=fmtp:101 0-15'
a=nortpproxy:yes'


U 2013/03/30 16:13:28.770108 208.73.199.250:5060 -> 173.218.47.226:5060
SIP/2.0 200 OK'
Record-Route: <sip:10.10.10.1:5062;lr=on;ftag=381199400;did=5b1.2f3;mpd=ii;rtpprx=yes;vsf=Yk1rc2RyRTlyYmdOdGFBa2diWXhiTWtzZHJFOXI->'
Record-Route: <sip:10.10.10.1;r2=on;lr=on;ftag=381199400;nat=yes;ngcplb=yes>'
Record-Route: <sip:208.73.199.250;r2=on;lr=on;ftag=381199400;nat=yes;ngcplb=yes>'
Via: SIP/2.0/UDP 192.168.1.15:5060;received=173.218.47.226;branch=z9hG4bK1893375477;rport=5060'
From: "4179730088" <sip:4179730088 at 208.73.199.250>;tag=381199400'
To: <sip:4172302354 at 208.73.199.250>;tag=747E2E0E-51575566000CD69C-903AE700'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF'
CSeq: 61 INVITE'
Content-Type: application/sdp'
Content-Length: 213'
Contact: <sip:ngcp-lb at 208.73.199.250:5060;ngcpct='sip:10.10.10.1:5080'>'
'
v=0'
o=- 1 1 IN IP4 208.73.199.250'
s=SIP Call'
c=IN IP4 208.73.199.250'
t=0 0'
m=audio 30094 RTP/AVP 0 101'
a=ptime:20'
a=rtpmap:0 PCMU/8000'
a=rtpmap:101 telephone-event/8000'
a=fmtp:101 0-15'
a=nortpproxy:yes'


U 2013/03/30 16:13:28.845401 173.218.47.226:5060 -> 208.73.199.250:5060
ACK sip:ngcp-lb at 208.73.199.250:5060;ngcpct='sip:10.10.10.1:5080' SIP/2.0'
Via: SIP/2.0/UDP 192.168.1.15:5060;branch=z9hG4bK236821213;rport'
Route: <sip:208.73.199.250;r2=on;lr=on;ftag=381199400;nat=yes;ngcplb=yes>'
Route: <sip:10.10.10.1;r2=on;lr=on;ftag=381199400;nat=yes;ngcplb=yes>'
Route: <sip:10.10.10.1:5062;lr=on;ftag=381199400;did=5b1.2f3;mpd=ii;rtpprx=yes;vsf=Yk1rc2RyRTlyYmdOdGFBa2diWXhiTWtzZHJFOXI->'
From: "4179730088" <sip:4179730088 at 208.73.199.250>;tag=381199400'
To: <sip:4172302354 at 208.73.199.250>;tag=747E2E0E-51575566000CD69C-903AE700'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF'
CSeq: 61 ACK'
Contact: <sip:4179730088 at 192.168.1.15:5060>'
Proxy-Authorization: Digest username="4179730088", realm="208.73.199.250", nonce="UVdWkVFXVWX1N74TGFNGQLVlLv5984ca", uri="sip:4172302354 at 208.73.199.250", response="70ce8a797b576a44b5161134b0f25f30", algorithm=MD5'
Max-Forwards: 70'
Supported: replaces, path, timer, eventlist'
User-Agent: Grandstream GXV3140 1.0.7.50'
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE'
Content-Length: 0'
'


U 2013/03/30 16:13:28.845675 10.10.10.1:5060 -> 10.10.10.1:5062
ACK sip:10.10.10.1:5080 SIP/2.0'
Record-Route: <sip:10.10.10.1;r2=on;lr=on;ftag=381199400;nat=yes;ngcplb=yes>'
Record-Route: <sip:208.73.199.250;r2=on;lr=on;ftag=381199400;nat=yes;ngcplb=yes>'
Via: SIP/2.0/UDP 10.10.10.1;branch=z9hG4bK4b2b.46e940990a07dce784849331e5ad883a.0'
Via: SIP/2.0/UDP 192.168.1.15:5060;received=173.218.47.226;branch=z9hG4bK236821213;rport=5060'
Route: <sip:10.10.10.1:5062;lr=on;ftag=381199400;did=5b1.2f3;mpd=ii;rtpprx=yes;vsf=Yk1rc2RyRTlyYmdOdGFBa2diWXhiTWtzZHJFOXI->'
From: "4179730088" <sip:4179730088 at 208.73.199.250>;tag=381199400'
To: <sip:4172302354 at 208.73.199.250>;tag=747E2E0E-51575566000CD69C-903AE700'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF'
CSeq: 61 ACK'
Contact: <sip:4179730088 at 192.168.1.15:5060;alias=173.218.47.226~5060~1>'
Proxy-Authorization: Digest username="4179730088", realm="208.73.199.250", nonce="UVdWkVFXVWX1N74TGFNGQLVlLv5984ca", uri="sip:4172302354 at 208.73.199.250", response="70ce8a797b576a44b5161134b0f25f30", algorithm=MD5'
Max-Forwards: 69'
Supported: replaces, path, timer, eventlist'
User-Agent: Grandstream GXV3140 1.0.7.50'
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE'
Content-Length: 0'
P-NGCP-Src-Ip: 173.218.47.226'
P-NGCP-Src-Port: 5060'
P-NGCP-Src-Proto: udp'
P-NGCP-Src-Af: 4'
P-NGCP-Src-Nat: 1'
'


U 2013/03/30 16:13:28.846136 10.10.10.1:5080 -> 208.73.198.12:5060
ACK sip:208.73.198.12:5060 SIP/2.0'
Via: SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKXCZQia5D;rport'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
CSeq: 10 ACK'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
Contact: <sip:10.10.10.1:5080>'
Supported: replaces, path, timer, eventlist'
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE'
Content-Length: 0'
'


U 2013/03/30 16:13:29.269359 208.73.198.12:5060 -> 208.73.199.250:5060
SIP/2.0 200 OK'
Via: SIP/2.0/UDP 208.73.199.250;branch=z9hG4bK3642.4b4263c869b06a079cb58096dc4f3039.0,SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKPfekkaGS;rport=5080'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
CSeq: 10 INVITE'
Contact: sip:208.73.198.12:5060'
Content-Type: application/sdp'
Content-Length: 193'
'
v=0'
o=- 1 1 IN IP4 208.73.198.12'
s=SIP Call'
c=IN IP4 208.73.198.13'
t=0 0'
m=audio 49166 RTP/AVP 0 101'
a=ptime:20'
a=rtpmap:0 PCMU/8000'
a=rtpmap:101 telephone-event/8000'
a=fmtp:101 0-15'


U 2013/03/30 16:13:29.269546 10.10.10.1:5060 -> 10.10.10.1:5080
SIP/2.0 200 OK'
Via: SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKPfekkaGS;rport=5080'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
CSeq: 10 INVITE'
Contact: sip:208.73.198.12:5060'
Content-Type: application/sdp'
Content-Length: 193'
P-NGCP-Src-Ip: 208.73.198.12'
P-NGCP-Src-Port: 5060'
P-NGCP-Src-Proto: udp'
P-NGCP-Src-Af: 4'
'
v=0'
o=- 1 1 IN IP4 208.73.198.12'
s=SIP Call'
c=IN IP4 208.73.198.13'
t=0 0'
m=audio 49166 RTP/AVP 0 101'
a=ptime:20'
a=rtpmap:0 PCMU/8000'
a=rtpmap:101 telephone-event/8000'
a=fmtp:101 0-15'


U 2013/03/30 16:13:29.269619 10.10.10.1:5080 -> 208.73.198.12:5060
ACK sip:208.73.198.12:5060 SIP/2.0'
Via: SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKXCZQia5D;rport'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
CSeq: 10 ACK'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
Contact: <sip:10.10.10.1:5080>'
Supported: replaces, path, timer, eventlist'
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE'
Content-Length: 0'
'


U 2013/03/30 16:13:30.269175 208.73.198.12:5060 -> 208.73.199.250:5060
SIP/2.0 200 OK'
Via: SIP/2.0/UDP 208.73.199.250;branch=z9hG4bK3642.4b4263c869b06a079cb58096dc4f3039.0,SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKPfekkaGS;rport=5080'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
CSeq: 10 INVITE'
Contact: sip:208.73.198.12:5060'
Content-Type: application/sdp'
Content-Length: 193'
'
v=0'
o=- 1 1 IN IP4 208.73.198.12'
s=SIP Call'
c=IN IP4 208.73.198.13'
t=0 0'
m=audio 49166 RTP/AVP 0 101'
a=ptime:20'
a=rtpmap:0 PCMU/8000'
a=rtpmap:101 telephone-event/8000'
a=fmtp:101 0-15'


U 2013/03/30 16:13:30.269533 10.10.10.1:5060 -> 10.10.10.1:5080
SIP/2.0 200 OK'
Via: SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKPfekkaGS;rport=5080'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
CSeq: 10 INVITE'
Contact: sip:208.73.198.12:5060'
Content-Type: application/sdp'
Content-Length: 193'
P-NGCP-Src-Ip: 208.73.198.12'
P-NGCP-Src-Port: 5060'
P-NGCP-Src-Proto: udp'
P-NGCP-Src-Af: 4'
'
v=0'
o=- 1 1 IN IP4 208.73.198.12'
s=SIP Call'
c=IN IP4 208.73.198.13'
t=0 0'
m=audio 49166 RTP/AVP 0 101'
a=ptime:20'
a=rtpmap:0 PCMU/8000'
a=rtpmap:101 telephone-event/8000'
a=fmtp:101 0-15'


U 2013/03/30 16:13:30.269607 10.10.10.1:5080 -> 208.73.198.12:5060
ACK sip:208.73.198.12:5060 SIP/2.0'
Via: SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKXCZQia5D;rport'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
CSeq: 10 ACK'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
Contact: <sip:10.10.10.1:5080>'
Supported: replaces, path, timer, eventlist'
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE'
Content-Length: 0'
'


U 2013/03/30 16:13:32.268897 208.73.198.12:5060 -> 208.73.199.250:5060
SIP/2.0 200 OK'
Via: SIP/2.0/UDP 208.73.199.250;branch=z9hG4bK3642.4b4263c869b06a079cb58096dc4f3039.0,SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKPfekkaGS;rport=5080'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
CSeq: 10 INVITE'
Contact: sip:208.73.198.12:5060'
Content-Type: application/sdp'
Content-Length: 193'
'
v=0'
o=- 1 1 IN IP4 208.73.198.12'
s=SIP Call'
c=IN IP4 208.73.198.13'
t=0 0'
m=audio 49166 RTP/AVP 0 101'
a=ptime:20'
a=rtpmap:0 PCMU/8000'
a=rtpmap:101 telephone-event/8000'
a=fmtp:101 0-15'


U 2013/03/30 16:13:32.269272 10.10.10.1:5060 -> 10.10.10.1:5080
SIP/2.0 200 OK'
Via: SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKPfekkaGS;rport=5080'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
CSeq: 10 INVITE'
Contact: sip:208.73.198.12:5060'
Content-Type: application/sdp'
Content-Length: 193'
P-NGCP-Src-Ip: 208.73.198.12'
P-NGCP-Src-Port: 5060'
P-NGCP-Src-Proto: udp'
P-NGCP-Src-Af: 4'
'
v=0'
o=- 1 1 IN IP4 208.73.198.12'
s=SIP Call'
c=IN IP4 208.73.198.13'
t=0 0'
m=audio 49166 RTP/AVP 0 101'
a=ptime:20'
a=rtpmap:0 PCMU/8000'
a=rtpmap:101 telephone-event/8000'
a=fmtp:101 0-15'


U 2013/03/30 16:13:32.269350 10.10.10.1:5080 -> 208.73.198.12:5060
ACK sip:208.73.198.12:5060 SIP/2.0'
Via: SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKXCZQia5D;rport'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
CSeq: 10 ACK'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
Contact: <sip:10.10.10.1:5080>'
Supported: replaces, path, timer, eventlist'
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE'
Content-Length: 0'
'


U 2013/03/30 16:13:36.268284 208.73.198.12:5060 -> 208.73.199.250:5060
SIP/2.0 200 OK'
Via: SIP/2.0/UDP 208.73.199.250;branch=z9hG4bK3642.4b4263c869b06a079cb58096dc4f3039.0,SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKPfekkaGS;rport=5080'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
CSeq: 10 INVITE'
Contact: sip:208.73.198.12:5060'
Content-Type: application/sdp'
Content-Length: 193'
'
v=0'
o=- 1 1 IN IP4 208.73.198.12'
s=SIP Call'
c=IN IP4 208.73.198.13'
t=0 0'
m=audio 49166 RTP/AVP 0 101'
a=ptime:20'
a=rtpmap:0 PCMU/8000'
a=rtpmap:101 telephone-event/8000'
a=fmtp:101 0-15'


U 2013/03/30 16:13:36.268485 10.10.10.1:5060 -> 10.10.10.1:5080
SIP/2.0 200 OK'
Via: SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKPfekkaGS;rport=5080'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
CSeq: 10 INVITE'
Contact: sip:208.73.198.12:5060'
Content-Type: application/sdp'
Content-Length: 193'
P-NGCP-Src-Ip: 208.73.198.12'
P-NGCP-Src-Port: 5060'
P-NGCP-Src-Proto: udp'
P-NGCP-Src-Af: 4'
'
v=0'
o=- 1 1 IN IP4 208.73.198.12'
s=SIP Call'
c=IN IP4 208.73.198.13'
t=0 0'
m=audio 49166 RTP/AVP 0 101'
a=ptime:20'
a=rtpmap:0 PCMU/8000'
a=rtpmap:101 telephone-event/8000'
a=fmtp:101 0-15'


U 2013/03/30 16:13:36.268563 10.10.10.1:5080 -> 208.73.198.12:5060
ACK sip:208.73.198.12:5060 SIP/2.0'
Via: SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKXCZQia5D;rport'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
CSeq: 10 ACK'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
Contact: <sip:10.10.10.1:5080>'
Supported: replaces, path, timer, eventlist'
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE'
Content-Length: 0'
'


U 2013/03/30 16:13:40.267734 208.73.198.12:5060 -> 208.73.199.250:5060
SIP/2.0 200 OK'
Via: SIP/2.0/UDP 208.73.199.250;branch=z9hG4bK3642.4b4263c869b06a079cb58096dc4f3039.0,SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKPfekkaGS;rport=5080'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
CSeq: 10 INVITE'
Contact: sip:208.73.198.12:5060'
Content-Type: application/sdp'
Content-Length: 193'
'
v=0'
o=- 1 1 IN IP4 208.73.198.12'
s=SIP Call'
c=IN IP4 208.73.198.13'
t=0 0'
m=audio 49166 RTP/AVP 0 101'
a=ptime:20'
a=rtpmap:0 PCMU/8000'
a=rtpmap:101 telephone-event/8000'
a=fmtp:101 0-15'


U 2013/03/30 16:13:40.267945 10.10.10.1:5060 -> 10.10.10.1:5080
SIP/2.0 200 OK'
Via: SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKPfekkaGS;rport=5080'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
CSeq: 10 INVITE'
Contact: sip:208.73.198.12:5060'
Content-Type: application/sdp'
Content-Length: 193'
P-NGCP-Src-Ip: 208.73.198.12'
P-NGCP-Src-Port: 5060'
P-NGCP-Src-Proto: udp'
P-NGCP-Src-Af: 4'
'
v=0'
o=- 1 1 IN IP4 208.73.198.12'
s=SIP Call'
c=IN IP4 208.73.198.13'
t=0 0'
m=audio 49166 RTP/AVP 0 101'
a=ptime:20'
a=rtpmap:0 PCMU/8000'
a=rtpmap:101 telephone-event/8000'
a=fmtp:101 0-15'


U 2013/03/30 16:13:40.268020 10.10.10.1:5080 -> 208.73.198.12:5060
ACK sip:208.73.198.12:5060 SIP/2.0'
Via: SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKXCZQia5D;rport'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
CSeq: 10 ACK'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
Contact: <sip:10.10.10.1:5080>'
Supported: replaces, path, timer, eventlist'
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE'
Content-Length: 0'
'


U 2013/03/30 16:13:44.267108 208.73.198.12:5060 -> 208.73.199.250:5060
SIP/2.0 200 OK'
Via: SIP/2.0/UDP 208.73.199.250;branch=z9hG4bK3642.4b4263c869b06a079cb58096dc4f3039.0,SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKPfekkaGS;rport=5080'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
CSeq: 10 INVITE'
Contact: sip:208.73.198.12:5060'
Content-Type: application/sdp'
Content-Length: 193'
'
v=0'
o=- 1 1 IN IP4 208.73.198.12'
s=SIP Call'
c=IN IP4 208.73.198.13'
t=0 0'
m=audio 49166 RTP/AVP 0 101'
a=ptime:20'
a=rtpmap:0 PCMU/8000'
a=rtpmap:101 telephone-event/8000'
a=fmtp:101 0-15'


U 2013/03/30 16:13:44.267513 10.10.10.1:5060 -> 10.10.10.1:5080
SIP/2.0 200 OK'
Via: SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKPfekkaGS;rport=5080'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
CSeq: 10 INVITE'
Contact: sip:208.73.198.12:5060'
Content-Type: application/sdp'
Content-Length: 193'
P-NGCP-Src-Ip: 208.73.198.12'
P-NGCP-Src-Port: 5060'
P-NGCP-Src-Proto: udp'
P-NGCP-Src-Af: 4'
'
v=0'
o=- 1 1 IN IP4 208.73.198.12'
s=SIP Call'
c=IN IP4 208.73.198.13'
t=0 0'
m=audio 49166 RTP/AVP 0 101'
a=ptime:20'
a=rtpmap:0 PCMU/8000'
a=rtpmap:101 telephone-event/8000'
a=fmtp:101 0-15'


U 2013/03/30 16:13:44.267591 10.10.10.1:5080 -> 208.73.198.12:5060
ACK sip:208.73.198.12:5060 SIP/2.0'
Via: SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKXCZQia5D;rport'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
CSeq: 10 ACK'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
Contact: <sip:10.10.10.1:5080>'
Supported: replaces, path, timer, eventlist'
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE'
Content-Length: 0'
'


U 2013/03/30 16:13:48.266595 208.73.198.12:5060 -> 208.73.199.250:5060
SIP/2.0 200 OK'
Via: SIP/2.0/UDP 208.73.199.250;branch=z9hG4bK3642.4b4263c869b06a079cb58096dc4f3039.0,SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKPfekkaGS;rport=5080'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
CSeq: 10 INVITE'
Contact: sip:208.73.198.12:5060'
Content-Type: application/sdp'
Content-Length: 193'
'
v=0'
o=- 1 1 IN IP4 208.73.198.12'
s=SIP Call'
c=IN IP4 208.73.198.13'
t=0 0'
m=audio 49166 RTP/AVP 0 101'
a=ptime:20'
a=rtpmap:0 PCMU/8000'
a=rtpmap:101 telephone-event/8000'
a=fmtp:101 0-15'


U 2013/03/30 16:13:48.266776 10.10.10.1:5060 -> 10.10.10.1:5080
SIP/2.0 200 OK'
Via: SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKPfekkaGS;rport=5080'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
CSeq: 10 INVITE'
Contact: sip:208.73.198.12:5060'
Content-Type: application/sdp'
Content-Length: 193'
P-NGCP-Src-Ip: 208.73.198.12'
P-NGCP-Src-Port: 5060'
P-NGCP-Src-Proto: udp'
P-NGCP-Src-Af: 4'
'
v=0'
o=- 1 1 IN IP4 208.73.198.12'
s=SIP Call'
c=IN IP4 208.73.198.13'
t=0 0'
m=audio 49166 RTP/AVP 0 101'
a=ptime:20'
a=rtpmap:0 PCMU/8000'
a=rtpmap:101 telephone-event/8000'
a=fmtp:101 0-15'


U 2013/03/30 16:13:48.266852 10.10.10.1:5080 -> 208.73.198.12:5060
ACK sip:208.73.198.12:5060 SIP/2.0'
Via: SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKXCZQia5D;rport'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
CSeq: 10 ACK'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
Contact: <sip:10.10.10.1:5080>'
Supported: replaces, path, timer, eventlist'
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE'
Content-Length: 0'
'


U 2013/03/30 16:13:52.265933 208.73.198.12:5060 -> 208.73.199.250:5060
SIP/2.0 200 OK'
Via: SIP/2.0/UDP 208.73.199.250;branch=z9hG4bK3642.4b4263c869b06a079cb58096dc4f3039.0,SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKPfekkaGS;rport=5080'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
CSeq: 10 INVITE'
Contact: sip:208.73.198.12:5060'
Content-Type: application/sdp'
Content-Length: 193'
'
v=0'
o=- 1 1 IN IP4 208.73.198.12'
s=SIP Call'
c=IN IP4 208.73.198.13'
t=0 0'
m=audio 49166 RTP/AVP 0 101'
a=ptime:20'
a=rtpmap:0 PCMU/8000'
a=rtpmap:101 telephone-event/8000'
a=fmtp:101 0-15'


U 2013/03/30 16:13:52.266118 10.10.10.1:5060 -> 10.10.10.1:5080
SIP/2.0 200 OK'
Via: SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKPfekkaGS;rport=5080'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
CSeq: 10 INVITE'
Contact: sip:208.73.198.12:5060'
Content-Type: application/sdp'
Content-Length: 193'
P-NGCP-Src-Ip: 208.73.198.12'
P-NGCP-Src-Port: 5060'
P-NGCP-Src-Proto: udp'
P-NGCP-Src-Af: 4'
'
v=0'
o=- 1 1 IN IP4 208.73.198.12'
s=SIP Call'
c=IN IP4 208.73.198.13'
t=0 0'
m=audio 49166 RTP/AVP 0 101'
a=ptime:20'
a=rtpmap:0 PCMU/8000'
a=rtpmap:101 telephone-event/8000'
a=fmtp:101 0-15'


U 2013/03/30 16:13:52.266195 10.10.10.1:5080 -> 208.73.198.12:5060
ACK sip:208.73.198.12:5060 SIP/2.0'
Via: SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKXCZQia5D;rport'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
CSeq: 10 ACK'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
Contact: <sip:10.10.10.1:5080>'
Supported: replaces, path, timer, eventlist'
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE'
Content-Length: 0'
'


U 2013/03/30 16:13:56.265284 208.73.198.12:5060 -> 208.73.199.250:5060
SIP/2.0 200 OK'
Via: SIP/2.0/UDP 208.73.199.250;branch=z9hG4bK3642.4b4263c869b06a079cb58096dc4f3039.0,SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKPfekkaGS;rport=5080'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
CSeq: 10 INVITE'
Contact: sip:208.73.198.12:5060'
Content-Type: application/sdp'
Content-Length: 193'
'
v=0'
o=- 1 1 IN IP4 208.73.198.12'
s=SIP Call'
c=IN IP4 208.73.198.13'
t=0 0'
m=audio 49166 RTP/AVP 0 101'
a=ptime:20'
a=rtpmap:0 PCMU/8000'
a=rtpmap:101 telephone-event/8000'
a=fmtp:101 0-15'


U 2013/03/30 16:13:56.265671 10.10.10.1:5060 -> 10.10.10.1:5080
SIP/2.0 200 OK'
Via: SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKPfekkaGS;rport=5080'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
CSeq: 10 INVITE'
Contact: sip:208.73.198.12:5060'
Content-Type: application/sdp'
Content-Length: 193'
P-NGCP-Src-Ip: 208.73.198.12'
P-NGCP-Src-Port: 5060'
P-NGCP-Src-Proto: udp'
P-NGCP-Src-Af: 4'
'
v=0'
o=- 1 1 IN IP4 208.73.198.12'
s=SIP Call'
c=IN IP4 208.73.198.13'
t=0 0'
m=audio 49166 RTP/AVP 0 101'
a=ptime:20'
a=rtpmap:0 PCMU/8000'
a=rtpmap:101 telephone-event/8000'
a=fmtp:101 0-15'


U 2013/03/30 16:13:56.265748 10.10.10.1:5080 -> 208.73.198.12:5060
ACK sip:208.73.198.12:5060 SIP/2.0'
Via: SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKXCZQia5D;rport'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
CSeq: 10 ACK'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
Contact: <sip:10.10.10.1:5080>'
Supported: replaces, path, timer, eventlist'
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE'
Content-Length: 0'
'


U 2013/03/30 16:14:00.264695 208.73.198.12:5060 -> 208.73.199.250:5060
SIP/2.0 200 OK'
Via: SIP/2.0/UDP 208.73.199.250;branch=z9hG4bK3642.4b4263c869b06a079cb58096dc4f3039.0,SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKPfekkaGS;rport=5080'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
CSeq: 10 INVITE'
Contact: sip:208.73.198.12:5060'
Content-Type: application/sdp'
Content-Length: 193'
'
v=0'
o=- 1 1 IN IP4 208.73.198.12'
s=SIP Call'
c=IN IP4 208.73.198.13'
t=0 0'
m=audio 49166 RTP/AVP 0 101'
a=ptime:20'
a=rtpmap:0 PCMU/8000'
a=rtpmap:101 telephone-event/8000'
a=fmtp:101 0-15'


U 2013/03/30 16:14:00.265098 10.10.10.1:5060 -> 10.10.10.1:5080
SIP/2.0 200 OK'
Via: SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKPfekkaGS;rport=5080'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
CSeq: 10 INVITE'
Contact: sip:208.73.198.12:5060'
Content-Type: application/sdp'
Content-Length: 193'
P-NGCP-Src-Ip: 208.73.198.12'
P-NGCP-Src-Port: 5060'
P-NGCP-Src-Proto: udp'
P-NGCP-Src-Af: 4'
'
v=0'
o=- 1 1 IN IP4 208.73.198.12'
s=SIP Call'
c=IN IP4 208.73.198.13'
t=0 0'
m=audio 49166 RTP/AVP 0 101'
a=ptime:20'
a=rtpmap:0 PCMU/8000'
a=rtpmap:101 telephone-event/8000'
a=fmtp:101 0-15'


U 2013/03/30 16:14:00.265170 10.10.10.1:5080 -> 208.73.198.12:5060
ACK sip:208.73.198.12:5060 SIP/2.0'
Via: SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKXCZQia5D;rport'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
CSeq: 10 ACK'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
Contact: <sip:10.10.10.1:5080>'
Supported: replaces, path, timer, eventlist'
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE'
Content-Length: 0'
'


U 2013/03/30 16:14:00.767830 208.73.198.12:5060 -> 208.73.199.250:5060
BYE sip:ngcp-lb at 208.73.199.250:5060 SIP/2.0'
Via: SIP/2.0/UDP 208.73.198.12:5060;branch=z9hG4bK-00101-gMkh00ht1ojP39k-0'
Max-Forwards: 70'
From: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
To: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
CSeq: 101 BYE'
'


U 2013/03/30 16:14:00.768143 10.10.10.1:5060 -> 10.10.10.1:5062
BYE sip:sems at 10.10.10.1:5080 SIP/2.0'
Record-Route: <sip:10.10.10.1;r2=on;lr=on;ftag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12;ngcplb=yes>'
Record-Route: <sip:208.73.199.250;r2=on;lr=on;ftag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12;ngcplb=yes>'
Via: SIP/2.0/UDP 10.10.10.1;branch=z9hG4bK81c.5ddccd5449792063c5fb1379142f1fb5.0'
Via: SIP/2.0/UDP 208.73.198.12:5060;rport=5060;branch=z9hG4bK-00101-gMkh00ht1ojP39k-0'
Max-Forwards: 69'
From: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
To: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
CSeq: 101 BYE'
P-NGCP-Src-Ip: 208.73.198.12'
P-NGCP-Src-Port: 5060'
P-NGCP-Src-Proto: udp'
P-NGCP-Src-Af: 4'
'


U 2013/03/30 16:14:00.768242 10.10.10.1:5062 -> 10.10.10.1:5060
SIP/2.0 100 Trying'
Via: SIP/2.0/UDP 10.10.10.1;branch=z9hG4bK81c.5ddccd5449792063c5fb1379142f1fb5.0'
Via: SIP/2.0/UDP 208.73.198.12:5060;rport=5060;branch=z9hG4bK-00101-gMkh00ht1ojP39k-0'
From: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
To: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
CSeq: 101 BYE'
Server: Sipwise NGCP Proxy 2.X'
Content-Length: 0'
'


U 2013/03/30 16:14:00.768414 208.73.199.250:5060 -> 208.73.198.12:5060
SIP/2.0 100 Trying'
Via: SIP/2.0/UDP 208.73.198.12:5060;rport=5060;branch=z9hG4bK-00101-gMkh00ht1ojP39k-0'
From: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
To: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
CSeq: 101 BYE'
Server: Sipwise NGCP Proxy 2.X'
Content-Length: 0'
'


U 2013/03/30 16:14:00.768701 10.10.10.1:5062 -> 10.10.10.1:5060
SIP/2.0 403 Domain not served here'
Via: SIP/2.0/UDP 10.10.10.1;branch=z9hG4bK81c.5ddccd5449792063c5fb1379142f1fb5.0'
Via: SIP/2.0/UDP 208.73.198.12:5060;rport=5060;branch=z9hG4bK-00101-gMkh00ht1ojP39k-0'
From: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
To: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
CSeq: 101 BYE'
Server: Sipwise NGCP Proxy 2.X'
Content-Length: 0'
'


U 2013/03/30 16:14:00.768807 208.73.199.250:5060 -> 208.73.198.12:5060
SIP/2.0 403 Domain not served here'
Via: SIP/2.0/UDP 208.73.198.12:5060;rport=5060;branch=z9hG4bK-00101-gMkh00ht1ojP39k-0'
From: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
To: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
CSeq: 101 BYE'
Server: Sipwise NGCP Proxy 2.X'
Content-Length: 0'
'


U 2013/03/30 16:14:07.275128 173.218.47.226:5060 -> 208.73.199.250:5060
BYE sip:ngcp-lb at 208.73.199.250:5060;ngcpct='sip:10.10.10.1:5080' SIP/2.0'
Via: SIP/2.0/UDP 192.168.1.15:5060;branch=z9hG4bK635526613;rport'
Route: <sip:208.73.199.250;r2=on;lr=on;ftag=381199400;nat=yes;ngcplb=yes>'
Route: <sip:10.10.10.1;r2=on;lr=on;ftag=381199400;nat=yes;ngcplb=yes>'
Route: <sip:10.10.10.1:5062;lr=on;ftag=381199400;did=5b1.2f3;mpd=ii;rtpprx=yes;vsf=Yk1rc2RyRTlyYmdOdGFBa2diWXhiTWtzZHJFOXI->'
From: "4179730088" <sip:4179730088 at 208.73.199.250>;tag=381199400'
To: <sip:4172302354 at 208.73.199.250>;tag=747E2E0E-51575566000CD69C-903AE700'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF'
CSeq: 62 BYE'
Contact: <sip:4179730088 at 192.168.1.15:5060>'
Max-Forwards: 70'
Supported: replaces, path, timer, eventlist'
User-Agent: Grandstream GXV3140 1.0.7.50'
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE'
Content-Length: 0'
'


U 2013/03/30 16:14:07.275430 10.10.10.1:5060 -> 10.10.10.1:5062
BYE sip:10.10.10.1:5080 SIP/2.0'
Record-Route: <sip:10.10.10.1;r2=on;lr=on;ftag=381199400;nat=yes;ngcplb=yes>'
Record-Route: <sip:208.73.199.250;r2=on;lr=on;ftag=381199400;nat=yes;ngcplb=yes>'
Via: SIP/2.0/UDP 10.10.10.1;branch=z9hG4bK1b2b.a4e6174164c66d48841ed530f1f2498f.0'
Via: SIP/2.0/UDP 192.168.1.15:5060;received=173.218.47.226;branch=z9hG4bK635526613;rport=5060'
Route: <sip:10.10.10.1:5062;lr=on;ftag=381199400;did=5b1.2f3;mpd=ii;rtpprx=yes;vsf=Yk1rc2RyRTlyYmdOdGFBa2diWXhiTWtzZHJFOXI->'
From: "4179730088" <sip:4179730088 at 208.73.199.250>;tag=381199400'
To: <sip:4172302354 at 208.73.199.250>;tag=747E2E0E-51575566000CD69C-903AE700'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF'
CSeq: 62 BYE'
Contact: <sip:4179730088 at 192.168.1.15:5060;alias=173.218.47.226~5060~1>'
Max-Forwards: 69'
Supported: replaces, path, timer, eventlist'
User-Agent: Grandstream GXV3140 1.0.7.50'
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE'
Content-Length: 0'
P-NGCP-Src-Ip: 173.218.47.226'
P-NGCP-Src-Port: 5060'
P-NGCP-Src-Proto: udp'
P-NGCP-Src-Af: 4'
P-NGCP-Src-Nat: 1'
'


U 2013/03/30 16:14:07.275544 10.10.10.1:5062 -> 10.10.10.1:5060
SIP/2.0 100 Trying'
Via: SIP/2.0/UDP 10.10.10.1;branch=z9hG4bK1b2b.a4e6174164c66d48841ed530f1f2498f.0'
Via: SIP/2.0/UDP 192.168.1.15:5060;received=173.218.47.226;branch=z9hG4bK635526613;rport=5060'
From: "4179730088" <sip:4179730088 at 208.73.199.250>;tag=381199400'
To: <sip:4172302354 at 208.73.199.250>;tag=747E2E0E-51575566000CD69C-903AE700'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF'
CSeq: 62 BYE'
Server: Sipwise NGCP Proxy 2.X'
Content-Length: 0'
'


U 2013/03/30 16:14:07.275721 208.73.199.250:5060 -> 173.218.47.226:5060
SIP/2.0 100 Trying'
Via: SIP/2.0/UDP 192.168.1.15:5060;received=173.218.47.226;branch=z9hG4bK635526613;rport=5060'
From: "4179730088" <sip:4179730088 at 208.73.199.250>;tag=381199400'
To: <sip:4172302354 at 208.73.199.250>;tag=747E2E0E-51575566000CD69C-903AE700'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF'
CSeq: 62 BYE'
Server: Sipwise NGCP Proxy 2.X'
Content-Length: 0'
'


U 2013/03/30 16:14:07.276511 10.10.10.1:5080 -> 208.73.198.12:5060
BYE sip:208.73.198.12:5060 SIP/2.0'
Via: SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKbxucua~d;rport'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
CSeq: 11 BYE'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
Content-Length: 0'
'


U 2013/03/30 16:14:07.276517 10.10.10.1:5062 -> 10.10.10.1:5060
SIP/2.0 200 OK'
Record-Route: <sip:10.10.10.1:5062;lr=on;ftag=381199400;rtpprx=yes>'
Record-Route: <sip:10.10.10.1;r2=on;lr=on;ftag=381199400;nat=yes;ngcplb=yes>'
Record-Route: <sip:208.73.199.250;r2=on;lr=on;ftag=381199400;nat=yes;ngcplb=yes>'
Via: SIP/2.0/UDP 10.10.10.1;branch=z9hG4bK1b2b.a4e6174164c66d48841ed530f1f2498f.0'
Via: SIP/2.0/UDP 192.168.1.15:5060;received=173.218.47.226;branch=z9hG4bK635526613;rport=5060'
From: "4179730088" <sip:4179730088 at 208.73.199.250>;tag=381199400'
To: <sip:4172302354 at 208.73.199.250>;tag=747E2E0E-51575566000CD69C-903AE700'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF'
CSeq: 62 BYE'
Server: Sipwise NGCP Application Server'
Content-Length: 0'
'


U 2013/03/30 16:14:07.276866 208.73.199.250:5060 -> 173.218.47.226:5060
SIP/2.0 200 OK'
Record-Route: <sip:10.10.10.1:5062;lr=on;ftag=381199400;rtpprx=yes>'
Record-Route: <sip:10.10.10.1;r2=on;lr=on;ftag=381199400;nat=yes;ngcplb=yes>'
Record-Route: <sip:208.73.199.250;r2=on;lr=on;ftag=381199400;nat=yes;ngcplb=yes>'
Via: SIP/2.0/UDP 192.168.1.15:5060;received=173.218.47.226;branch=z9hG4bK635526613;rport=5060'
From: "4179730088" <sip:4179730088 at 208.73.199.250>;tag=381199400'
To: <sip:4172302354 at 208.73.199.250>;tag=747E2E0E-51575566000CD69C-903AE700'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF'
CSeq: 62 BYE'
Server: Sipwise NGCP Application Server'
Content-Length: 0'
'


U 2013/03/30 16:14:07.763044 10.10.10.1:5080 -> 208.73.198.12:5060
BYE sip:208.73.198.12:5060 SIP/2.0'
Via: SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKbxucua~d;rport'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
CSeq: 11 BYE'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
Content-Length: 0'
'


U 2013/03/30 16:14:08.763043 10.10.10.1:5080 -> 208.73.198.12:5060
BYE sip:208.73.198.12:5060 SIP/2.0'
Via: SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKbxucua~d;rport'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
CSeq: 11 BYE'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
Content-Length: 0'
'


U 2013/03/30 16:14:12.763043 10.10.10.1:5080 -> 208.73.198.12:5060
BYE sip:208.73.198.12:5060 SIP/2.0'
Via: SIP/2.0/UDP 10.10.10.1:5080;branch=z9hG4bKbxucua~d;rport'
From: <sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>;tag=gNkh00ht1ojP39p-IPTrunk-127-17-20at208.73.198.12'
CSeq: 11 BYE'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
Content-Length: 0'
'


From skchopperguy at gmail.com  Sun Mar 31 13:33:14 2013
From: skchopperguy at gmail.com (Skyler)
Date: Sun, 31 Mar 2013 10:33:14 -0700
Subject: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
In-Reply-To: <3eb26bda-cede-43bc-a8f0-20c907977c71@usbsnpzimbra01v.servers.rystec.com>
References: <3eb26bda-cede-43bc-a8f0-20c907977c71@usbsnpzimbra01v.servers.rystec.com>
Message-ID: <5158735A.1030308@gmail.com>

Hi,

  Seems your call is dropping due to confused signalling. You should not 
need to input the 10.10.10.1 into your Taqua gw control list. One thing 
I see which does not look correct is the 100 trying response after the 
initial invite (spce > peer). as below:

U 2013/03/30 16:13:10.848816 208.73.198.12:5060 -> 208.73.199.250:5060
SIP/2.0 100 Trying'
Via: SIP/2.0/UDP 
208.73.199.250;branch=z9hG4bK3642.4b4263c869b06a079cb58096dc4f3039.0,SIP/2.0/UDP 
10.10.10.1:5080;branch=z9hG4bKPfekkaGS;rport=5080'
From: 
<sip:4179730088 at 208.73.199.250>;tag=57FE8F4B-51575566000CD752-90AB5700'
To: <sip:4172302354 at 208.73.198.12>'
Call-ID: 1454999203-5060-7 at BJC.BGI.B.BF_b2b-1'
CSeq: 10 INVITE'
Content-Length: 0'
'

  Notice this first response from the peer contains a single Via having 
'SIP/2.0/UDP' twice. As though (assuming) the Taqua gw is appending the 
ngcpct= to the via. This could be a case of needing to code the quirks 
of this Taqua into spce to overcome its (Taqua) needs for signalling.

What is your spce version?
Latest updates?
Do you have any customtt.tt2 files in your lb or proxy?

  My first inclination in troubleshooting is to rule out any individual 
customtt files which may have accidentally broken something in spce. Try 
moving customtt files to /tmp then ngcpcfg apply, do a test-call again 
to see if (or which) customtt may be causing this.

  This is only my suggestion, I cannot duplicate or test this behavior.

Skyler

On 3/30/2013 2:22 PM, Marc Rys wrote:
> Sorry for the delay.  Attached is the file as requested.
>
> Marc
>
> ----- Original Message -----
> From: "Skyler" <skchopperguy at gmail.com>
> To: spce-user at lists.sipwise.com
> Sent: Wednesday, March 27, 2013 3:05:30 PM
> Subject: Re: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
>
> I would run..
>
> ngrep-sip b -d any -qt -W byline port 5060 >/tmp/ngcpct_check.txt
>
> and make a test call. Then attach the txt to the list.
>
> Skyler
>
> On 3/27/2013 10:30 AM, Marc Rys wrote:
>> Ok, Im not sure if this is related, but I've been having struggles with our Taqua Gateway peering to SPCE.  OUr Gateway kept replying back with a 403 Forbidden whenever SPCE tried to send a call to it.
>>
>> After seeing your dialog about the contact headers, i looked at one our wireshark captures and our Invites were showing the same thing...
>> sip:ngcp-lb at your.ext.ip.address:5060;ngcpct='sip:127.0.0.1:5080'
>>
>> I followed the instructions for the audiocodes fix and changed our dummy address to 10.10.10.1  On our Taqua Gateway I had to add the dummy IP to the IP control list.. (10.10.10.1:5080) and now the Taqua accepts calls from SPCE.
>>
>> I'm not sure if this is solution for you to use with the other providers listed in the thread.
>>
>> In my attached wireshark cap, it appears that SPCE is sending an ACK back from my dummy address 10.10.10.1 to my gateway.  Thats not right is it?  at this point now, my calls to our gateway dont stay up for longer than 32Seconds then the audio drops out of the call and the person on the PSTN side of the call is disconnected completely, and my SPCE UA appears to have the call up still with no audio until I end the call.
>>
>>
>>
>> Marc Rys
>> http://www.tri-lakes.net
>> http://www.rystec.com
>>
>> ----- Original Message -----
>> From: "Skyler" <skchopperguy at gmail.com>
>> To: "Gary Nieboer" <gary at netdiverse.com>
>> Cc: spce-user at lists.sipwise.com
>> Sent: Wednesday, March 27, 2013 11:48:44 AM
>> Subject: Re: [Spce-user] ngcpct='sip:127.0.0.1:5080' in Contact header
>>
>> My guess (as I've not actually had this issue) is that your contact
>> would look something like:
>>
>> Contact: <sip:ngcp-lb at your.ext.ip.address:5060;ngcpct='sip:127.0.0.1:5080'>
>>
>> Skyler
>>
>> On 3/27/2013 9:42 AM, Gary Nieboer wrote:
>>> I will give a couple of the items advised a try.
>>>
>>> Should there be more than one uri in the Contact field?  If so, that may
>>> be the quick fix.  The contact field I captured a few weeks ago only has
>>> one, and it is the loopback/proxy address of 127.0.0.1:5080
>>> <http://127.0.0.1:5080>
>>>
>>> -Gary
>>> --
>>> Gary Nieboer
>>>
>>>
>>> 877-763-1014 Toll Free  |  801-657-5484 Utah  |  615-800-6946 Tennessee
>>> | 801-694-6106 Mobile
>>> gary at netdiverse.com <mailto:gary at netdiverse.com>  | www.netdiverse.com
>>> <http://www.netdiverse.com/>
>>>
>>>
>>> On Wed, Mar 27, 2013 at 10:31 AM, Skyler <skchopperguy at gmail.com
>>> <mailto:skchopperguy at gmail.com>> wrote:
>>>
>>>       That makes sense, also might just be that the ngcpct sip uri is the
>>>       last uri read in the contact.
>>>
>>>         Maybe Sylvester and Gary could test this out as they have the OP.
>>>
>>>       What would a good test look like to base64-encode that ngcpct in a
>>>       customtt.tt2?
>>>
>>>       Skyler
>>>
>>>
>>>       On 3/27/2013 9:07 AM, Andreas Granig wrote:
>>>
>>>           Hi,
>>>
>>>           On 03/27/2013 04:55 PM, Skyler wrote:
>>>
>>>                  I created an lo:0 with a private IP address
>>>               (10.10.xxx.xxx) for lb >
>>>               b2buas: & proxies > sip:. That may help in this case as its
>>>               likely the
>>>               127.0.0.1 which is confusing the upstream proxy.
>>>
>>>
>>>           I've heard of endpoints disliking the SIP uri with the quotes...
>>>           so one
>>>           option could be to convert the "ngcpct='sip:127.0.0.1:5080
>>>           <http://127.0.0.1:5080>'" and
>>>           base64-encode the "sip:127.0.0.1:5080 <http://127.0.0.1:5080>",
>>>           so it gets something like
>>>           "ngcpct=__c2lwOjEyNy4wLjAuMTo1MDgwCg==" instead.
>>>
>>>           Andreas
>>>
>>>
>>>       _________________________________________________
>>>       Spce-user mailing list
>>>       Spce-user at lists.sipwise.com <mailto:Spce-user at lists.sipwise.com>
>>>       http://lists.sipwise.com/__listinfo/spce-user
>>>       <http://lists.sipwise.com/listinfo/spce-user>
>>>
>>>
>>
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/listinfo/spce-user
>>
>>
>>
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/listinfo/spce-user
>>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
>
>
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
>