[Spce-user] Intro

Mike Hammett sipwise at ics-il.net
Tue May 7 17:17:23 EDT 2013


As of now, I am just expecting the SIPwise voicemail to answer the call. 

I did enter some Vitelity specific rewrite rules from Jeremy Chism. 

Previously, I saw a 404 in the ngcp-sip command, but I'll run it again (not sure what may have changed) and attach that log. 




----- 
Mike Hammett 
Intelligent Computing Solutions 
http://www.ics-il.com 

----- Original Message -----

From: "Kevin Masse" <kmasse at questblue.com> 
To: "Mike Hammett" <sipwise at ics-il.net> 
Cc: spce-user at lists.sipwise.com 
Sent: Tuesday, May 7, 2013 3:52:03 PM 
Subject: Re: [Spce-user] Intro 



I am not looking at any logs but my guess is that vitelity is sending the 1 and you need to strip it. 


Are you sending this to a PBX? If so run the CLI on the PBX and see if you are receiving the 1 


Watch the CLI or do ngrep-sip on the sipwise 


Kevin 










Sent from my Verizon Wireless 4G LTE DROID 

Mike Hammett <sipwise at ics-il.net> wrote: 

I did change it over to IP authentication. The inbound calls hit my SIPwise box, but it rejects them. I don't have any devices associated with it, but I do have VM setup. Ideas? 

BTW: Maybe a beginner\quick setup guide and then a more advanced manual or set of wiki articles for the more advanced topics? The basics should include what I have to do to connect to origination and termination peers, provision a client device and have incoming calls route to that client and their voicemail. 



----- 
Mike Hammett 
Intelligent Computing Solutions 
http://www.ics-il.com 

----- Original Message ----- 
From: "Mike Hammett" <sipwise at ics-il.net> 
To: spce-user at lists.sipwise.com 
Sent: Monday, May 6, 2013 7:01:18 AM 
Subject: Re: [Spce-user] Intro 

===== 
A call to your DID 815981XXXX from 815739XXXX has failed at 5:48am on 05/06/2013 MDT. We received 'CONGESTION' when attempting to route the call to your server or device. This number is configured to route to the peer 65.182.XXX.XXX. 

This error usually means your server or device does not recognize the number being dialed. If using asterisk, make sure you have the correct inbound context specified on your inbound trunk and that you have correctly added an inbound route/extension logic for this DID. 
===== 

So apparently whatever I did is wrong. 



----- 
Mike Hammett 
Intelligent Computing Solutions 
http://www.ics-il.com 

----- Original Message ----- 
From: "Mike Hammett" <sipwise at ics-il.net> 
To: spce-user at lists.sipwise.com 
Sent: Monday, May 6, 2013 6:23:41 AM 
Subject: Re: [Spce-user] Intro 

I did see where to change to IP based authentication and did so. Why is IP authentication preferred over user authentication? Well, other than it is more complicated to do this with SIPWise. 

I haven't tested it yet, but I made two peer groups, one for inbound and one for outbound. I didn't create any peering rules for the inbound group. 



----- 
Mike Hammett 
Intelligent Computing Solutions 
http://www.ics-il.com 

----- Original Message ----- 
From: "Kevin Masse" <kmasse at questblue.com> 
To: "Skyler" <skchopperguy at gmail.com>, spce-user at lists.sipwise.com 
Sent: Monday, May 6, 2013 6:08:15 AM 
Subject: Re: [Spce-user] Intro 

Good morning, I noticed something that needs to be addressed when using Vitelity. 

>>> Proxy: sip28.vitelity.net ( 66.241.99.27 ) 
>>> Outbound Proxy: outbound.vitelity.net 

Not all accounts with Vitelity will use the same inbound and outbound settings. 
Vitelity may send you traffic from any of the following 
64.2.142.0/24 
66.241.99.0/24 

If you are using settings provided by another Vitelity user it may not be the same as yours. Make sure you know what your provisioned for with Vitelity and use those settings. 

Additionally it is not suggested to use registration with providers. If you can avoid it, use IP Authentication. In the Vitelity portal go to the support link, then click on sub accounts. You will be able to setup your IP there for IP Authentication. 

Thanks, 
Kevin 



-----Original Message----- 
From: spce-user-bounces at lists.sipwise.com [mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of Skyler 
Sent: Monday, May 06, 2013 4:14 AM 
To: spce-user at lists.sipwise.com 
Subject: Re: [Spce-user] Intro 

Woops, apparently I need sleep. sry. 

On 5/5/2013 9:53 PM, Skyler wrote: 
> Jeremie, 
> 
> Do you also have user/pass on your vitality peer? 
> 
> "I'm not sure how I'm supposed to put this into Sipwise." 
> 
> 
> 
> On 5/5/2013 6:10 PM, Jeremie Chism wrote: 
>> You enter the information into sip peers. I use vitality as one of my 
>> upstreams. I think we have 5 or so now. If you use the accept all 
>> rule for the peer it will allow any call to come in. Rewrite rules 
>> are used to normalize incoming calls to e164 format. A rewrite rule 
>> can be used for many things. Ex. If you want to use *98 for voicemail 
>> instead of the default 2000 a rewrite rule will do that. The users 
>> manual is pretty detailed for peers. The only thing you may need help 
>> for is the rewrite rules. 
>> 
>> Sent from my iPhone 
>> 
>> On May 5, 2013, at 7:35 PM, Mike Hammett <sipwise at ics-il.net> wrote: 
>> 
>>> Little of the peering section makes sense to me. One of my upstreams 
>>> has given me the following information: 
>>> 
>>> Proxy: sip28.vitelity.net ( 66.241.99.27 ) 
>>> Outbound Proxy: outbound.vitelity.net 
>>> Login: XXXXXXXXXX 
>>> Password: XXXXXXXXXX 
>>> 
>>> I'm not sure how I'm supposed to put this into Sipwise. I want to be 
>>> able to receive calls from this provider as well as send them calls. 
>>> As I add more providers, I want to be able to receive calls from 
>>> them and send my outbound to them. 
>>> 
>>> I see there was a lot of attention paid to the Rewrite Rule Set 
>>> section, of which I'm not sure why you even need that. Just have a 
>>> check box on each client (Force outbound caller-ID to DID). Done, or 
>>> not. 
>>> 
>>> 
>>> 
>>> ----- 
>>> Mike Hammett 
>>> Intelligent Computing Solutions 
>>> http://www.ics-il.com 
>>> 
>>> ----- Original Message ----- 
>>> From: "Jon Bonilla" <jbonilla at sipwise.com> 
>>> To: spce-user at lists.sipwise.com 
>>> Sent: Thursday, May 2, 2013 5:10:06 AM 
>>> Subject: Re: [Spce-user] Intro 
>>> 
>>> El Sun, 28 Apr 2013 16:38:02 -0500 (CDT) sipwise at ics-il.net 
>>> escribió: 
>>> 
>>>> I'm just setting it up now. I've used Asterisk (pre-1.2 aka flat 
>>>> files, no gui help) and a few Asterisk based systems over the 
>>>> years, so I at least have the basics of VoIP down. The install was 
>>>> much more pleasant than PBX in a Flash. That asks a million 
>>>> unnecessary questions and takes forever. 
>>>> 
>>>> Thus far my commentary on the setup is thorough, but very confusing. 
>>>> Very 
>>>> little of the terminology used makes sense to me. I am forging 
>>>> ahead through the setup, but I'm not sure what I'm doing or how the 
>>>> pieces fit together. 
>>>> Maybe I'll understand better once it's running, but for now... I'm 
>>>> lost. 
>>>> 
>>>> 
>>> 
>>> Well, the installation process is quite unattended as you've seen. 
>>> The Handbook 
>>> gives you an overview of what you have but also a quick start guide. 
>>> Follow it 
>>> and try to understand the terms behind it. Don't try to think in 
>>> "pbx" terms because the spce is not a pbx or an asterisk system. 
>>> 
>>> Ask here when you get stucked. And you can always ask for a personal 
>>> training/webinar to our sales team. We're organizing trainings in 
>>> several countries too. 
>>> 
>>> cheers, 
>>> 
>>> Jon 
>>> 
>>> 
>>> _______________________________________________ 
>>> 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 
>> 

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#
U 2013/05/07 16:13:29.502382 66.241.XXX.XXX:5060 -> 65.182.XXX.XXX:5060
INVITE sip:815981XXXX at 65.182.XXX.XXX:5060 SIP/2.0'
Via: SIP/2.0/UDP 66.241.XXX.XXX:5060;branch=z9hG4bK0992b6f9;rport'
From: "MIKE HAMMETT" <sip:815739XXXX at 66.241.XXX.XXX>;tag=as62df844a'
To: <sip:815981XXXX at 65.182.XXX.XXX:5060>'
Contact: <sip:815739XXXX at 66.241.XXX.XXX>'
Call-ID: 4f8b5a9d1a959a404b053cbe5fa9cd2f at 66.241.XXX.XXX'
CSeq: 102 INVITE'
User-Agent: packetrino'
Max-Forwards: 70'
Date: Tue, 07 May 2013 21:13:29 GMT'
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO'
Supported: replaces'
Content-Type: application/sdp'
Content-Length: 334'
'
v=0'
o=root 13643 13643 IN IP4 66.241.XXX.XXX'
s=session'
c=IN IP4 66.241.XXX.XXX'
t=0 0'
m=audio 19210 RTP/AVP 0 8 3 18 101'
a=rtpmap:0 PCMU/8000'
a=rtpmap:8 PCMA/8000'
a=rtpmap:3 GSM/8000'
a=rtpmap:18 G729/8000'
a=fmtp:18 annexb=no'
a=rtpmap:101 telephone-event/8000'
a=fmtp:101 0-16'
a=silenceSupp:off - - - -'
a=ptime:20'
a=sendrecv'

#
U 2013/05/07 16:13:29.506010 127.0.0.1:5060 -> 127.0.0.1:5062
INVITE sip:815981XXXX at 65.182.XXX.XXX:5060 SIP/2.0'
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as62df844a;ngcplb=yes>'
Record-Route: <sip:65.182.XXX.XXX;r2=on;lr=on;ftag=as62df844a;ngcplb=yes>'
Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK4145.ca414f6aa7da16d423e8c13a2a0fca2c.0'
Via: SIP/2.0/UDP 66.241.XXX.XXX:5060;branch=z9hG4bK0992b6f9;rport=5060'
From: "MIKE HAMMETT" <sip:815739XXXX at 66.241.XXX.XXX>;tag=as62df844a'
To: <sip:815981XXXX at 65.182.XXX.XXX:5060>'
Contact: <sip:815739XXXX at 66.241.XXX.XXX>'
Call-ID: 4f8b5a9d1a959a404b053cbe5fa9cd2f at 66.241.XXX.XXX'
CSeq: 102 INVITE'
User-Agent: packetrino'
Max-Forwards: 69'
Date: Tue, 07 May 2013 21:13:29 GMT'
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO'
Supported: replaces'
Content-Type: application/sdp'
Content-Length: 334'
P-NGCP-Src-Ip: 66.241.XXX.XXX'
P-NGCP-Src-Port: 5060'
P-NGCP-Src-Proto: udp'
P-NGCP-Src-Af: 4'
P-Sock-Info: udp:65.182.XXX.XXX:5060'
'
v=0'
o=root 13643 13643 IN IP4 66.241.XXX.XXX'
s=session'
c=IN IP4 66.241.XXX.XXX'
t=0 0'
m=audio 19210 RTP/AVP 0 8 3 18 101'
a=rtpmap:0 PCMU/8000'
a=rtpmap:8 PCMA/8000'
a=rtpmap:3 GSM/8000'
a=rtpmap:18 G729/8000'
a=fmtp:18 annexb=no'
a=rtpmap:101 telephone-event/8000'
a=fmtp:101 0-16'
a=silenceSupp:off - - - -'
a=ptime:20'
a=sendrecv'

#
U 2013/05/07 16:13:29.508005 127.0.0.1:5062 -> 127.0.0.1:5060
SIP/2.0 100 Trying'
Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK4145.ca414f6aa7da16d423e8c13a2a0fca2c.0'
Via: SIP/2.0/UDP 66.241.XXX.XXX:5060;branch=z9hG4bK0992b6f9;rport=5060'
From: "MIKE HAMMETT" <sip:815739XXXX at 66.241.XXX.XXX>;tag=as62df844a'
To: <sip:815981XXXX at 65.182.XXX.XXX:5060>'
Call-ID: 4f8b5a9d1a959a404b053cbe5fa9cd2f at 66.241.XXX.XXX'
CSeq: 102 INVITE'
P-Out-Socket: udp:65.182.XXX.XXX:5060'
Server: Sipwise NGCP Proxy 2.X'
Content-Length: 0'
'

#
U 2013/05/07 16:13:29.509982 65.182.XXX.XXX:5060 -> 66.241.XXX.XXX:5060
SIP/2.0 100 Trying'
Via: SIP/2.0/UDP 66.241.XXX.XXX:5060;branch=z9hG4bK0992b6f9;rport=5060'
From: "MIKE HAMMETT" <sip:815739XXXX at 66.241.XXX.XXX>;tag=as62df844a'
To: <sip:815981XXXX at 65.182.XXX.XXX:5060>'
Call-ID: 4f8b5a9d1a959a404b053cbe5fa9cd2f at 66.241.XXX.XXX'
CSeq: 102 INVITE'
Server: Sipwise NGCP Proxy 2.X'
Content-Length: 0'
'

#
U 2013/05/07 16:13:29.545677 127.0.0.1:5062 -> 127.0.0.1:5060
SIP/2.0 404 Not Found'
Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK4145.ca414f6aa7da16d423e8c13a2a0fca2c.0'
Via: SIP/2.0/UDP 66.241.XXX.XXX:5060;branch=z9hG4bK0992b6f9;rport=5060'
From: "MIKE HAMMETT" <sip:815739XXXX at 66.241.XXX.XXX>;tag=as62df844a'
To: <sip:815981XXXX at 65.182.XXX.XXX:5060>;tag=f3067022b00c564156251ba2f28f331f-7953'
Call-ID: 4f8b5a9d1a959a404b053cbe5fa9cd2f at 66.241.XXX.XXX'
CSeq: 102 INVITE'
P-Out-Socket: udp:65.182.XXX.XXX:5060'
Server: Sipwise NGCP Proxy 2.X'
Content-Length: 0'
'

#
U 2013/05/07 16:13:29.546007 65.182.XXX.XXX:5060 -> 66.241.XXX.XXX:5060
SIP/2.0 404 Not Found'
Via: SIP/2.0/UDP 66.241.XXX.XXX:5060;branch=z9hG4bK0992b6f9;rport=5060'
From: "MIKE HAMMETT" <sip:815739XXXX at 66.241.XXX.XXX>;tag=as62df844a'
To: <sip:815981XXXX at 65.182.XXX.XXX:5060>;tag=f3067022b00c564156251ba2f28f331f-7953'
Call-ID: 4f8b5a9d1a959a404b053cbe5fa9cd2f at 66.241.XXX.XXX'
CSeq: 102 INVITE'
Server: Sipwise NGCP Proxy 2.X'
Content-Length: 0'
'

#
U 2013/05/07 16:13:29.590091 66.241.XXX.XXX:5060 -> 65.182.XXX.XXX:5060
ACK sip:815981XXXX at 65.182.XXX.XXX:5060 SIP/2.0'
Via: SIP/2.0/UDP 66.241.XXX.XXX:5060;branch=z9hG4bK0992b6f9;rport'
From: "MIKE HAMMETT" <sip:815739XXXX at 66.241.XXX.XXX>;tag=as62df844a'
To: <sip:815981XXXX at 65.182.XXX.XXX:5060>;tag=f3067022b00c564156251ba2f28f331f-7953'
Contact: <sip:815739XXXX at 66.241.XXX.XXX>'
Call-ID: 4f8b5a9d1a959a404b053cbe5fa9cd2f at 66.241.XXX.XXX'
CSeq: 102 ACK'
User-Agent: packetrino'
Max-Forwards: 70'
Content-Length: 0'
'

#
U 2013/05/07 16:13:29.594224 127.0.0.1:5060 -> 127.0.0.1:5062
ACK sip:815981XXXX at 65.182.XXX.XXX:5060 SIP/2.0'
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as62df844a;ngcplb=yes>'
Record-Route: <sip:65.182.XXX.XXX;r2=on;lr=on;ftag=as62df844a;ngcplb=yes>'
Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK4145.ca414f6aa7da16d423e8c13a2a0fca2c.0'
Via: SIP/2.0/UDP 66.241.XXX.XXX:5060;branch=z9hG4bK0992b6f9;rport=5060'
From: "MIKE HAMMETT" <sip:815739XXXX at 66.241.XXX.XXX>;tag=as62df844a'
To: <sip:815981XXXX at 65.182.XXX.XXX:5060>;tag=f3067022b00c564156251ba2f28f331f-7953'
Contact: <sip:815739XXXX at 66.241.XXX.XXX>'
Call-ID: 4f8b5a9d1a959a404b053cbe5fa9cd2f at 66.241.XXX.XXX'
CSeq: 102 ACK'
User-Agent: packetrino'
Max-Forwards: 69'
Content-Length: 0'
P-NGCP-Src-Ip: 66.241.XXX.XXX'
P-NGCP-Src-Port: 5060'
P-NGCP-Src-Proto: udp'
P-NGCP-Src-Af: 4'
P-Sock-Info: udp:65.182.XXX.XXX:5060'
'


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