[Spce-user] Intro

Skyler skchopperguy at gmail.com
Wed May 8 23:39:43 EDT 2013 

Just double checked here and re-created the same results you are 
experiencing. I'm talking about the SIP URI field of the subscriber.

When you created the subscriber you entered a8159814493 into the SIP URI 
field. You can verify this right now by looking at the VoIP Account. It 
lists the subscribers. You will see a8159814493 at voip1.ics-il.net which 
is the problem.

Now you must delete/terminate that subscriber and recreate a new one. 
Cannot edit the existing SIP URI otherwise.

So delete/terminate, recreate a new one. Test again, keeping the 
'Inbound Vitelity 3' rewrite rule under (each) Domain Preferences.

Report back.

Skyler

On 5/8/2013 7:58 PM, Jeremie Chism wrote:
> He is talking about the e164 area. This must be the actual phone number.  The username can be whatever.
>
> Sent from my iPhone
>
> On May 8, 2013, at 9:50 PM, Mike Hammett <sipwise at ics-il.net> wrote:
>
>> Right, just following what the manual told me to do.
>>
>>
>>
>> -----
>> Mike Hammett
>> Intelligent Computing Solutions
>> http://www.ics-il.com
>>
>> ----- Original Message -----
>> From: "Skyler" <skchopperguy at gmail.com>
>> To: spce-user at lists.sipwise.com
>> Sent: Wednesday, May 8, 2013 6:56:33 PM
>> Subject: Re: [Spce-user] Intro
>>
>> CC = country code. Are there letters in a phone number? No.
>> AC = area code. 3-digits for north america.
>> SN = subscriber number. 7-digits for north america.
>>
>> No wonder spce is 404'ing your subscriber. So many frustrated words just
>> left my mouth.
>>
>> You have to help us help you man.
>>
>> Skyler
>>
>> On 5/8/2013 4:46 PM, Skyler wrote:
>>> Wait a second. Hold the phone.
>>>
>>> The Subscriber Preferences.pdf you sent shows
>>>
>>> a8159814493 at voip1.ics-il.net
>>>
>>> Why is there an "a" in your subscriber number?
>>>
>>> Delete that man.
>>>
>>> Skyler
>>>
>>> On 5/8/2013 4:25 PM, Skyler wrote:
>>>> Wow. Alright. Need to see your proxy log then.
>>>>
>>>> Prepare for an inbound test call, just before you 'send' execute:
>>>>
>>>> tail -f /var/log/ngcp/kamailio-proxy.log >
>>>> /tmp/vitelity-inbound-proxy-log.txt
>>>>
>>>> hangup the failed call then CTRL+C the terminal.
>>>>
>>>> Prepare for an inbound test call, just before you 'send' execute:
>>>>
>>>> tail -f /var/log/ngcp/kamailio-lb.log > /tmp/vitelity-inbound-lb-log.txt
>>>>
>>>> hangup the failed call then CTRL+C the terminal.
>>>>
>>>> Send vitelity-inbound-proxy-log.txt and vitelity-inbound-lb-log.txt to
>>>> the list.
>>>>
>>>> Skyler
>>>>
>>>> On 5/8/2013 2:40 PM, Mike Hammett wrote:
>>>>> No dice.
>>>>>
>>>>>
>>>>>
>>>>> -----
>>>>> Mike Hammett
>>>>> Intelligent Computing Solutions
>>>>> http://www.ics-il.com
>>>>>
>>>>> ----- Original Message -----
>>>>> From: "Skyler" <skchopperguy at gmail.com>
>>>>> To: spce-user at lists.sipwise.com
>>>>> Sent: Wednesday, May 8, 2013 2:22:24 PM
>>>>> Subject: Re: [Spce-user] Intro
>>>>>
>>>>> ok, so its 404'ing before loading the rewrite.
>>>>>
>>>>> 1. Set 'Inbound Vitelity 3' rewrite rule under (each) Domain
>>>>> Preferences.
>>>>> 2. Unset rewrite rule for the subscriber(s) and peer(s).
>>>>>
>>>>> This should apply the rewrite to all INVITE(s) _prior_ to the start of
>>>>> processing the call. So the '1' will be added and proxy will find the
>>>>> right subscriber.
>>>>>
>>>>> Skyler
>>>>>
>>>>> On 5/8/2013 11:37 AM, Mike Hammett wrote:
>>>>>> No dice and here's my results.
>>>>>>
>>>>>>
>>>>>>
>>>>>> -----
>>>>>> Mike Hammett
>>>>>> Intelligent Computing Solutions
>>>>>> http://www.ics-il.com
>>>>>>
>>>>>> ----- Original Message -----
>>>>>> From: "Skyler" <skchopperguy at gmail.com>
>>>>>> To: spce-user at lists.sipwise.com
>>>>>> Sent: Wednesday, May 8, 2013 11:45:26 AM
>>>>>> Subject: Re: [Spce-user] Intro
>>>>>>
>>>>>> Mike,
>>>>>>
>>>>>>     No there isn't. Best thing is to:
>>>>>>
>>>>>> ngrep-sip b -d any -qt -W byline port 5060 > /tmp/vitelity-problem.txt
>>>>>>
>>>>>> Then email that to the list.
>>>>>>
>>>>>> BTW, looking at your congestion.txt and seeing the other suggestions to
>>>>>> re-write rule(s). Seems you are trying to strip the leading '1', but
>>>>>> from what I see, you need to add a '1'. Like so:
>>>>>>
>>>>>> ^([2-9][0-9]+)$   1\1   10-digits to e164
>>>>>>
>>>>>> Put the above into your rewrite rule set, then ensure the set is
>>>>>> enabled
>>>>>> for the subscriber AND the Vitelity peer(s).
>>>>>>
>>>>>> Once confirmed, run a test call while running the ngrep above and send
>>>>>> to the list.
>>>>>>
>>>>>> --Skyler
>>>>>>
>>>>>> On 5/8/2013 9:02 AM, Mike Hammett wrote:
>>>>>>> That didn't pan out for me either.
>>>>>>>
>>>>>>> Is there some sort of debugging that has more verbosity than ngcp-sip?
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> -----
>>>>>>> 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>, spce-user at lists.sipwise.com
>>>>>>> Sent: Tuesday, May 7, 2013 4:31:15 PM
>>>>>>> Subject: RE: [Spce-user] Intro
>>>>>>>
>>>>>>> Mike, try to verify this:
>>>>>>>
>>>>>>> Do you have a Peering Server for Vitelity
>>>>>>> You should have a Peering Server Vitel-Inbound with an IP that you
>>>>>>> are instructed to use from Vitelity
>>>>>>>
>>>>>>> Then a Peering Rule under that Peering Server
>>>>>>> In this area just put a 1 in the Callee Prefix nothing in Callee
>>>>>>> Pattern or Caller Pattern
>>>>>>>
>>>>>>> Then in REWRITE RULES
>>>>>>>
>>>>>>> Set the following in Outbound Rewrite Rules for Callee   (YES I did
>>>>>>> say Outbound Rewrite Rules, don’t think of it the way it may appear)
>>>>>>>
>>>>>>> Match Pattern: ^(1|)([1-9][0-9][0-9]+)$   Replacement Pattern:
>>>>>>> \2     Description:  Take off the 1
>>>>>>>
>>>>>>> Now inbounds calls should arrive and work with either your e164
>>>>>>> number assigned to the subscriber /OR/ your alias number assigned to
>>>>>>> that subscriber.
>>>>>>>
>>>>>>>
>>>>>>> Vitelity was quite clear with their error message sent to you with
>>>>>>> the call fail notification.
>>>>>>> That indicates that you are close.
>>>>>>>
>>>>>>> Now, there is one reason that most will agree with in the forum.
>>>>>>> This is offered for free and basically do your best to figure out.
>>>>>>>
>>>>>>> Sipwise offers paid services and a commercial product which will
>>>>>>> work out of the box.  I think if they truly offered all the answers
>>>>>>> in a manual it could be more difficult to offer their commercial
>>>>>>> product / pro product.
>>>>>>>
>>>>>>> I am with you and most others, a good detailed screen shot with
>>>>>>> examples it really missing.
>>>>>>>
>>>>>>> Let me know how you make out with the above.  Using vitelity is
>>>>>>> really straight forward once you get the above setting working.
>>>>>>>
>>>>>>> The examples I provided you are with us running it a SBC mode we do
>>>>>>> not connect any devices to sipwise and only route traffic to
>>>>>>> asterisk boxes in the field for our clients.  No SIP devices
>>>>>>> register to our configuration.  We use trusted sources and send
>>>>>>> traffic to the end user PBX.
>>>>>>>
>>>>>>> I hope this helps.
>>>>>>>
>>>>>>> Kevin
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> -----Original Message-----
>>>>>>> From: spce-user-bounces at lists.sipwise.com
>>>>>>> [mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of Mike Hammett
>>>>>>> Sent: Tuesday, May 07, 2013 4:44 PM
>>>>>>> To: spce-user at lists.sipwise.com
>>>>>>> Subject: Re: [Spce-user] Intro
>>>>>>>
>>>>>>> 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
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> 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
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> 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
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> 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
>

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