[asterisk-users] PJSIP configuration question

[Dan Cropp]

Ok, it didn't quite solve everything.

There is one slight issue.  When I answer the call on my cell phone, Asterisk sees it as answered.
I can play audio, send dtmfs, etc and hear it on my phone.
However, a short while later, Vitelity tears down that call and Asterisk is never notified about it.

I tell Asterisk to hang up the call (via AMI) and it is removed from Asterisk.

I gather the pjsip trace.  Then, I shut down that VM, fired up another running chan_sip.  Did the same behavior and gathered the sip trace.
Using chan_sip, the call worked flawlessly.


Vitelity sends Asterisk the ACK (for the answer).
Asterisk send an ACK in response.  For the sip.conf system, the ACK includes the Contact for the response.  For PJSIP, the Contact field is not in the ACK

Is there a setting to indicate whether the Contact field should be sent as part of the ACK (response to the OK)?

Have a great day!
Dan


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Dan Cropp
Sent: Thursday, December 11, 2014 9:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PJSIP configuration question

This fixed the problem.

Developer before me wrote some code to build up the dial string.
Always thought that string appeared off, but it worked so I left it alone.

Thanks Joshua and George for helping with this.

Have a great day!

Dan

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Dan Cropp
Sent: Thursday, December 11, 2014 8:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PJSIP configuration question

Thank you Joshua.

I will make the modifications this morning and give it a try.

Have a great day!

Dan



-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joshua Colp
Sent: Wednesday, December 10, 2014 7:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PJSIP configuration question

<snip>

>
> I translated those settings to the following for pjsip.conf...
>
> [transport1]
> type = transport
> bind = 0.0.0.0
> protocol = udp
>
> [outbound.vitelity.net]
> type = aor
> remove_existing = yes
> contact = sip:64.2.142.93 at 5060

This is incorrect. The contact should be:

contact = sip:64.2.142.93

It will use a default port of 5060.

I also believe I've covered your origination issue in a separate email. 
Your dial string should be:

PJSIP/8005555555 at outbound.vitelity.net

Cheers,

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org

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