[asterisk-users] Cisco 7940 and PJSIP registration

Wed Jul 22 00:38:58 CDT 2015

I’ve gotten to the bottom of this;

Seems that the pjsip.endpoint_custom.conf isn’t getting included properly, or my syntax is wrong.

If I put force_rport=no into pjsip.endpoint.conf and reload only Asterisk, everything works perfectly.  Unfortunately, I’m using FreePBX, so it owns this file and my changes won’t persist a FreePBX reload.

Brendan Ord
OntheNet - Network Engineer
P 07 5553 9222
F 07 5593 3557
Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map<https://goo.gl/maps/p25WF>)
www.OntheNet.com.au<http://www.onthenet.com.au/>

From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Nilesh Govindrajan
Sent: Wednesday, 22 July 2015 11:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco 7940 and PJSIP registration

 I had exact same issue with pjsip instead of sip - I was able to solve it by setting the password to blank. But I switched to asterisk 11 because the chan_mobile module was giving me troubles in 13.

On Wed, Jul 22, 2015 at 7:07 AM, Brendan Ord <bord at staff.onthenet.com.au<mailto:bord at staff.onthenet.com.au>> wrote:
Hi list,

I’ve been googling this issue and found some good resources however I am still running into problems with the following combo … Here’s my story;

-      Asterisk 13.4 with FreePBX 12.

-      Migrating from Asterisk 11 / FreePBX 2.11

-      Mix of Cisco 79xx handsets, mostly 7940G’s.

My problems started with (the very common) issue of the 7940 not replying to 401 UNAUTHORIZED with a second REGISTER containing the auth digest details.  A quick Google found a heap of information in various forums, all with replies from Joshua Colp stating that force_rport=no needs to be set for these endpoints, see http://forums.digium.com/viewtopic.php?f=1&t=91699

So, (being that this is FreePBX and the main conf files are controlled by that) I jumped into /etc/asterisk/pjsip.endpoint_custom.conf and added;

[233]
force_rport=no

Reloaded everything, recreated the extension and tested again, watching what goes between this endpoint with ‘ngrep –W byline host 172.22.3.228’ and now I get something which I don’t fully understand;

U 172.22.3.228:51440<http://172.22.3.228:51440> -> 172.22.4.8:5060<http://172.22.4.8:5060>
REGISTER sip:172.22.4.8 SIP/2.0.
Via: SIP/2.0/UDP 172.22.3.228:5060;branch=z9hG4bK505b3494.
From: <sip:233 at 172.22.4.8<mailto:sip%3A233 at 172.22.4.8>>;tag=001469a7180c0011603d4433-6cef1ff3.
To: <sip:233 at 172.22.4.8<mailto:sip%3A233 at 172.22.4.8>>.
Call-ID: 001469a7-180c0002-58faebd6-05b99917 at 172.22.3.228<mailto:001469a7-180c0002-58faebd6-05b99917 at 172.22.3.228>.
Max-Forwards: 70.
Date: Wed, 22 Jul 2015 00:41:48 GMT.
CSeq: 114 REGISTER.
User-Agent: Cisco-CP7940G/8.0.
Contact: <sip:233 at 172.22.3.228:5060;user=phone;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-001469a7180c>";+u.sip!model.ccm.cisco.com<http://model.ccm.cisco.com>="8".
Content-Length: 0.
Expires: 120.
.

#
I 172.22.4.8 -> 172.22.3.228 3:3
....E..:)... at ................&..REGISTER<mailto:... at ................&..REGISTER> sip:172.22.4.8 SIP/2.0.
Via: SIP/2.0/UDP 172.22.3.228:5060;branch=z9hG4bK505b3494.
From: <sip:233 at 172.22.4.8<mailto:sip%3A233 at 172.22.4.8>>;tag=001469a7180c0011603d4433-6cef1ff3.
To: <sip:233 at 172.22.4.8<mailto:sip%3A233 at 172.22.4.8>>.
Call-ID: 001469a7-180c0002-58faebd6-05b99917 at 172.22.3.228<mailto:001469a7-180c0002-58faebd6-05b99917 at 172.22.3.228>.
Max-Forwards: 70.
Date: Wed, 22 Jul 2015 00:41:48 GMT.
CSeq: 114 REGISTER.
User-Agent: Cisco-CP7940G/8.0.
Contact: <sip:233 at 172.22.3.228:5060;user=phone;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-001469a7180c>";+u.sip!model.ccm.cisco.com<http://model.ccm.cisco.com>="8".
Content-Lengt

I don’t understand this reply from Asterisk (172.22.4.8) – why it’s not complete and what’s this 3:3?

If anyone has input or experience with this problem I would be forever grateful.  I have read that people can get these handsets working with chan_sip (and, indeed they do, as these handsets are working perfectly using chan_sip in Asterisk 11), but I would really like to keep everything using pjsip (for the reason that, this is where development and improvements are heading, and I like to be using the best technology if possible).

Thank you…

Brendan Ord
OntheNet - Network Engineer
P 07 5553 9222
F 07 5593 3557
Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map<https://goo.gl/maps/p25WF>)
www.OntheNet.com.au<http://www.onthenet.com.au/>

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