[asterisk-users] Cisco 7940 and PJSIP registration

[Nilesh Govindrajan]

 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>
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 -> 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>;tag=001469a7180c0011603d4433-6cef1ff3.
>
> To: <sip:233 at 172.22.4.8>.
>
> Call-ID: 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="8".
>
> Content-Length: 0.
>
> Expires: 120.
>
> .
>
>
>
> #
>
> I 172.22.4.8 -> 172.22.3.228 3:3
>
> ....E..:)... 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>;tag=001469a7180c0011603d4433-6cef1ff3.
>
> To: <sip:233 at 172.22.4.8>.
>
> Call-ID: 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="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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