[asterisk-users] Cisco 7940 and PJSIP registration

Wed Jul 22 06:29:09 CDT 2015

On 7/22/15 1:38 AM, Brendan Ord wrote:
>
> 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.
>
Last time I checked you have to put a plus sign to combine parameters 
from main and custom file.  Like this:

[233](+)

force_rport=no
>
> 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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