Cisco 7940 And PJSIP Registration
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:
.
#
I 172.22.4.8 -> 172.22.3.228 3:3
….E..:)…@…………….&..REGISTER sip:172.22.4.8 SIP/2.0. Via: SIP/2.0/UDP 172.22.3.228:5060;branch=z9hG4bK505b3494. From:
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
www.OntheNet.com.au
6 thoughts on - 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.
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DQo
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
Thank you.
I read that last yesterday afternoon, and I could’ve sworn I tried that but I will look into it again (I’ve tried so many different things it was getting cloudy what I’ve tried and what worked etc, combined that the extension config gets messed up after playing with it so much so I’m often recreating it as well).
I also found a bug report in the FreePBX bug tracker http://issues.freepbx.org/browse/FREEPBX-8517
It’s not exactly the same, but it’s very similar and the closing comment was “limitation of pjsip”. I might be getting ahead of myself, but would anyone be able to comment on that? Anyway, this looks like a FreePBX issue so this isn’t the ideal forum to discuss their bugs.
Brendan Ord 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>>;tag=001469a7180c0011603d4433-6cef1ff3. To: >. Call-ID: 001469a7-180c0002-58faebd6-05b99917@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.instance=”“;+u.sip!model.ccm.cisco.com<http://model.ccm.cisco.com>=”8″. Content-Length: 0. Expires: 120.
REGISTER sip:172.22.4.8 SIP/2.0. Via: SIP/2.0/UDP 172.22.3.228:5060;branch=z9hG4bK505b3494. From:
.
# sip:172.22.4.8 SIP/2.0. Via: SIP/2.0/UDP 172.22.3.228:5060;branch=z9hG4bK505b3494. From: >;tag=001469a7180c0011603d4433-6cef1ff3. To: >. Call-ID: 001469a7-180c0002-58faebd6-05b99917@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.instance=”“;+u.sip!model.ccm.cisco.com<http://model.ccm.cisco.com>=”8″. Content-Lengt
I 172.22.4.8 -> 172.22.3.228 3:3
….E..:)…@…………….&..REGISTER
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>)
http://www.OntheNet.com.au<http://www.onthenet.com.au/>
Hello,
Putting this into pjsip.endpoint_custom.conf;
[233](+)
force_rport=no
Didn’t work – as can be seen below, Asterisk is still replying on the source port instead of 5060;
#;tag=001469a7180c00a6368fbc8e-7fedf772. To: . Call-ID: 001469a7-180c0002-58f81e51-5a17e8fb@172.22.4.50. Max-Forwards: 70. Date: Thu, 23 Jul 2015 02:53:26 GMT. CSeq: 265 REGISTER. User-Agent: Cisco-CP7940G/8.0. Contact: ;+sip.instance=”“;+u.sip!model.ccm.cisco.com=”8”. Content-Length: 0. Expires: 120.
U 172.22.4.50:51037 -> 172.22.4.8:5060
REGISTER sip:172.22.4.8 SIP/2.0. Via: SIP/2.0/UDP 172.22.4.50:5060;branch=z9hG4bK6c6b20ee. From:
.
#. From: ;tag=001469a7180c00a6368fbc8e-7fedf772. To: ;tag=z9hG4bK6c6b20ee. CSeq: 265 REGISTER. WWW-Authenticate: Digest realm=”asterisk”,nonce=”1437620030/5d32272e81266a723b2c090074799fe2″,opaque=”65e177e35509170b”,algorithm=md5,qop=”auth”. Server: FPBX-AsteriskNOW-12.0.73(13.4.0). Content-Length: 0.
U 172.22.4.8:5060 -> 172.22.4.50:51037
SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 172.22.4.50:5060;rport=51037;received=172.22.4.50;branch=z9hG4bK6c6b20ee. Call-ID: 001469a7-180c0002-58f81e51-5a17e8fb@172.22.4.50
.
Brendan Ord 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>)
http://www.OntheNet.com.au<http://www.onthenet.com.au/>
Another FreePBX forum member has logged a bug report for me, so this thread can be closed.
http://issues.freepbx.org/browse/FREEPBX-9810
Thanks for everyone’s input with this problem.
Brendan Ord 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>)
http://www.OntheNet.com.au<http://www.onthenet.com.au/>