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 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: [email protected] [mailto:[email protected]] On Behalf Of Andres Sent: Wednesday, 22 July 2015 9:29 PM To: [email protected] Subject: Re: [asterisk-users] Cisco 7940 and PJSIP registration 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: [email protected]<mailto:[email protected]> [mailto:[email protected]] 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 <[email protected]<mailto:[email protected]>> 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:[email protected]<mailto:sip%[email protected]>>;tag=001469a7180c0011603d4433-6cef1ff3. To: <sip:[email protected]<mailto:sip%[email protected]>>. Call-ID: [email protected]<mailto:[email protected]>. Max-Forwards: 70. Date: Wed, 22 Jul 2015 00:41:48 GMT. CSeq: 114 REGISTER. User-Agent: Cisco-CP7940G/8.0. Contact: <sip:[email protected]: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..:)...@................&..REGISTER<mailto:...@................&..REGISTER> sip:172.22.4.8 SIP/2.0. Via: SIP/2.0/UDP 172.22.3.228:5060;branch=z9hG4bK505b3494. From: <sip:[email protected]<mailto:sip%[email protected]>>;tag=001469a7180c0011603d4433-6cef1ff3. To: <sip:[email protected]<mailto:sip%[email protected]>>. Call-ID: [email protected]<mailto:[email protected]>. Max-Forwards: 70. Date: Wed, 22 Jul 2015 00:41:48 GMT. CSeq: 114 REGISTER. User-Agent: Cisco-CP7940G/8.0. Contact: <sip:[email protected]: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/> -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Technical Support http://www.cellroute.net
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
