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:[email protected]>;tag=001469a7180c0011603d4433-6cef1ff3. To: <sip:[email protected]>. Call-ID: [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="8". Content-Length: 0. Expires: 120. . # 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: <sip:[email protected]>;tag=001469a7180c0011603d4433-6cef1ff3. To: <sip:[email protected]>. Call-ID: [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="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
