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]> 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:[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 >
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