issue with Asterisk 11.13.1 on Jessie

I upgraded my Dockstar from wheeze to jessie today. As part of the upgrade, asterisk 1.8 is change to 11.13.1. After the upgrade, google voice with asterisk doesn't work anymore. I got the following messages when I tried an outgoing call:

[Aug 9 21:41:13] WARNING[3292][C-00000005]: channel.c:6000 ast_request: No channel type registered for 'Gtalk'
[Aug 9 21:41:13] WARNING[3292][C-00000005]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'Gtalk' (cause 66 - Channel not implemented)
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'SIP/101-00000006' status is 'CHANUNAVAIL'

The phone connects to the asterisk does not ring for incoming call.

I used the 1.8 conf files (extension, sip, gtalk & jabber) with asterisk 11. Are the conf files need to be changed?

Although Google Voice still supports accessing their XMPP service using Gmail account-names and passwords, they consider this a security concern and it's unclear how much longer they will permit this. They would like people to move to the OAUTH2.0 validation model. (Actually, they'd probably prefer to do away with XMPP entirely.) As a result, I'm not sure the later Asterisks have support for the original account-name/password module.

You may want to review the following rather lengthy thread:

http://www.dslreports.com/forum/r30021383-Upgrade-Asterisk-to-an-OAUTH2-0-connection-with-Google-Voice

I'm still using Asterisk 11 and its original module, and it continues to work for me. I'm also keeping my fingers crossed. When it stops working, I guess I'll look into using OAUTH2.0, or just punt on Asterisk and let my ObiHAI ATA handle the connection directly.

If you do get GV working with Asterisk 13, let us know the secrets!

When you say GV continues to work with Asterisk 11 and its original module, does that mean you don't to have to create motif.conf & rtp.conf? From Google search, it seems like the 2 are the ones I need to add for Asterisk 11.

ETA: I am half way there. After adding motif.conf & rtp.conf, I have outgoing call working.



Edited 1 time(s). Last edit at 08/11/2015 09:17AM by funtoy1001.

I followed twinclouds' thread here:

http://forum.doozan.com/read.php?2,1647

I wound up having to change: extensions.conf, gtalk.conf, jabber.conf, modules.conf, and sip.conf.

I don't even have a motif.conf in my implementation.

Sounds like you're making progress. Keep us informed.

I followed the instrucitons from
http://dantheman2865.com/blog/2013/01/asterisk-11-google-voice/

to modify my conf files and able to dial out and receive calls. Will need to do more testing...