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        <title>issue with Asterisk 11.13.1 on Jessie</title>
        <description> 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&amp;#039;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 &amp;#039;Gtalk&amp;#039;
[Aug  9 21:41:13] WARNING[3292][C-00000005]: app_dial.c:2437 dial_exec_full: Unable to create channel of type &amp;#039;Gtalk&amp;#039; (cause 66 - Channel not implemented)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel &amp;#039;SIP/101-00000006&amp;#039; status is &amp;#039;CHANUNAVAIL&amp;#039;

The phone connects to the asterisk does not ring for incoming call.

I used the 1.8 conf files (extension, sip, gtalk &amp;amp; jabber) with asterisk 11. Are the conf files need to be changed?</description>
        <link>https://forum.doozan.com/read.php?2,23202,23202#msg-23202</link>
        <lastBuildDate>Sun, 15 Mar 2026 15:56:05 -0500</lastBuildDate>
        <generator>Phorum 5.2.23</generator>
        <item>
            <guid>https://forum.doozan.com/read.php?2,23202,23234#msg-23234</guid>
            <title>Re: issue with Asterisk 11.13.1 on Jessie</title>
            <link>https://forum.doozan.com/read.php?2,23202,23234#msg-23234</link>
            <description><![CDATA[ I followed the instrucitons from<br />
<a href="http://dantheman2865.com/blog/2013/01/asterisk-11-google-voice/"  rel="nofollow">http://dantheman2865.com/blog/2013/01/asterisk-11-google-voice/</a><br />
<br />
to modify my conf files and able to dial out and receive calls. Will need to do more testing...]]></description>
            <dc:creator>funtoy1001</dc:creator>
            <category>Debian</category>
            <pubDate>Tue, 11 Aug 2015 11:34:21 -0500</pubDate>
        </item>
        <item>
            <guid>https://forum.doozan.com/read.php?2,23202,23232#msg-23232</guid>
            <title>Re: issue with Asterisk 11.13.1 on Jessie</title>
            <link>https://forum.doozan.com/read.php?2,23202,23232#msg-23232</link>
            <description><![CDATA[ I followed twinclouds&#039; thread here:<br />
<br />
<a href="http://forum.doozan.com/read.php?2,1647"  rel="nofollow">http://forum.doozan.com/read.php?2,1647</a><br />
<br />
I wound up having to change: extensions.conf,  gtalk.conf,  jabber.conf,  modules.conf,  and sip.conf.<br />
<br />
I don&#039;t even have a motif.conf in my implementation.<br />
<br />
Sounds like you&#039;re making progress.  Keep us informed.]]></description>
            <dc:creator>restamp</dc:creator>
            <category>Debian</category>
            <pubDate>Tue, 11 Aug 2015 10:12:49 -0500</pubDate>
        </item>
        <item>
            <guid>https://forum.doozan.com/read.php?2,23202,23230#msg-23230</guid>
            <title>Re: issue with Asterisk 11.13.1 on Jessie</title>
            <link>https://forum.doozan.com/read.php?2,23202,23230#msg-23230</link>
            <description><![CDATA[ When you say GV continues to work with Asterisk 11 and its original module, does that mean you don&#039;t to have to create motif.conf &amp; rtp.conf? From Google search, it seems like the 2 are the ones I need to add for Asterisk 11.<br />
<br />
ETA: I am half way there. After adding motif.conf &amp; rtp.conf, I have outgoing call working.]]></description>
            <dc:creator>funtoy1001</dc:creator>
            <category>Debian</category>
            <pubDate>Tue, 11 Aug 2015 07:34:10 -0500</pubDate>
        </item>
        <item>
            <guid>https://forum.doozan.com/read.php?2,23202,23226#msg-23226</guid>
            <title>Re: issue with Asterisk 11.13.1 on Jessie</title>
            <link>https://forum.doozan.com/read.php?2,23202,23226#msg-23226</link>
            <description><![CDATA[ Although Google Voice still supports accessing their XMPP service using Gmail account-names and passwords, they consider this a security concern and it&#039;s unclear how much longer they will permit this.  They would like people to move to the OAUTH2.0 validation model.  (Actually, they&#039;d probably prefer to do away with XMPP entirely.)  As a result, I&#039;m not sure the later Asterisks have support for the original account-name/password module.<br />
<br />
You may want to review the following rather lengthy thread:<br />
<br />
<a href="http://www.dslreports.com/forum/r30021383-Upgrade-Asterisk-to-an-OAUTH2-0-connection-with-Google-Voice"  rel="nofollow">http://www.dslreports.com/forum/r30021383-Upgrade-Asterisk-to-an-OAUTH2-0-connection-with-Google-Voice</a><br />
<br />
I&#039;m still using Asterisk 11 and its original module, and it continues to work for me.  I&#039;m also keeping my fingers crossed.  When it stops working, I guess I&#039;ll look into using OAUTH2.0, or just punt on Asterisk and let my ObiHAI ATA handle the connection directly.<br />
<br />
If you do get GV working with Asterisk 13, let us know the secrets!]]></description>
            <dc:creator>restamp</dc:creator>
            <category>Debian</category>
            <pubDate>Mon, 10 Aug 2015 23:01:23 -0500</pubDate>
        </item>
        <item>
            <guid>https://forum.doozan.com/read.php?2,23202,23202#msg-23202</guid>
            <title>issue with Asterisk 11.13.1 on Jessie</title>
            <link>https://forum.doozan.com/read.php?2,23202,23202#msg-23202</link>
            <description><![CDATA[ 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&#039;t work anymore. I got the following messages when I tried an outgoing call:<br />
<br />
[Aug  9 21:41:13] WARNING[3292][C-00000005]: channel.c:6000 ast_request: No channel type registered for &#039;Gtalk&#039;<br />
[Aug  9 21:41:13] WARNING[3292][C-00000005]: app_dial.c:2437 dial_exec_full: Unable to create channel of type &#039;Gtalk&#039; (cause 66 - Channel not implemented)<br />
  == Everyone is busy/congested at this time (1:0/0/1)<br />
    -- Auto fallthrough, channel &#039;SIP/101-00000006&#039; status is &#039;CHANUNAVAIL&#039;<br />
<br />
The phone connects to the asterisk does not ring for incoming call.<br />
<br />
I used the 1.8 conf files (extension, sip, gtalk &amp; jabber) with asterisk 11. Are the conf files need to be changed?]]></description>
            <dc:creator>funtoy1001</dc:creator>
            <category>Debian</category>
            <pubDate>Sun, 09 Aug 2015 23:48:12 -0500</pubDate>
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