<br><br><div><span class="gmail_quote"></span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><br>Date: Thu, 3 Aug 2006 17:07:32 +0200<br>From: "Theo Belder" <
<a href="mailto:T.Belder@trends.nl">T.Belder@trends.nl</a>><br>Subject: [asterisk-dev] audiostream with jingle<br>To: <<a href="mailto:asterisk-dev@lists.digium.com">asterisk-dev@lists.digium.com</a>><br>Message-ID:
<br> <<a href="mailto:9EA9361DE1D5B442AB1E1A179FCB2E4A66727C@exch03-01.trends.nl">9EA9361DE1D5B442AB1E1A179FCB2E4A66727C@exch03-01.trends.nl</a>><br>Content-Type: text/plain; charset="us-ascii"
<br><br>Hello guys,<br><br>I'm trying to make phone calls between GoogleTalk and Asterisk.<br>Everything is working except the audio stream. Is this still in<br>development? Or can you tell me what I might doing wrong?<br>
<br>My output on the CLI:<br><br>========================================================================<br>====<br>Asterisk -> GoogleTalk<br>------------------------------------------------------------------------<br>
------ Executing [200@default:1] Dial("SIP/100-09b9a670",<br>"<a href="mailto:JINGLE/asterisk/tbelder@gmail.com">JINGLE/asterisk/tbelder@gmail.com</a>") in new stack<br> -- Called <a href="mailto:asterisk/tbelder@gmail.com">
asterisk/tbelder@gmail.com</a><br> -- <a href="mailto:Jingle/tbelder@gmail.com-448c">Jingle/tbelder@gmail.com-448c</a> is ringing<br> -- <a href="mailto:Jingle/tbelder@gmail.com-448c">Jingle/tbelder@gmail.com-448c</a>
answered SIP/100-09b9a670<br>[Aug 3 17:00:25] WARNING[21056]: rtp.c:2523 ast_rtp_bridge: Can't find<br>native functions for channel '<a href="mailto:Jingle/tbelder@gmail.com-448c">Jingle/tbelder@gmail.com-448c</a>'<br> -- Native bridging SIP/100-09b9a670 and
<br><a href="mailto:Jingle/tbelder@gmail.com-448c">Jingle/tbelder@gmail.com-448c</a> ended<br> == Spawn extension (default, 200, 1) exited non-zero on<br>'SIP/100-09b9a670'<br>========================================================================
<br>====<br><br><br>========================================================================<br>====<br>GoogleTalk -> Asterisk<br>------------------------------------------------------------------------<br>----<br>-- Executing [
s@tbelder:1] NoOp("Jingle/tbelder-a12b", "EXTEN : s") in<br>new stack<br> -- Executing [s@tbelder:2] Answer("Jingle/tbelder-a12b", "") in new<br>stack<br> -- Executing [s@tbelder
:3] Dial("Jingle/tbelder-a12b", "SIP/100") in<br>new stack<br> -- Called 100<br> -- SIP/100-09b9db30 is ringing<br>[Aug 3 17:07:50] NOTICE[21123]: chan_jingle.c:1177 jingle_indicate:<br>Don't know how to indicate condition '3'
<br> -- SIP/100-09b9db30 is ringing<br> -- SIP/100-09b9db30 is ringing<br> -- SIP/100-09b9db30 is ringing<br> -- SIP/100-09b9db30 answered Jingle/tbelder-a12b<br>[Aug 3 17:07:54] NOTICE[21123]: chan_jingle.c:1177 jingle_indicate:
<br>Don't know how to indicate condition '-1'<br>[Aug 3 17:07:54] WARNING[21123]: rtp.c:2517 ast_rtp_bridge: Can't find<br>native functions for channel 'Jingle/tbelder-a12b'<br> -- Native bridging Jingle/tbelder-a12b and SIP/100-09b9db30 ended
<br> == Spawn extension (tbelder, s, 3) exited non-zero on<br>'Jingle/tbelder-a12b'<br>[Aug 3 17:07:59] NOTICE[20967]: chan_jingle.c:560 jingle_hangup_farend:<br>Whoa, didn't find call!<br>========================================================================
<br>====<br><br><br>Greetings,<br>Theo Belder<br>The Netherlands<br><br><br></blockquote></div>I'm having this problem as well. I've talked with Matt over it- and when we both can match some schedules, I'm going to dig deeper into it with him directly- according to him, this problem only occurs in a minority of cases. Exact same symptoms. For now, I've held off filing it on Mantis, as the jingle code is still pretty new, and Matt is pretty accessible.
<br><br>-pbd<br>