[asterisk-dev] audiostream with jingle

[Danish Samad]

Hey guys,

I was surfing through the archives and noticed that you guys faced the exact
problem I am facing now. Whenever I try to bridge a call between SIP
entities and googletalk I get this warning
"ast_rtp_bridge: Can't find native functions for channel 'Jingle/XYZ-9cd8'"

The signaling works fine.
Did any one find a solution to this problem. If so, please let me know.

Regards,
Danish
On 8/3/06, Paul Davidson <planac at gmail.com> wrote:
>
>
>
>
> > Date: Thu, 3 Aug 2006 17:07:32 +0200
> > From: "Theo Belder" < T.Belder at trends.nl>
> > Subject: [asterisk-dev] audiostream with jingle
> > To: <asterisk-dev at lists.digium.com>
> > Message-ID:
> >         <9EA9361DE1D5B442AB1E1A179FCB2E4A66727C at exch03-01.trends.nl>
> > Content-Type: text/plain;       charset="us-ascii"
>
>
>
> Hello guys,
>
> I'm trying to make phone calls between GoogleTalk and Asterisk.
> Everything is working except the audio stream. Is this still in
> development? Or can you tell me what I might doing wrong?
>
> My output on the CLI:
>
> ========================================================================
> ====
> Asterisk -> GoogleTalk
> ------------------------------------------------------------------------
> ------ Executing [200 at default:1] Dial("SIP/100-09b9a670",
> "JINGLE/asterisk/tbelder at gmail.com") in new stack
>     -- Called asterisk/tbelder at gmail.com
>     -- Jingle/tbelder at gmail.com-448c is ringing
>     -- Jingle/tbelder at gmail.com-448c answered SIP/100-09b9a670
> [Aug  3 17:00:25] WARNING[21056]: rtp.c:2523 ast_rtp_bridge: Can't find
> native functions for channel 'Jingle/tbelder at gmail.com-448c'
>     -- Native bridging SIP/100-09b9a670 and
> Jingle/tbelder at gmail.com-448c ended
>   == Spawn extension (default, 200, 1) exited non-zero on
> 'SIP/100-09b9a670'
> ========================================================================
> ====
>
>
> ========================================================================
> ====
> GoogleTalk -> Asterisk
> ------------------------------------------------------------------------
> ----
> -- Executing [ s at tbelder:1] NoOp("Jingle/tbelder-a12b", "EXTEN : s") in
> new stack
>     -- Executing [s at tbelder:2] Answer("Jingle/tbelder-a12b", "") in new
> stack
>     -- Executing [s at tbelder :3] Dial("Jingle/tbelder-a12b", "SIP/100") in
> new stack
>     -- Called 100
>     -- SIP/100-09b9db30 is ringing
> [Aug  3 17:07:50] NOTICE[21123]: chan_jingle.c:1177 jingle_indicate:
> Don't know how to indicate condition '3'
>     -- SIP/100-09b9db30 is ringing
>     -- SIP/100-09b9db30 is ringing
>     -- SIP/100-09b9db30 is ringing
>     -- SIP/100-09b9db30 answered Jingle/tbelder-a12b
> [Aug  3 17:07:54] NOTICE[21123]: chan_jingle.c:1177 jingle_indicate:
> Don't know how to indicate condition '-1'
> [Aug  3 17:07:54] WARNING[21123]: rtp.c:2517 ast_rtp_bridge: Can't find
> native functions for channel 'Jingle/tbelder-a12b'
>     -- Native bridging Jingle/tbelder-a12b and SIP/100-09b9db30 ended
>   == Spawn extension (tbelder, s, 3) exited non-zero on
> 'Jingle/tbelder-a12b'
> [Aug  3 17:07:59] NOTICE[20967]: chan_jingle.c:560 jingle_hangup_farend:
> Whoa, didn't find call!
> ========================================================================
> ====
>
>
> Greetings,
> Theo Belder
> The Netherlands
>
>
> 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.
>
> -pbd
>
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