[asterisk-dev] audiostream with jingle

Theo Belder T.Belder at trends.nl
Thu Aug 3 08:07:32 MST 2006


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



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