On 02/10/2011 06:15 AM, Benoit Panizzon wrote:
Hi Faisal
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
Thank you, but I seem to miss the option which tells asterisk to pass audio
even if no 183 or 200 is received.
No, we don't set the 'r' Flag while dialing out.
So, my question ist sill the same.
Sould asterisk pass audio of it didn't yet receive a 183 or 200 message, or is
the carrier doing wrong in sending early audio without 183?
This does indeed sound like an Asterisk bug; Asterisk should be ready
and willing to accept audio from the called SIP endpoint as soon as the
INVITE is sent out with an SDP offer to receive audio.
Now the real issue here may be the Dial() application not forwarding
that audio to the caller, rather than Asterisk not 'accepting' the audio
and turning it into internal media frames. The net result for you is the
same, but the source of the problem is quite different.
This can of course cause complications if Dial() is used to dial
multiple endpoints... because then there could be multiple audio streams
received from them as the call proceeds towards one of them answering.
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