[asterisk-users] Playback noanswer & SIP

Jorge Mendoza jmendozat at tcc.com.pe
Fri May 20 13:37:17 CDT 2011


Hi,
I would to send a message to an incoming call with no answer. My Asterisk server receive the incoming call by a BRI/SIP gateway (Could be a PRI, for instance).
I do the command playback with option noanswer, Asterisk send 183 followed by RTP and finish with 603. But the BRI gateway do not allow to pass the RTP without a 200 OK.
The question is: are there a SIP command to indicate the gateway to allow pass the RTP without the 200?
This is an usual case when the Service Provider play a message like "I'm sorry, you have dialed a wrong number...". So, I assume that the SIP protocol have foreseen the commands to implement this feature, I hope.
Thank You
--
Jorge Mendoza



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