|Summary:||ASTERISK-06673: Dial(type/identifier, timeout, m(message)) doesn't send RTP till some RTP arrives|
|Date Opened:||2006-03-31 06:49:24.000-0600||Date Closed:||2006-03-31 08:48:24.000-0600|
|Description:||I'm using something like|
My Asterisk sends INVITE to the callee, then send 183 Session progres with SDP to the caller (sip-proxy of my provider), starts MOH, but sends NO RTP streams till some RTP arrives!
And becauseof it is inband info, provider can't send me audio BEFORE I answer the line - it could only listen.
This bug is codec independent and type of MOH playing app (i.e. files, custom..) independant also...
BTW: Playback(message,noanswer) works well.
|Comments:||By: BJ Weschke (bweschke) 2006-03-31 08:48:24.000-0600|
You may want to try putting in an Answer() in the step before Dial. That should force RTP to get nailed up. If not.....
The fact that asterisk doesn't send RTP until it receives RTP is a known issue with the current implementation. You may want to take a look at the "internal timing" feature that was recently added to the /trunk (development) version for more assistance with your issue but this isn't something that is going to be fixed in 1.2.X.