[Asterisk-Users] Early media problems...

[James Kelley]

The problem is * not supporting or handling early media.  I have looked
through the sniffer traces and I see the RTP stream being setup between
* and the gateway during the invite and or 183 message, but * does not
setup a corresponding stream to the client until it sees an OK (200)
message.  The result is the end user never hears ringing, although the
call is completed.  
 
I have looked trough the messages, we, wiki and there seems to only be a
few messages on this subject.   I have tried to "fake" the ringing by
adding "r" to the dial command, but it does not seem like a good
solution.  It works great in the case where there is little chance of
getting a busy signal or when the called party has an answering machine
or voice mail. 
 
 
Any ideas, work around?
 
 
Thanks
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