|
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 |
_______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
