Ronald Voermans wrote:
For canreinvite=yes to work, I think I need to remove the t argument in the Dial(SIP/ext|60|t) application. Otherwise, Asterisk will allways stay in the middle. I don't want that, so I removed the 't' argument. That works. Now, when two UA are calling, Asterisk gets out of the RTP stream. However, when removing the 't' argument, the Music On Hold doesn't work anymore between these two UA. If I put one UA on hold, Asterisk states that it is starting Music On Hold, but the holding party doesn't hear the audio stream.
Umm.. "DUH!" If you remove the RTP stream from asterisk, asterisk can't send audio (the rtp stream) to the phones.
-Matthew _______________________________________________ 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
