For Asterisk to play MOH, it will need to have an RTP connection, right? How otherwise, would you want to play MOH?
Rene Kluwen Chimit > 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. > > Is this resolvable? > > Thanks, > > Ronald Voermans > _______________________________________________ > 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 _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- 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
