Hi! > I don't think that is what keeping the original poster's system from > working. The issue is "one" extension is configured for canreinvite=no > and the other is canreinvite=yes. One extension believes all RTP must > be passed through * while the other is attempting to negotiate a > phone-to-phone RTP session, thus dropping the audio.
Are you sure this is 100% correct? I have some doubts since: - you'd have to consider all possible connection permutations between all clients and then set canreinvite= accordingly, which doesn't sound like it makes much sense - sip.conf is for * only, the data are not seen or read by the SIP UA themselves. Thus it would appear that it is up to * to permit/not permit a reinvite between the two UAs So bascially from my understanding things work like this: Once one of the SIP call parties has a canreinvite=no it won't matter what the other party's setting looks like, RTP traffic will travel through * anyway. Am I wrong here? Philipp _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
