Irakli Natsvlishvili wrote: > If SIP extensions are in DIFFERENT CONTEXTS, then RTP traffic between > them will ALWAYS go via Asterisk. Dial plan contexts has nothing to do with how we set up RTP traffic.
> I.e. Asterisk WILL NOT issue Re-INVITE even if: > > 1. Both UAs have canreinvite=yes in their SIP.CONF If canreinvite=yes, we *will* issue a re-invite if possible. > 2. Both UAs have same codecs > 3. There are no t, T settings in Dial command. Or h,H or nat=yes. It is easier to turn it around: Asterisk will issue a re-invite unless there is a reason to keep the audio going through Asterisk * NAT traversal issues * Canreinvite=no * Anything that needs asterisk to listen for DTMF in call * Codecs that needs to be transcoded /Olle --- Astricon 2005 - where you will learn about Asterisk and re-invites! http://www.astricon.net/2005/ October 12-14 Anaheim, California _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
