Hi Adrian - > When I use re-invite, does the Asterisk server stay in the SIP conversation, > and just RTP traffic diverts, or does the SIP transfer away from the A*k > server too ?
I'm sure somebody will correct me if this is wrong, but I believe the signalling must stay with asterisk, as asterisk needs to know if it should provide any services for the call (music on hold, transfer, etc). - Noah _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
