Noah Miller schrieb: > 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).
yes, 'only' rtp goes direct, SIP stay on asterisk since it might be a hangup or something else comes in. Yours Martin _______________________________________________ -- 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
