Hi All, 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 ? Thanks, Adrian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080715/93453030/attachment.htm
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