Hello, I am trying to re-use attempt_transfer() found in chan_sip.c to change the bridging of channels. (check http://lists.digium.com/pipermail/asterisk-dev/2005-November/017206.html if you wonder why the heck I'm doing that)
When i use attempt_transfer(sip_pvt *p1, sip_pvt *p2) both of the channels associated with p1 and p2 are hungup by *. No debug or other verbose message is printed on CLI. There are initially 3 sip callid, represented by say p1, p2 and p3. p1 is bridged with p3 and I attempt_transfer(p1, p2) . The goal being to bridge p2 with p3. By looking at the bugtracker I kind of conclude that the support for call transfer with SIP is work in progress. How many have successfully used the attempt_transfer function in chan_sip ? This is only used when a REFER with Replaces is received by * and the callid mentioned in Replaces is found on *. Cheers - Arnaud _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
