[Asterisk-Dev] REFER with Replaces

[Arnaud]

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