[Asterisk-Dev] SIP : transfer and display update

[Christian Cayeux]

Hello,

I'm confused in the way Asterisk handles the transfer with SIP.
Just say that A makes call to B, holds B, then makes call to C and make the
transfer.
At the end B is in call with C.
On a SIP point of view, A sends a refer to Asterisk, with a refer-to header
and replaces extension.
On receiving of this refer, Asterisk makes reinvite to B and reinvite to C
with right sdp, so that B speaks to C.
The problem is that on its display, B is still with A, and C is still with A
also.
Is there any way so that * handles refer in a different way?

--
Christian.