[Asterisk-Dev] SIP : transfer and display update

[Olle E. Johansson]

Christian Cayeux wrote:
> Hello,
> 
> I'm confused in the way Asterisk handles the transfer with SIP.
So am I, and I have been trying to fix it for a month... :-)

> 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?
...working on it, bit by bit. Stay tuned.

/Olle