[Asterisk-Dev] SIP : transfer and display update
Olle E. Johansson
oej at edvina.net
Thu Jun 9 06:29:42 MST 2005
Christian Cayeux wrote:
> 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
> 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
> Is there any way so that * handles refer in a different way?
...working on it, bit by bit. Stay tuned.
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