[Asterisk-Dev] SIP : transfer and display update
devans at invores.com
Thu Jun 9 12:46:37 MST 2005
Does * really need fixing? I think there is only one issue (see below).
In our system, we receive an external call (any protocol), * forwards it
to a IVR voice recognition autoattendant using SIP. The AA determines
where the call is to be sent. It reinvites *, and then does a REFER. *
takes back the call and sends it to the REFER'ed destination.
The only issue we found is the CSeq: header. * wants this to be the same
number as the initial INVITE tranaction, but you could read the SIP RFC
as requiring a new number. If a trace of this would help you, shoot me
an email off list.
Olle E. Johansson wrote:
> 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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