[Asterisk-Dev] SIP : transfer and display update

[Dan Evans]

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.

Dan

Olle E. Johansson wrote:
> 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
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