[Asterisk-Dev] SIP : transfer and display update

[Fredrik Chabot]

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.
>  
>
I have the strong impression there is a bug in the sip transfer when the
call commes from an queue, when a agent logs in on a sip terminal, then
gets an call from the queue and tries to tranfer that to an other sip
terminal you get oneway audio. with canreinvite=NO it works ok.

Fredrik

>/Olle
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