[Asterisk-Dev] SIP : transfer and display update

[Olle E. Johansson]

Fredrik Chabot wrote:
> 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.
> 
As always, I would like to see a SIP debug with debug=4 and verbose=4

/O :-)