[Asterisk-Dev] SIP : transfer and display update

[fredrik chabot]

Olle E. Johansson wrote:

>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
>  
>
Ok, full became quite large so i send it directly to you and not on the 
list hope thats ok,

>/O :-)
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