[Users] REFER To: (how to handle in OpenSER)

Bogdan-Andrei Iancu bogdan at voice-system.ro
Fri Jul 14 16:49:16 CEST 2006


Hi,

take a look on the network trace and see if the REFER is properly routed 
also in the second case.

regards,
bogdan

Frogger wrote:

>I have done a lot of investigation and debugging. 
>Here is where I am so far.
>
>I am able to get transfers working in the following
>situation:
>
>++This works++
>==============
>SIP-ua1 calls pstn phone through gw
>
>SIP-ua1 transfers pstn phone to final-callee:  SIP-ua2
>==============
>
>
>
>**This does NOT work**
>==============
>PSTN phone calls SIP-ua1
>
>SIP-ua1 transfers pstn phone to final-callee:  SIP-ua2
>==============
>
>Basically, if sip-ua1 is the primary caller, the
>transfer works.  If pstn gw is the primary caller, the
>transfer does not work.
>
>Any thoughts?
>
>Please, even the slightest comment can be helpful to
>crack this case.
>
>Thank you!  FR  
>
>
>--- Juha Heinanen <jh at tutpro.com> wrote:
>
>  
>
>>Frogger writes:
>>
>> > I am concerned about the "@sip.refer.com".  I am
>>not
>> > sure how the gateway is handling this.
>>
>>my understanding is that cisco doesn't look host
>>part at all.  as i
>>said, debug your dial plan when refer comes in.
>>
>>-- juha
>>
>>    
>>
>
>
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