[OpenSIPS-Users] dispatcher and attended transfers

[Bogdan-Andrei Iancu]

Hi Stan,

if I got it right, you want to have a kind of dispatching to guarantee 
that all in or out calls for user A are going through the same PBX. Correct?

And the problem is when you to a REFER....you have A talking to PBX1 and 
it wants to do a transfer ? Or?

Regards,
Bogdan


Stanisław Pitucha wrote:
> 2009/4/7 Adrian Georgescu <ag at ag-projects.com>:
>   
>> You cannot do this reliable the way you propose. The only reliable way is to
>> sit behind a PBX/B2BUA that your control and behaves in a consistent and
>> reliable way. Otherwise you are at the mercy at the combinations of the SIP
>> User Agents that are involved in the call transfer operation.
>>     
>
> There is only one specific scenario I want to support:
> - phone has a dialog already open to a PBX
> - phone sends an new call INVITE  to a PBX
> - phone joins the call legs with a REFER
>
> I think, this is the PBX/B2BUA situation you're talking about?
>
> I'm not sure what you mean by "the combinations of the SIP User Agents
> that are involved". I didn't have any problems with this setup as long
> as the same phone always uses the same pbx.
>
>   
>> If you will try to fix incrementally every problem your discover in the SIP
>> Proxy for call transfer you will be busy forever solving this because is
>> end-point implementation dependent.
>>     
>
> I'm only trying to solve failover + distribution over PBXes in the
> proxy. Transfers are properly handled by N asterisk hosts.
> To be specific - my network looks like this:
> UAs <-> openser (with dispatcher) <-> N identical asterisk boxes
> All calls go through one of the asterisk boxes.
>
> Stan
>
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