[OpenSIPS-Users] dispatcher and attended transfers

[Stanisław Pitucha]

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