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 <a...@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 > > _______________________________________________ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users