2009/4/7 Adrian Georgescu <[email protected]>: > 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 [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
