From: Bruce Atherton <[EMAIL PROTECTED]> My problem is that I don't want the PBX to respond "200 OK" to the INVITE until the PSTN line has actually been picked up. In other words, the SIP call status should be based on the PSTN call status. I'm not worried about "180 Ringing" state as it is difficult to detect ringing on the PSTN, but "200 OK" must wait for the voltage drop that indicates a connection on the PSTN.
Remember that the endpoint user agents in the SIP dialog are your user agent and the PSTN gateway device -- the PBX just passes SIP messages between them. Generally speaking PSTN gateways only return 200 to an INVITE when the PSTN reports that the other end of the PSTN call has answered. Dale _______________________________________________ Sip-implementors mailing list [email protected] https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
