That is really good news. So the problem must just be with Asterisk, which happens to be the PBX I am testing with. I will take the issue up with the developers there.
Thanks a lot for your help. I was worried that I was looking at a much larger problem. Dale.Worley-at-comcast.net |SIP Implementors| wrote: > 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 > _______________________________________________ Sip-implementors mailing list [email protected] https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
