Hi, I'm using IAX2 between our SIP and PSTN servers, both running Asterisk 1.6.2. Users connect to the SIP server and dial; the SIP server forwards the call to the PSTN server over IAX2, which then dials out over the connected PRI. Since users need detailed call progress feedback, the first action in the dialplan on the PSTN server side is Answer().
In this scenario it's easy for a human to know when a call has been answered. However, the SIP-side Asterisk treats the call as answered the moment the PSTN server executes Answer(). Is there any way of determining on the SIP side when the called party actually picks up the phone? Or if she doesn't, the status of the call as it progresses? Regards, -- Raj -- Raj Mathur r...@kandalaya.org http://kandalaya.org/ GPG: 78D4 FC67 367F 40E2 0DD5 0FEF C968 D0EF CC68 D17F PsyTrance & Chill: http://schizoid.in/ || It is the mind that moves -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
