Hi, I got a problem with asterisk 1.8.9.2. The same scenario is working fine in 1.8.8.2.
Asterisk calls a SIP phone via a proxy, proxy phone and asterisk are on the same LAN, no NAT. Asterisk sends the INVITE to the proxy, the proxy sends INVITE to the phone. The phone sends 180 RINGING back to the proxy. The proxy sends 180 RINGING to asterisk. So far so good. If the calling side decides to cancel the call, asterisk sends the CANCEL directly to the phone. The phone doesn't find the call and answers 404. In asterisk 1.8.8.2 asterisk sends the CANCEL to the proxy, which sends the CANCEL to the phone and all ist fine. I think, the new behavior comes from the lines parse_ok_contact(p, req); if (!reinvite) { build_route(p, req, 1); } which are inserted in the handling of provisional SIP response. Am I doing something wrong or is this a bug? Thanks, Karsten -- _____________________________________________________________________ -- 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