Hi, just wondering whether the following scenario (which it's not possible in kamailio 1.5) is possible in 3.X:
1) Alice calls a PSTN number. 2) Proxy receives the call and routes it to a media server which plays an early-announcement (30 seconds long). 3) After N seconds (maybe 10 seconds) proxy starts a new parallel branch to the PSTN gateway *without* cancelling the previous branch with the media server. 4) Let's suppose the PSTN phone rings and proxy gets 183 with SDP (ringing in real audio). Proxy converts such 183 into a 180 with no SDP (already possible in Kamailio). 5) After 30 seconds media server rejects the call with 480. PSTN branch remains ringing as usual. I just would like to know if Kamailio 3.X allows point 3. I think it's hard but maybe I miss some feature. Thanks a lot. -- Iñaki Baz Castillo <i...@aliax.net> _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
