1.4 was working fine. I thought I would try 1.6 beta 9 from my asteirsk 1.4 server to my asterisk client 1.6beta it wont accept the call.
[Jul 18 20:34:55] NOTICE[966]: chan_sip.c:16416 handle_request_invite: Call from 'JJ' to extension 'jj_audio' rejected because extension not found. I changed nothing in the config files. I tried setting debug level to 5 and verbose to 5 all I still get is the one liner above. Has something changed in 1.6 that affects incoming calls (that I have not found) my sip.conf still has the context set to the correct value (as 1.4 did), my extensions.conf still has that context. Thanks for any pointers. Jerry _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
