Hello Everyone, I have a new problem where when placing the call, asterisk will automatically go into music on hold until the call is connected (ie, no ringing). It was kind of confusing because sometimes `SESSION PROGRESS` takes longer than others, during this time we are in MOH. The call does eventually connect and the MOH stops. When debugging I saw the following debug message:
[Sep 10 10:56:12] DEBUG[7930]: res_rtp_asterisk.c:1228 ast_rtp_write: No remote address on RTP instance '0xb6e00b20' so dropping frame [Sep 10 10:56:12] DEBUG[7930]: res_rtp_asterisk.c:1228 ast_rtp_write: No remote address on RTP instance '0xb6e00b20' so dropping frame [Sep 10 10:56:12] DEBUG[7930]: res_rtp_asterisk.c:1228 ast_rtp_write: No remote address on RTP instance '0xb6e00b20' so dropping frame [Sep 10 10:56:12] DEBUG[7930]: res_rtp_asterisk.c:1228 ast_rtp_write: No remote address on RTP instance '0xb6e00b20' so dropping frame This is a straight SIP channel. No DAHDI. Your Help is Greatly Appreciated. This is a new one for me :) Nick. -- _____________________________________________________________________ -- 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
