We are having a weird problem where calls get cut off in the middle. I'm not a SIP expert but could the INVITE with an empty SDP be the problem?
|Time | 209.220.119.18 | | | | 208.88.61.150 | |9687.369 | INVITE SDP (g729 telephone-eventRTPType-101) |SIP From: <sip:[email protected] To:<sip:[email protected] | |(5060) ------------------> (5060) | |9687.390 | 100 Trying| |SIP Status | |(5060) <------------------ (5060) | |9687.394 | 200 OK SDP (g729 telephone-eventRTPType-101) |SIP Status | |(5060) <------------------ (5060) | |9687.395 | ACK | |SIP Request | |(5060) ------------------> (5060) | |9687.396 | INVITE SDP (g729 telephone-eventRTPType-101) |SIP From: <sip:[email protected] To:<sip:[email protected] | |(5060) ------------------> (5060) | |9687.416 | 100 Trying| |SIP Status | |(5060) <------------------ (5060) | |9687.422 | 200 OK SDP (g729 telephone-eventRTPType-101) |SIP Status | |(5060) <------------------ (5060) | |9687.422 | ACK | |SIP Request | |(5060) ------------------> (5060) | |9687.630 | RTP (g729) |RTP Num packets:149 Duration:2.959s SSRC:0x3C4C9C24 | |(14594) ------------------> (12310) | |9690.591 | INVITE SDP (g729 telephone-eventRTPType-101) |SIP Request | |(5060) <------------------ (5060) | |9690.592 | 100 Trying| |SIP Status | |(5060) ------------------> (5060) | |9690.592 | 200 OK SDP (g729 telephone-eventRTPType-101) |SIP Status | |(5060) ------------------> (5060) | |9690.610 | RTP (g729) |RTP Num packets:35400 Duration:707.979s SSRC:0x3C4C9C24 | |(14594) ------------------> (12310) | |9690.611 | ACK | |SIP Request | |(5060) <------------------ (5060) | |10398.610| RTP (g729) |RTP Num packets:1 Duration:0.000s SSRC:0x6260DF4F | |(14594) ------------------> (12310) | |10398.702| INVITE SDP () |SIP From: <sip:[email protected] To:<sip:[email protected] | |(5060) ------------------> (5060) | |10398.719| 488 Not acceptable here |SIP Status | |(5060) <------------------ (5060) | |10398.719| ACK | |SIP Request | |(5060) ------------------> (5060) | |10398.720| INVITE SDP (g729 telephone-eventRTPType-101) |SIP From: <sip:[email protected] To:<sip:[email protected] | |(5060) ------------------> (5060) | |10398.740| 100 Trying| |SIP Status | |(5060) <------------------ (5060) | |10398.746| 200 OK SDP (g729 telephone-eventRTPType-101) |SIP Status | |(5060) <------------------ (5060) | |10398.746| RTP (g729) |RTP Num packets:1 Duration:0.000s SSRC:0x468D9F03 | |(14594) <------------------ (12310) | |10398.746| ACK | |SIP Request | |(5060) ------------------> (5060) | |10398.746| BYE | |SIP Request | |(5060) ------------------> (5060) | |10398.763| 200 OK | |SIP Status | |(5060) <------------------ (5060) | -- _____________________________________________________________________ -- 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
