Michael Stahl
2005-May-15 20:21 UTC
[Asterisk-Users] Hang up error: Didn't get a frame from channel
I'm using EyeBeam from xten, and whenever I call another user, the callee phone rings but my SIP phone immediately hangs up. The other end keeps on ringing but when the callee answers, there is no sounds. I have found the "Didn't get frame from channel" error occurring in each such call. What does this mean? How can I fix it? -Mike- May 15 22:31:10 DEBUG[4792]: sip_answer(SIP/2433-9716) May 15 22:31:10 VERBOSE[4792]: -- Attempting native bridge of SIP/2433-9716 and SIP/2463-2f7a May 15 22:31:10 DEBUG[4792]: Got RTCP report of 84 bytes May 15 22:31:10 DEBUG[4792]: Ooh, format changed from unknown to ulaw May 15 22:31:10 DEBUG[4792]: Got RTCP report of 118 bytes May 15 22:31:15 DEBUG[4792]: Got RTCP report of 84 bytes May 15 22:31:17 DEBUG[4792]: Didn't get a frame from channel: SIP/2463-2f7a May 15 22:31:17 DEBUG[4792]: Bridge stops bridging channels SIP/2433-9716 and SIP/2463-2f7a May 15 22:31:17 DEBUG[4792]: Hanging up channel 'SIP/2463-2f7a' May 15 22:31:17 DEBUG[4792]: sip_hangup(SIP/2463-2f7a) May 15 22:31:17 DEBUG[4792]: update_user_counter(2463) - decrement outUse counter May 15 22:31:17 DEBUG[4792]: Exiting with DIALSTATUS=ANSWER. May 15 22:31:17 DEBUG[4792]: Exiting with ANSWERTIME=7. May 15 22:31:17 DEBUG[4792]: Spawn extension (macro-stdexten,s,4) exited non-zero on 'SIP/2433-9716' in macro 'stdexten' May 15 22:31:17 DEBUG[4792]: Spawn extension (menuinternal,2460,1) exited non-zero on 'SIP/2433-9716' May 15 22:31:17 DEBUG[4792]: Launching 'Goto' May 15 22:31:17 VERBOSE[4792]: -- Executing Goto("SIP/2433-9716", "menuinternal|i|1") in new stack May 15 22:31:17 VERBOSE[4792]: -- Goto (menuinternal,i,1) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050515/9eeb24b6/attachment.htm