[asterisk-users] Didn't get a frame from channel

Sergio de los Santos ssantos at hispasec.com
Tue Jan 16 02:01:29 MST 2007


Using tdm400. While transfering a call from outside to another
extensions, while this "outside call" is waiting with music, the
"another extension" call hangs up suddenly, and the call is back to the
"outside call" suddenly.

Wathcing logs:

Jan 15 13:32:44 DEBUG[30148] res_musiconhold.c: Read 462 bytes of audio
while expecting 640
Jan 15 13:32:55 DEBUG[27850] channel.c: Didn't get a frame from channel:
SIP/219-081d4d60
Jan 15 13:32:55 DEBUG[27850] channel.c: Bridge stops bridging channels
SIP/219-081d4d60 and Zap/1-1
Jan 15 13:32:55 DEBUG[27850] chan_zap.c: Hangup: channel: 1 index = 0,
normal = 16, callwait = -1, thirdcall = -1
Jan 15 13:32:55 DEBUG[27850] chan_zap.c: disabled echo cancellation on
channel 1
Jan 15 13:32:55 DEBUG[27850] chan_zap.c: Set option TDD MODE, value:
OFF(0) on Zap/1-1
Jan 15 13:32:55 DEBUG[27850] chan_zap.c: Updated conferencing on 1, with
0 conference users
15 13:32:55 VERBOSE[27850] logger.c:     -- Hungup 'Zap/1-1'
Jan 15 13:32:55 DEBUG[27850] app_dial.c: Exiting with DIALSTATUS=ANSWER.
Jan 15 13:32:55 VERBOSE[27850] logger.c:   == Spawn extension

This may be the cause:

"Didn't get a frame from channel..."

I googled. It is recommended to disable busydetect, but no solution. Any
ideas?


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