Hello,
I have upgraded to 1.0.4 version of asterisk. After that asterisk crash
every time
On receiving an call from iax2 trunk to musiconhold application. SIP
calls to
MusicOnHold is however working. I already upgraded to 1.0.5, but the
problem still
Remainig.
Any idea ?
Iax2 : call proceding :
Jan 25 17:29:40 DEBUG[9997]: pbx.c:1261 pbx_extension_helper: Launching
'WaitMusicOnHold'
-- Executing WaitMusicOnHold("IAX2/radko@radko/3",
"201") in new
stack
Jan 25 17:29:40 DEBUG[9997]: channel.c:1551 ast_prod: Prodding channel
'IAX2/radko@radko/3'
Urgent handler
Ouch ... error while writing audio data: : Broken pipe
Sip : call proceding :
Jan 25 17:34:04 DEBUG[10020]: pbx.c:1261 pbx_extension_helper: Launching
'WaitMusicOnHold'
-- Executing WaitMusicOnHold("SIP/192.168.1.38-082257a0",
"201") in
new stack
Jan 25 17:34:04 DEBUG[10020]: channel.c:1551 ast_prod: Prodding channel
'SIP/192.168.1.38-082257a0'
Jan 25 17:34:04 DEBUG[10020]: channel.c:1707 ast_set_write_format: Set
channel SIP/192.168.1.38-082257a0 to write format slin
-- Started music on hold, class 'default', on
SIP/192.168.1.38-082257a0
Jan 25 17:34:04 DEBUG[10020]: channel.c:1128 ast_settimeout: Scheduling
timer at 160 sample intervals
Urgent handler
Jan 25 17:34:04 DEBUG[10020]: channel.c:1379 ast_read: Generator got
voice, switching to phase locked mode
Jan 25 17:34:04 DEBUG[10020]: channel.c:1128 ast_settimeout: Scheduling
timer at 0 sample intervals
Jan 25 17:34:04 DEBUG[10020]: rtp.c:1188 ast_rtp_write: Ooh, format
changed from unknown to alaw
Radovan
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