[Asterisk-Users] coredumping on MusicOnHold

Tue, 25 Jan 2005 09:10:17 -0800

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/[EMAIL PROTECTED]/3", "201") in new stack

Jan 25 17:29:40 DEBUG[9997]: channel.c:1551 ast_prod: Prodding channel 'IAX2/[EMAIL PROTECTED]/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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