[Asterisk-Users] Asterisk Server Hangs

[Giovanni Miano]

whats kernel version ?
check in "dmesg" for system messages

Cheers,
Giovanni Miano

2005/12/29, Dushyanth Harinath <dushyanth.h at directi.com>:
>
> Hey guys,
> Asterisk Server Specs :
>
> Asterisk version :
>
> CLI> show version
> Asterisk SVN-trunk-r7230 built by root at asterisk on a i686 running Linux
> on 2005-12-25 16:14:47 UTC
>
> System details :
>
> Centos 4.2 (Final)
> Linux ip-pbx 2.6.9-22.ELsmp #1 SMP
> Intel Dual Xeon 3.06Ghz
> Intel SE7501CW2 Motherboard
>
> Digium cards : T110P (E1) , TDM22B, TDM31B, TDM24012B
>
> I added TDM24012B yes'day but haven't configured or used it yet. Its
> just connected to the system. The same problem used to occur before
> adding TDM24012B to the mix.
>
> This setup hangs up i,e total freeze cant ssh, cant login even from the
> system console and nothing in system logs or asterisk logs point me to
> any obvious problem. There is no coredump in /tmp too.
>
> Asterisk also freezes up the server when i issue a stop now command in
> the CLI sometimes.
>
> The only call traffic at this moment are SIP to SIP internal calls, SIP
> to ZAP external calls and ZAP to SIP incoming calls. In all there must
> be a total of 10 simultaneous calls.
>
> Im using queues, rxfax, txfax, voicemail, meetme (still testing).
>
> This happens three or four times in a day.
>
> I cant see any IRQ misses in zttool and zttest output is below.
>
> Opened pseudo zap interface, measuring accuracy...
> 99.987793% 99.987793% 99.987793% 99.987793% 100.000000% 100.000000%
> 99.987793%
> 99.987793% 100.000000% 100.000000% 100.000000% 99.987793% 100.000000%
> 99.987793% 100.000000%
> ....
> ....
> ....
> Best: 100.000000 -- Worst: 99.987793 -- Average: 99.992300
>
> Found the below messages in dmesg but seems informational rather than a
> error.
>
> Dec 27 22:04:24 asterisk kernel: zaptel Disabled echo canceller because
> of tone (tx) on channel 32
> Dec 29 21:02:12 asterisk kernel: zaptel Disabled echo canceller because
> of tone (rx) on channel 35
>
> I dont know what the problem could be. I followed the doc at
> http://www.voip-info.org/wiki-Asterisk+debugging and started asterisk
> using safe_asterisk and applied the logger related changes.
>
> Wat else i can do to debug this issue ?
>
> Dushyanth
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>



--
Giovanni Miano
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051229/7107d505/attachment.htm