[asterisk-bugs] [Asterisk 0017210]: Asterisk consumes 100% CPU, high interupt load, calls stay at ringing state

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=17210 
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Reported By:                atopo
Assigned To:                
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Project:                    Asterisk
Issue ID:                   17210
Category:                   General
Reproducibility:            have not tried
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.30 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2010-04-20 09:32 CDT
Last Modified:              2010-04-20 10:54 CDT
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Summary:                    Asterisk consumes 100% CPU, high interupt load,
calls stay at ringing state
Description: 
Two servers that have been recently ugraded from Asterisk 1.2 to Asterisk
1.4.30 show the same problem:

Randomly the cpu usage climbs up to 2 x 100%, and calls are no longer
handled. The problem appears every couple of hours and can only be resolved
by restarting asterisk.

The servers are acting as gateways between E1 ISDN Primary lines
(incoming) and IAX connections (outgoing), with an average load of 30
Channels in use.

One server is equipped with a digium 410P card and Dahdi 2.3, the other
with a Sangoma 4port card and Dahdi 2.2.1.1, and a current libpri.
Signalling is pri_net for all channels.

Base system is a debian 5.0.4 with current kernel.

During the high cpu load we noticed a high rate of "local timer
interupts", according to munin at a rate of 400000 Interrupts per second.
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 (0120640) pabelanger (manager) - 2010-04-20 10:54
 https://issues.asterisk.org/view.php?id=17210#c120640 
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We are going to need debug logs to see what is going on.

(http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt)
---
Thank you for taking the time to report this bug and helping to make
Asterisk better. 

Unfortunately, we cannot work on this bug because your description did not
include enough information. 

You may find it helpful to read the Asterisk Issue Guidelines
http://www.asterisk.org/developers/bug-guidelines. 

We\'d be grateful if you would then provide a more complete description of
the problem.

At a minimum, we need:
1. the specific steps or actions you took that caused you to encounter the
problem,
2. the behavior you expected, and
3. the behavior you actually encountered (in as much detail as possible).

This likely includes output from the console with debug level logging, a
SIP trace (if this is SIP related), and configuration information such as
dialplan (e.g. extensions.conf) and channel configuration (e.g. sip.conf).

Thanks! 

Issue History 
Date Modified    Username       Field                    Change               
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2010-04-20 10:54 pabelanger     Note Added: 0120640                          
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