| Summary: | ASTERISK-15981: Asterisk consumes 100% CPU, high interupt load, calls stay at ringing state | ||
| Reporter: | Alexander Topolanek (atopo) | Labels: | |
| Date Opened: | 2010-04-20 09:32:59 | Date Closed: | 2011-06-07 14:04:50 |
| Priority: | Minor | Regression? | No |
| Status: | Closed/Complete | Components: | General |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ||
| 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. | ||
| Comments: | By: Paul Belanger (pabelanger) 2010-04-20 10:54:16 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! By: Leif Madsen (lmadsen) 2010-04-26 11:28:14 If you use a version slightly older than 1.4.30 (like say 1.4.27.1) do you have the same issues? There is very little to go on in this bug report. For example, there is no information on how to reproduce this issue. By: Paul Belanger (pabelanger) 2010-05-12 13:05:06 Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested. Further information can be found at http://www.asterisk.org/developers/bug-guidelines | ||