A A ANEES-RJD876
2009-Nov-19 05:34 UTC
[asterisk-users] SIP Calls on Asterisk fails after 25000 calls
Hi, I am trying to use asterisk open source version(asterisk-1.6.0.5) with MySQL (using res_odbc)support for extensions and users list. The call rate is 7 calls / second and each call stays for 120 seconds. after making 25000 successful calls , calls started failing with following message on CLI. [Nov 11 08:50:04] WARNING[2258]: app_dial.c:1502 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) [Nov 11 08:50:04] WARNING[2259]: app_dial.c:1502 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) Is there any configuration parameters I missed out ? please provide your valuable suggestions on the same. Regards Anees -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091119/e675bc38/attachment.htm
Ioan Indreias
2009-Nov-20 06:52 UTC
[asterisk-users] SIP Calls on Asterisk fails after 25000 calls
Hi Anees, Have you tried to monitor the number of active channels? Something like: watch 'asterisk -rx "show channels" | grep active' According with your setup the maximum number of active calls should be 7x120=840 - or near this number. Maybe the calls are not closed properly and you reached some limitations (like the maximum number of open files, etc) HTH, Ioan www.modulo.ro On Thu, Nov 19, 2009 at 7:34 AM, A A ANEES-RJD876 <anees at motorola.com>wrote:> Hi, > > I am trying to use asterisk open source version(asterisk-1.6.0.5) with > MySQL (using res_odbc)support for extensions and users list. > > The call rate is 7 calls / second and each call stays for 120 seconds. > after making 25000 successful calls , calls started > failing with following message on CLI. > > *[Nov 11 08:50:04] WARNING[2258]: app_dial.c:1502 dial_exec_full: Unable > to create channel of type 'SIP' (cause 20 - Unknown) > [Nov 11 08:50:04] WARNING[2259]: app_dial.c:1502 dial_exec_full: Unable to > create channel of type 'SIP' (cause 20 - Unknown)* > ** > ** > Is there any configuration parameters I missed out ? please provide your > valuable suggestions on the same. > > Regards > Anees > ** > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091120/24aa5023/attachment.htm