Roger Schreiter
2007-Dec-17 17:49 UTC
[asterisk-users] SIP call interrupted after 64 seconds
Hi, some months ago, I had the problem with an asterisk-1.4.x- Version, that some calls (but not all) were interrupted 64 seconds after connect (a call limit of 86400 seconds was installed using the S()-parameter). It was just a test machine, and later, I switched to callweaver, and the problem had gone. Thus, I never investigated this problem. Now, I upgraded a machine for production use to asterisk-1.4.8, and do encounter the same problem. I have other asterisk machines running, using the same dialplan, without this problem. Did anyone else observe this strange behaviour of calls ending after 64 secondes of uptime? My os is Suse-Linux 10.2. Thanks for any hints! Roger.
Jaswinder Singh
2007-Dec-17 18:30 UTC
[asterisk-users] SIP call interrupted after 64 seconds
Can you post the part of your dialplan which causes this behaviour ? On Dec 17, 2007 11:19 PM, Roger Schreiter <roger at planinternet.de> wrote:> Hi, > > some months ago, I had the problem with an asterisk-1.4.x- > Version, that some calls (but not all) were interrupted > 64 seconds after connect (a call limit of 86400 seconds > was installed using the S()-parameter). > > It was just a test machine, and later, I switched to callweaver, > and the problem had gone. Thus, I never investigated this problem. > > Now, I upgraded a machine for production use to asterisk-1.4.8, > and do encounter the same problem. > > I have other asterisk machines running, using the same > dialplan, without this problem. > > Did anyone else observe this strange behaviour of calls ending > after 64 secondes of uptime? > > My os is Suse-Linux 10.2. > > > Thanks for any hints! > Roger. > > > _______________________________________________ > --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 >
Olle E Johansson
2007-Dec-17 19:20 UTC
[asterisk-users] SIP call interrupted after 64 seconds
17 dec 2007 kl. 18.49 skrev Roger Schreiter:> Hi, > > some months ago, I had the problem with an asterisk-1.4.x- > Version, that some calls (but not all) were interrupted > 64 seconds after connect (a call limit of 86400 seconds > was installed using the S()-parameter). > > It was just a test machine, and later, I switched to callweaver, > and the problem had gone. Thus, I never investigated this problem. > > Now, I upgraded a machine for production use to asterisk-1.4.8, > and do encounter the same problem. > > I have other asterisk machines running, using the same > dialplan, without this problem. > > Did anyone else observe this strange behaviour of calls ending > after 64 secondes of uptime?There is a hidden reason somewhere and you need to add verbose logging to your Asterisk, maybe also debug logging so that you can find out what's going on - where the call fails. With the log files, it's often very simple for a trained eye to spot what goes on. It seems like some kind of signalling problem as it is kind of close to the SIP timeouts. If you think it is a bug, don't hesitate to file a bug report and add your log output with verbose set to 4 and debug set to 4, sip debug also turned on! /Olle