[asterisk-users] SIP interface status and calllimit

James Fromm fromm at omnis.com
Thu Feb 22 07:47:53 MST 2007


I've reviewed the bugs reports. I didn't see anything that applied to 
this.  Have you?  Could you point it out to me?


Olle E Johansson wrote:
> 
> 21 feb 2007 kl. 15.50 skrev James Fromm:
> 
>> Anybody seen this behavior?
>>
>> To determine if it's my config or a bug, could I trouble someone 
>> running Asterisk 1.4.0 to set call-limit=5 on a moderately busy SIP 
>> interface as a test?  After a few hours a 'sip show inuse' should 
>> indicate the interface is on calls that it isn't. The incorrect count 
>> can be cleared up by ringing the interface for how ever many calls are 
>> incorrect.
>>
>> Beware, removing call-limit will require a restart to take effect. 
>> Thanks in advance for any help.
> 
> A good way to check is to visit the bug tracker at bugs.digium.com
> 
> If you do, you will find a few bug reports and also notice a few that 
> has been resolved in Asterisk 1.4 svn,
> which is the base for the coming 1.4.1 release.
> 
> Please try with latest 1.4 from subversion to test if the behaviour is 
> fixed.
> 
> Thanks,
> /Olle



More information about the asterisk-users mailing list