[asterisk-users] SIP interface status and calllimit

[James Fromm]

Nevermind, I found it.  I'll put up an SVN version in my dev environment 
today.

Thanks.

James Fromm wrote:
> 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
> 
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