[asterisk-users] SIP interface status and calllimit

[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.

James Fromm wrote:
> It does.
> 
> Eric "ManxPower" Wieling wrote:
>> Maybe Queue doesn't consider a SIP account that returns "BUSY" as "in 
>> use".  That would be the only case where I could see needing call-limit.
>>
>> James Fromm wrote:
>>> We do the same thing only we use ringinuse=no and autopause=yes for 
>>> the queue.  With autopause, if the agent is busy their interface in 
>>> the queue gets paused.  Setting call-limit for the SIP interface is 
>>> the only way to make ringinuse=no work.
>>>
>>> Eric "ManxPower" Wieling wrote:
>>>> James Fromm wrote:
>>>>> There is an issue when using call-limit for a SIP interface in
>>>>> sip.conf.  The call count does not properly reset when some calls
>>>>> end.  The problem happens regardless of which side of the connection
>>>>> ends the call.  It happens on all calls including calls from SIP
>>>>> interface to SIP interface (with no reinvite) within the same Asterisk
>>>>> server.  I have not been able to determine a definite pattern.  I 
>>>>> can call from one interface to another 50 times before it happens 
>>>>> and sometimes it happens after only 2 calls.
>>>>>
>>>>> We have to enable call-limit for our customer service queue agents 
>>>>> so that the ringinuse option in queues.conf will work properly.
>>>>>
>>>>> Has anyone else seen this issue?  Any ideas?
>>>>
>>>> This doesn't really help you, but might help others when deciding 
>>>> how to design their Asterisk system.  On our phones we set call 
>>>> waiting off and each line appearance registers as a separate SIP 
>>>> user.  This avoids all this silliness with call limits, group 
>>>> limits, etc.  This also allows us total control about which call 
>>>> appearance a call shows up on, roll over and hunting features, etc.  
>>>> It does require a little more work in the dialplan, but for our 
>>>> needs it is well worth it.
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