[asterisk-users] SIP interface status and calllimit

[James Fromm]

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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