[asterisk-users] SIP interface status and calllimit

[Eric "ManxPower" Wieling]

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.