[asterisk-users] SIP interface status and calllimit

[James Fromm]

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.
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>