[Asterisk-Users] call waiting disable in sip

Anton Yurchenko phila at dg.net.ua
Mon Dec 1 06:33:50 MST 2003


Paul Liew wrote:

>----- Original Message ----- 
>From: "Anton Yurchenko" <phila at dg.net.ua>
>To: <asterisk-users at lists.digium.com>
>Sent: Saturday, November 29, 2003 3:34 AM
>Subject: Re: [Asterisk-Users] call waiting disable in sip
>
>
>
>  
>
>>what would happend if all operators are busy? would app_queue exit?
>>would it schedule the call to wait and until the number of them reaches
>>the maxlen ( it is defined in queues.conf) ?
>>
>>    
>>
>
>Hi Anton,
>
>Before I submitted the patch to bugtracker to fix this problem, I tested
>this for both the Dial and Queue apps, and it works as per other channels,
>ie when all the queue operators are busy,  the calling party will stay in
>the queue until an agent becomes free. All parameters within the queue.conf
>apply.
>
>The only parameter you need to specify in sip.conf is the "incominglimit"
>for this to work. For GS phones, set this to 1.
>
>By the way, this is no longer a patch as it has been incorporated into the
>CVS as of 26/11/03.
>
>Let me know if you encounter any problems.
>
>Paul
>
>  
>
I have a problem, when caller is in Queue and the operator is busy 
answering other call he/she still hears the call waiting signal.
I have the latest cvs and incominglimit is set to 1. But here is what * 
shows when the operator is answering ( that is his phone is busy):

Username        incoming        Limit           outgoing        Limit
107             0               1               0               1

and operator is getting a call waiting tone.
Coould I be missing something?

here is my sip.conf:

[107]
type=friend
host=dynamic
dtmfmode=rfc2833                ; Choices are inband, rfc2833, or info
defaultip=172.22.0.137
mailbox=201             ; Mailbox for message waiting indicator
callerid="ipphone1" <201>
callgroup=1
pickupgroup=1
incominglimit=1
outgoinglimit=1

extensions.conf is very simple. it just calls Queue:

exten => 101, 1, Queue(phila)


may I be missing something in granstream phones?

Thanks a lot

-- 

Anton Yurchenko<phila at dg.net.ua>
Digital Generation





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