[asterisk-users] ACD problem

Salman Zafar msalman212 at gmail.com
Wed Apr 10 15:27:25 CDT 2013


This line :
exten => *DID number*,2,Dial(SIP/1000)  is redundant and useless when you
are already using Queues. So just remove it and it should work.

What happen is, your dial-plan executes at 2nd priority DIAL a SIP
extension 1000 .. produce a call and at hang-up finishes no Queue/ACD
functionality is executed.




On Thu, Apr 11, 2013 at 1:08 AM, Tommy Cooper <tomcooper83 at yahoo.com> wrote:

>   Hi,
>
> I am working on a small inbound call center solution that uses an ACD
> system. I might add an IVR system later on. I only have 2 extensions set up
> (extensions 1000 and 1001), I want the system to put new calls in a queue
> if both extensions are busy. I am currently subscribed with a SIP trunk
> provider and can successfully recieve calls. I want to design a system
> where customers can call my number, that call will then be directed to
> either extension 1000 or 1001. If both extensions are in use, I want that
> 3rd call to be queued.
> I don't think that the config below will direct calls to extension 1001
> because the second line states that any incomming calls should be routed to
> extension 1000. How do I change this so that calls are directed to all of
> my exensions?
>
> extensions.conf
> [from-myprovider]
> exten => *DID number*,1,Answer
> exten => *DID number*,2,Dial(SIP/1000)
> exten => *DID number*,3,Queue(support) ;not sure if this line belongs here
> exten => *DID number*,4,Hangup
>
> queues.conf
>
> [general]
> [support]
>
> musicclass=default
> strategy=rrmemory
> joinempty=no
> leavewhenempty=yes
> ringinuse=no
> Member => SIP/1000
> Member => SIP/1001
>
> agent => 1000,1000
> agent => 1001,1001
>
> When using the current config the caller will listen to the 'music on
> hold' until the agent answers but calls are only being forwarded to
> extension 1000 as stated above
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Regards

**************************
Muhammad Salman
***************************
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130411/b760061a/attachment.htm>


More information about the asterisk-users mailing list