On 13-04-10 04:08 PM, Tommy Cooper 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?

Forget dialling the phones directly, let the queue deal with it. Dump everything in to the queue, then just wait for somebody to answer.


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



--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

--
_____________________________________________________________________
-- 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

Reply via email to