<div>Hi, </div><div> </div><div>You can check extension status using chanisavail function. And extension is not free, you can divert your call to queue.</div><div> </div><div> </div><div><a href="http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanIsAvail">http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanIsAvail</a></div>
<div> </div><div>Regards,</div><div> </div><div>Bharat Lalcheta<br><br></div><div class="gmail_quote">On Thu, Apr 11, 2013 at 1:38 AM, Tommy Cooper <span dir="ltr"><<a href="mailto:tomcooper83@yahoo.com" target="_blank">tomcooper83@yahoo.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;padding-left:1ex;border-left-color:rgb(204,204,204);border-left-width:1px;border-left-style:solid"><div><div style="font-family:times new roman,new york,times,serif;font-size:12pt">
<div> <var></var>Hi, <br><br>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.<br>
</div>
<div style="color:rgb(40,98,197);font-family:times new roman,new york,times,serif;font-size:16px;font-style:normal;background-color:transparent"><font color="#000000">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?<br>
<br></font></div>
<div style="color:rgb(40,98,197);font-family:times new roman,new york,times,serif;font-size:16px;font-style:normal;background-color:transparent"><font color="#000000">extensions.conf</font></div>
<div style="font-family:times new roman,new york,times,serif;font-size:16px;font-style:normal;background-color:transparent">[from-myprovider]</div>
<div>exten => *DID number*,1,Answer<br>exten => *DID number*,2,Dial(SIP/1000)<br>exten => *DID number*,3,Queue(support) ;not sure if this line belongs here<br>exten => *DID number*,4,Hangup</div>
<div> </div>
<div>queues.conf</div>
<div> </div>
<div>[general]<br>[support]<br><br>musicclass=default<br>strategy=rrmemory<br>joinempty=no<br>leavewhenempty=yes<br>ringinuse=no<br>Member => SIP/1000<br>Member => SIP/1001<br><br>agent => 1000,1000</div>
<div>agent => 1001,1001</div>
<div> </div>
<div>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<br></div></div></div><br>--<br>
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