<html><body><div style="color:#000; background-color:#fff; font-family:times new roman, new york, times, serif;font-size:12pt"><div style="RIGHT: auto"><SPAN style="RIGHT: auto">Thank you for your prompt reply. I removed that redundant line and now everything seems to work fine. Except outgoing calls that is, whenever i try to call an outside number the phone rings, the user can even answer back but then it hangs up after about 5 sec.</SPAN></div>
<div style="BACKGROUND-COLOR: transparent; FONT-STYLE: normal; FONT-FAMILY: times new roman, new york, times, serif; COLOR: rgb(0,0,0); FONT-SIZE: 16px; RIGHT: auto"><SPAN style="RIGHT: auto"></SPAN> </div>
<div style="RIGHT: auto"><SPAN style="RIGHT: auto">extensions.conf:</SPAN></div>
<div style="RIGHT: auto"><SPAN style="RIGHT: auto">[sip-phone]<BR>;This is the context setup for outgoing calls</SPAN></div><SPAN style="RIGHT: auto">
<div style="RIGHT: auto">;exten => _NXXXXXXX.,2,Set(CALLERID(name)=*my number*)<BR>;exten => _NXXXXXXX.,3,Set(CALLERID(num)=*my number*)<BR>;exten => _NXXXXXXXXXX,1,Dial(<A style="RIGHT: auto" href="mailto:SIP/$%7BEXTEN%7D@myprovider.com">mailto:SIP/$%7BEXTEN%7D@myprovider.com</A>)</div>
<div style="RIGHT: auto"> </div>
<div style="RIGHT: auto">;exten => _3XXXX.,1,Answer<BR>exten => _3XXXX.,1,Dial(SIP/myprovider/${EXTEN:1},60) ;This is the only line that seems to work but the phone hangs up shortly after answering, as described above</div>
<div style="RIGHT: auto">;exten => _3XXXX.,5,Hangup</div>
<div style="RIGHT: auto"> </div>
<div style="RIGHT: auto">;exten => _X.,1,Answer<BR>;exten => _X.,2,Set(CALLERID(name)=*my number*)<BR>;exten => _X.,3,Set(CALLERID(num)=*my number*)<BR>;exten => _X.,4,Dial(<A style="RIGHT: auto" href="mailto:SIP/$%7BEXTEN%7D@myprovider,30,Tt">SIP/${EXTEN}@myprovider<VAR id=yui-ie-cursor></VAR>,30,Tt</A>)<BR>;exten => _X.,5,Hangup<BR></SPAN></div>
<div><BR></div>
<DIV style="FONT-FAMILY: times new roman, new york, times, serif; FONT-SIZE: 12pt">
<DIV style="FONT-FAMILY: times new roman, new york, times, serif; FONT-SIZE: 12pt">
<DIV dir=ltr><FONT size=2 face=Arial>
<DIV style="BORDER-BOTTOM: #ccc 1px solid; BORDER-LEFT: #ccc 1px solid; PADDING-BOTTOM: 0px; LINE-HEIGHT: 0; MARGIN: 5px 0px; PADDING-LEFT: 0px; PADDING-RIGHT: 0px; HEIGHT: 0px; FONT-SIZE: 0px; BORDER-TOP: #ccc 1px solid; BORDER-RIGHT: #ccc 1px solid; PADDING-TOP: 0px" class=hr contentEditable=false readonly="true"></DIV><B><SPAN style="FONT-WEIGHT: bold">From:</SPAN></B> Salman Zafar <msalman212@gmail.com><BR><B><SPAN style="FONT-WEIGHT: bold">To:</SPAN></B> Tommy Cooper <tomcooper83@yahoo.com>; Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> <BR><B><SPAN style="FONT-WEIGHT: bold">Sent:</SPAN></B> Wednesday, April 10, 2013 10:27 PM<BR><B><SPAN style="FONT-WEIGHT: bold">Subject:</SPAN></B> Re: [asterisk-users] ACD problem<BR></FONT></DIV><BR>
<META content=off http-equiv=x-dns-prefetch-control>
<DIV id=yiv1030603403>
<DIV dir=ltr>
<DIV>This line :<BR>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.<BR><BR></DIV>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.<BR><BR><BR></DIV>
<DIV class=yiv1030603403gmail_extra><BR><BR>
<DIV class=yiv1030603403gmail_quote>On Thu, Apr 11, 2013 at 1:08 AM, Tommy Cooper <SPAN dir=ltr><<A href="mailto:tomcooper83@yahoo.com" rel=nofollow target=_blank ymailto="mailto:tomcooper83@yahoo.com">tomcooper83@yahoo.com</A>></SPAN> wrote:<BR>
<BLOCKQUOTE style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class=yiv1030603403gmail_quote>
<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="BACKGROUND-COLOR: transparent; FONT-STYLE: normal; FONT-FAMILY: times new roman, new york, times, serif; COLOR: rgb(40,98,197); FONT-SIZE: 16px"><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="BACKGROUND-COLOR: transparent; FONT-STYLE: normal; FONT-FAMILY: times new roman, new york, times, serif; COLOR: rgb(40,98,197); FONT-SIZE: 16px"><FONT color=#000000>extensions.conf</FONT></DIV>
<DIV style="BACKGROUND-COLOR: transparent; FONT-STYLE: normal; FONT-FAMILY: times new roman, new york, times, serif; FONT-SIZE: 16px">[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>_____________________________________________________________________<BR>-- Bandwidth and Colocation Provided by http://www.api-digital.com --<BR>New to Asterisk? Join us for a live introductory webinar every Thurs:<BR> http://www.asterisk.org/hello<BR><BR>asterisk-users mailing list<BR>To UNSUBSCRIBE or update options visit:<BR> http://lists.digium.com/mailman/listinfo/asterisk-users<BR></BLOCKQUOTE></DIV><BR><BR clear=all><BR>-- <BR><FONT face="'times new roman', serif">Regards</FONT>
<DIV><FONT face="'times new roman', serif"><BR></FONT>
<DIV><PRE><FONT face="'times new roman', serif">**************************
Muhammad Salman
***************************</FONT>
</PRE></DIV></DIV></DIV></DIV>
<META content=on http-equiv=x-dns-prefetch-control><BR><BR></DIV></DIV></div></body></html>