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. extensions.conf: [sip-phone] ;This is the context setup for outgoing calls;exten => _NXXXXXXX.,2,Set(CALLERID(name)=*my number*) ;exten => _NXXXXXXX.,3,Set(CALLERID(num)=*my number*) ;exten => _NXXXXXXXXXX,1,Dial(mailto:SIP/$%7bexten...@myprovider.com) ;exten => _3XXXX.,1,Answer exten => _3XXXX.,1,Dial(SIP/myprovider/${EXTEN:1},60) ;This is the only line that seems to work but the phone hangs up shortly after answering, as described above ;exten => _3XXXX.,5,Hangup ;exten => _X.,1,Answer ;exten => _X.,2,Set(CALLERID(name)=*my number*) ;exten => _X.,3,Set(CALLERID(num)=*my number*) ;exten => _X.,4,Dial(SIP/${EXTEN}@myprovider,30,Tt) ;exten => _X.,5,Hangup
________________________________ From: Salman Zafar <msalman...@gmail.com> To: Tommy Cooper <tomcoope...@yahoo.com>; Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Sent: Wednesday, April 10, 2013 10:27 PM Subject: Re: [asterisk-users] ACD problem 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 <tomcoope...@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 ***************************
-- _____________________________________________________________________ -- 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