? 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130410/07094374/attachment.htm>
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.belanger at polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger
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>
Hi, You can check extension status using chanisavail function. And extension is not free, you can divert your call to queue. http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanIsAvail Regards, Bharat Lalcheta On Thu, Apr 11, 2013 at 1:38 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 >-- Bharat Lalcheta -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130411/08b17f27/attachment-0001.htm>
I am not sure I understand the required routing pattern, but I'm sure queues are your friends, as you can dynamically add and remove member and you can have a first-level queue easily move fall-through to another queue in case all members should be busy or none should be available. Plus by using queues you decouple the "what" you want to do from the "who" is doing it. 2013/4/10 Tommy Cooper <tomcooper83 at yahoo.com>> 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 >-- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.com -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130411/e9117e61/attachment.htm>