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 <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 > -- Bharat Lalcheta
-- _____________________________________________________________________ -- 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