I do link the idea of have a queue answer the calls and route to the extensions, but will have to figure out a way to do this with have the SIP extensions logging into the queues.
On Thu, May 8, 2008 at 1:53 PM, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > An option to rotate between numbers is to add a queue to the system and add > 1111 and 2222 as agents and pick the proper strategy (rrmemory > or leastrecent). This has some advantages: > - the calls are devided as you have in mind > - when there are more calls coming in they are queued instead of a busy > tone > - you can scale by just adding an agent to the queue > see http://www.voip-info.org/wiki-Asterisk+call+queues for further info > > Erik de Wild > Tripple-o > Your Asterisk migration partner > > I'm trying to come up with a quick, easy solution to have a static > > inbound number in my dialplan, rotate calling 2 numbers. Example: > > > 1st call into asterisk > > exten => 1234,1,Dial(sip/1111,10) > > exten => 1234,n,Dial(sip/2222,10) > > 2nd call into asterisk > > exten => 1234,1,Dial(sip/2222,10) > > exten => 1234,n,Dial(sip/1111,10) > > We're kind off looking to do load balancing via the dial plan. > > But I'm having a little trouble getting the logic to trace 1st call > > in, 2nd call in, 1st call in, 2nd call in, etc. > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users