[asterisk-users] help with rotating number plan

Steve Totaro stotaro at totarotechnologies.com
Thu May 8 16:21:19 CDT 2008


I  second the queues idea.  You can make static queues including the
sip channels.  The previous mentioned ideas, while they may work, are
a little more intricate than then the queue idea.

I think that if you do not need 1.4, 1.2 has much less bugs than 1.4.
This conclusion is not from experience but from bug reports.

Thanks,
Steve Totaro

On Thu, May 8, 2008 at 2:56 PM, Paul Belanger <pabelanger at gmail.com> wrote:
> 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, info at tripple-o.nl <info at tripple-o.nl> 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
>



More information about the asterisk-users mailing list