[asterisk-users] Split incoming call volume across queues on several asterisk servers
Alex Balashov
abalashov at evaristesys.com
Fri Sep 26 22:59:24 CDT 2008
Asterisk is not a SIP proxy. You would have to use another piece of
software, such as Kamailio/OpenSIPS (formerly OpenSER).
Haider Raza wrote:
>
> I guess what I want to ask is...how do I setup a proxy? In a
> nutshell...how are calls transfered or handed off to other asterisk
> servers leaving the originating server free from all call handling once
> the transfer is done. What dialplan command would do that? Do I setup a
> trunk and then Dial the call to the trunk? Maybe write an agi script to
> connect to manager interfaces on the different asterisk servers to see
> who has a spot free on their queue and then transfer on a trunk.
>
> I guess what I am not clear on is, are IAX trunks between asterisk
> servers what I need to accomplish this (Using a proxy or daisy chained
> asterisk servers)?
>
> --
> Dr. Haider Raza
> BM 5203
> 3508 North West 114 Av.
> Doral, Florida 33178
>
> Mobile +(809)-659-0623
>
> On Fri, Sep 26, 2008 at 11:36 PM, Alex Balashov
> <abalashov at evaristesys.com <mailto:abalashov at evaristesys.com>> wrote:
>
> Proxies do not handle media, so, one can definitely handle 300
> simultaneous calls.
>
> Haider Raza wrote:
>
> But will this allow the proxy to handle a load of 300
> simultaneous calls? I mean will the calls be sent off to other
> asterisk servers and the proxy be left load-free to route new calls?
>
> --
> Dr. Haider Raza
> BM 5203
> 3508 North West 114 Av.
> Doral, Florida 33178
>
> Mobile +(809)-659-0623
>
> On Fri, Sep 26, 2008 at 10:37 PM, Alex Balashov
> <abalashov at evaristesys.com <mailto:abalashov at evaristesys.com>
> <mailto:abalashov at evaristesys.com
> <mailto:abalashov at evaristesys.com>>> wrote:
>
> You can set up a proxy to round-robin/load-balance the incoming
> calls across three servers.
>
> If you need to do this with a view to queue utilisation, an
> outside
> process can be set up to mediate this via the Manager API and
> provide this information to the proxy process in real time.
>
> A proxy can also be set up to roll calls over to another Asterisk
> server if that server returns an error status code because
> all the
> agents are unavailable, such as 486 Busy or temporarily
> unavailable.
>
> You can, also, of course, do this in the Asterisk dial plan
> itself -
> fiddle with the timeout values on the Queue() app. However,
> in this
> paradigm, the first Asterisk box is going to have to
> cross-connect
> the call to others in the series, in a daisy chain. But if
> you can
> avoid media handling in such scenarios (i.e. use re-INVITEs),
> that
> shouldn't be too bad.
>
> Haider Raza wrote:
>
> Hi,
> I was wondering if there is anyway to split, say, 300
> calls
> that come in from the SIP provider across 10 asterisk servers
> with 30 agents each, without having the telco do the
> splitting.
> Is there any way to do call distribution, e.g. we send an
> incoming call to a similar queue on the next asterisk
> server if
> all agents on the first asterisk server are busy and the
> queue
> already has a certain number of calls in it?
>
> Thanks,
> -- Dr. Haider Raza
>
>
>
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>
> -- Alex Balashov
> Evariste Systems
> Web : http://www.evaristesys.com/
> Tel : (+1) (678) 954-0670
> Direct : (+1) (678) 954-0671
> Mobile : (+1) (706) 338-8599
>
>
>
>
>
>
> --
> Alex Balashov
> Evariste Systems
> Web : http://www.evaristesys.com/
> Tel : (+1) (678) 954-0670
> Direct : (+1) (678) 954-0671
> Mobile : (+1) (706) 338-8599
>
>
>
>
--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
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