[asterisk-users] Split incoming call volume across queues on several asterisk servers

[Haider Raza]

I will now look into reinvites and openser. Thank you so much for your time
and all the excellent advice.

-- 
Dr. Haider Raza
BM 5203
3508 North West 114 Av.
Doral, Florida 33178

Mobile    +(809)-659-0623


On Fri, Sep 26, 2008 at 11:59 PM, Alex Balashov
<abalashov at evaristesys.com>wrote:

> 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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