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

2008-09-26 Thread 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
<[EMAIL PROTECTED]>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 <
>> [EMAIL PROTECTED] > 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
>><[EMAIL PROTECTED] 
>>>
>>>> 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
>>
>>
>>
>> 
>>
>>   ___
>>   -- Bandwidth and Colocation Provided by
>>   http://www.api-digital.com 
>> --
>>
>>
>>   AstriCon 2008 - September 22 - 25 Phoenix, Arizona
>>   Register Now: http://www.astricon.net
>> 
>>
>>
>>   asterisk-users mailing list
>>   To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>>
>>   --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 :

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

2008-09-26 Thread Alex Balashov 
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 
> <[EMAIL PROTECTED] > 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
> <[EMAIL PROTECTED] 
>  >> 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
> 
> 
>  
>  
> 
> 
>___
>-- Bandwidth and Colocation Provided by
>http://www.api-digital.com 
>  --
> 
> 
>AstriCon 2008 - September 22 - 25 Phoenix, Arizona
>Register Now: http://www.astricon.net
>  
> 
> 
>asterisk-users mailing list
>To UNSUBSCRIBE or update options visit:
>  http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
> 
>--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

__

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

2008-09-26 Thread Haider Raza 
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
<[EMAIL PROTECTED]>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 <
>> [EMAIL PROTECTED] > 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
>>
>>
>>
>>  
>>
>>___
>>-- Bandwidth and Colocation Provided by
>>http://www.api-digital.com  --
>>
>>AstriCon 2008 - September 22 - 25 Phoenix, Arizona
>>Register Now: http://www.astricon.net 
>>
>>asterisk-users mailing list
>>To UNSUBSCRIBE or update options visit:
>>  http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>>
>>--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
>
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

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

2008-09-26 Thread Alex Balashov 
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 
> <[EMAIL PROTECTED] > 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
> 
> 
> 
> 
> 
> ___
> -- Bandwidth and Colocation Provided by
> http://www.api-digital.com  --
> 
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net 
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
> 
> -- 
> 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

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

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

2008-09-26 Thread Haider Raza 
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
<[EMAIL PROTECTED]>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
>>
>>
>> 
>>
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
>> Register Now: http://www.astricon.net
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> --
> Alex Balashov
> Evariste Systems
> Web: http://www.evaristesys.com/
> Tel: (+1) (678) 954-0670
> Direct : (+1) (678) 954-0671
> Mobile : (+1) (706) 338-8599
>
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

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

2008-09-26 Thread Alex Balashov 
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
> 
> 
> 
> 
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users


-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

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

2008-09-26 Thread Haider Raza 
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
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
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   http://lists.digium.com/mailman/listinfo/asterisk-users