[asterisk-dev] How to get to 10.000 open calls

[Alex Balashov]

How on earth is it going to work exactly like OpenSIPS?  OpenSIPS is a 
comparatively lightweight proxy, not a PBX application server.  It 
doesn't even a small fraction of the stuff Asterisk has in its event 
loop and/or processing core.

There is virtually zero imaginable correlation between how OpenSIPS and 
Asterisk work -- either from a conceptual perspective, or (especially) 
anatomically.

Venefax wrote:

> If it does not work, can somebody make it work for a fee? It should work
> exactly like in OpenSIP.
> 
> -----Original Message-----
> From: asterisk-dev-bounces at lists.digium.com
> [mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Alex Balashov
> Sent: Wednesday, April 22, 2009 3:02 AM
> To: Asterisk Developers Mailing List
> Subject: Re: [asterisk-dev] How to get to 10.000 open calls
> 
> Also, directrtpsetup still doesn't work AFAIK.
> 
> Venefax wrote:
> 
>> I am using 1.6.2 and directrtp=yes. I need to scale to 10.000 open calls 
>> on a box with 1288 GB or RAM and 16 Cores. Is there any modification to 
>> the source code that would be obvious, any bottlenecks? I will never to 
>> transcoding and the media should, theoretically, flow outside. I have 15 
>> IP addresses already configured in the same box, on two different nics, 
>> to spread the interrupts. Is this a dream or will this work with some 
>> tweaking?
>>
>>  
>>
>> F. Alves
>>
>>
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>
>> asterisk-dev mailing list
>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-dev
> 
> 


-- 
Alex Balashov
Evariste Systems
Web    : http://www.evaristesys.com/
Tel    : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775