[asterisk-users] Platform decision ...

[Mauro Sergio Ferreira Brasil]

Man... I need to be very frank with you... I don't know any more.

We started analysing what can be done to get Asterisk working on a way 
we want it to work, that is: totally dynamic dial plan generated by an 
external server (responsible for business logic and legacy interface), 
and retrieved through an new configuration driver (something like 
"res_config_legacy.c").
This point is clear to us now that is reachable without much effort.

We considered, at first, a infraestructure with a 
redirect-server/load-balancer (played by OpenSIPS) directing the voip 
calls to final Asterisk instances.
The problem is that after getting the first issue solved (about the 
driver acessing the legacy interface explained above), I started a 
research about Asterisk scalability and I didn't liked of what I found.

Consulting some friends of mine that work with Voip (but that 
unfortunatelly don't need the PBX features) the impression was worst.
One of them told me that on the only part of their infraestructure where 
Asterisk is used they want at all costs to remove it.

Making things short, I need to have sure that Asterisk can handle a 
considerable number of concurrent calls, or I need an indication of 
another PBX that is scalable to be placed on Asterisk's place and that 
can be changed to retrieve the dialplan (or what it uses on call 
routing) from another server.

Does anyone have any idea ?

Thanks and best regards,
Mauro.



C. Savinovich escreveu:
> It all depends what are you going to use Asterisk for.  Sounds like it is
> for conferencing.  Would you care to elaborate?
>
> CS
>
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Mauro Sergio
> Ferreira Brasil
> Sent: Tuesday, August 18, 2009 10:23 PM
> To: asterisk-users at lists.digium.com
> Subject: [asterisk-users] Platform decision ...
>
> Hello there!
>
> During some research on Internet I found the following comparison on site
> Voip-Info (see, "http://www.voip-info.org/wiki/view/OpenPBX.org+FAQ"):
>
> The main points listed on Asterisk's "CONS" that concerned me were:
>
>    * Conferencing on Asterisk depends on Zaptel hardware and/or kernel
> modules for timing;
>    * Lack of built-in STUN support for SIP NAT traversal;
>    * Asterisk doesn't use SpanDSP;
>    * Use of no longer maintained Berkeley DB1 engine as its internal
> database;
>    * Asterisk doesn't allow CSRC entries in RTP;
>    * Asterisk doesn't have an universal jitterbuffer for use with any
> channel type;
>    * Asterisk doesn't use POSIX realtime extensions (having dependency with
> Zaptel timing);
>
> We were considering Asterisk as the chosen platform, but after reading this
> I got a little worried.
> The comparison considers 1.4 old version of Asterisk.
>
> So, can someone give me an update on what have changed for this items
> considering new 1.6 version ?
> Maybe someone can point me a site with an updated comparison.
>
> As long as I could see by now SpanDSP is present on new version of Asterisk,
> so this item isn't a difference any more. Right ?
>
> Thanks and best regards,
>
>   

-- 
__At.,                                                                                                                             
   _
 
*Technology and Quality on Information*
Mauro Sérgio Ferreira Brasil
Coordenador de Projetos e Analista de Sistemas
+ mauro.brasil at tqi.com.br <mailto:@tqi.com.br>
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