asterisk users - Aug 2009 - Platform decision ...

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   * Use of no longer maintained Berkeley DB1 engine as its internal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 S?rgio Ferreira Brasil
Coordenador de Projetos e Analista de Sistemas
+ mauro.brasil at tqi.com.br <mailto:@tqi.com.br>
: www.tqi.com.br <http://www.tqi.com.br> ( + 55 (34)3291-1700 ( + 55
(34)9971-2572


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

AstriCon 2009 - October 13 - 15 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

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 S?rgio Ferreira Brasil
Coordenador de Projetos e Analista de Sistemas
+ mauro.brasil at tqi.com.br <mailto:@tqi.com.br>
: www.tqi.com.br <http://www.tqi.com.br>
( + 55 (34)3291-1700
( + 55 (34)9971-2572

On Tue, Aug 18, 2009 at 1:52 PM, Mauro Sergio Ferreira Brasil <
mauro.brasil at tqi.com.br> wrote:
> 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 S?rgio Ferreira Brasil
> Coordenador de Projetos e Analista de Sistemas
> + mauro.brasil at tqi.com.br <mailto:@tqi.com.br>
> : www.tqi.com.br <http://www.tqi.com.br>
> ( + 55 (34)3291-1700
> ( + 55 (34)9971-2572
>
>
Don't forget to add FreeSwitch to your comparison chart too.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20090818/56186ba0/attachment.htm