[Asterisk-Users] Using * for large VoIP implementation

[Jess Magnaye]

I have it on my network but it is still on a day-to-day evaluation case for
me.  Sometimes the quality is not at its best, probably because I am only
using 800MHz CPU running SIP, Digium, T1-PRI, with g729 licenses. Maybe if
you could spare time to put it on a high-end machine and evaluate, you'll
find better results.  Just take note that echo-cancellation, etc. of * is
based on software (and is CPU intensive).  Unlike existing VoIP gateways on
the market, they have hardware chips (DSPs).  Plus, most users of * is g711.
Finding support for G729 is difficult though.  For calling card application,
it will be quite successful, as number of users for calling card is not as
many as for wholesale.



----- Original Message ----- 
From: "John Matte" <rwenzorione at hotmail.com>
To: <asterisk-users at lists.digium.com>
Sent: Saturday, February 21, 2004 11:32 AM
Subject: [Asterisk-Users] Using * for large VoIP implementation


> Hi,
>
> I've searched the archives and can't find a satisfactory answer to this
> question: Does anybody out there have experience using * in large VoIP
> deployments? That is, switching large number of minutes from PSTN to VoIP
> and vice versa? How stable is it, what are the user experiences. I know
> Voicepulse and a few others do that, but I'd be interested to hear the
> experiences, what hardware is in use (e.g. who is using digium T1/E1
cards).
>
> Would greatly appreciate input on this
>
> Thanks
>
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