[Asterisk-Users] Hardware dimesioning issues

[Cameron Beattie]

Have a look at the wiki and although it doesn't provide specifics, 
transcoding is the biggest CPU hit. Since you can control that (i.e. make 
sure the codec is the same on both sides) I doubt you need much grunt at 
all.

Sorry I can't be more specific.

Regards

Cameron
----- Original Message ----- 
From: "David John Walsh" <davidjohnwalsh at gmail.com>
To: <asterisk-users at lists.digium.com>
Sent: Saturday, April 09, 2005 9:59 PM
Subject: [Asterisk-Users] Hardware dimesioning issues


> Hello
>
> I am in the process of putting together a short term calling card
> solution that is rapidly deployable for charity events, and would
> apreciate some guidence on hardware dimensioning for the solution
>
> I have a test system running on an old P3 laptop, so in principle the
> solution works : It is configured as follows:
>
> Latest CVS of asterisk (well as of about 3 weeks ago)
> AreskiCC as the card solution
> Latest RPM of PostgreSQL
> Latest RPM of apache
> Latest RPM of php / pgphp
> 4 SIP accounts for the phones
> 1 SIP account with 4 concurrent calls for the "lines"
> Sipura 1001's as the ATA, DTMF phones on the end.
>
> It has a simple extension.conf
>
> User dials 7777 -> runs DeadAGI(Areskicc.php)
> User goes on to enter PIN, phone number and then is connected (subject
> to credit and b-number being availible)
>
> The only difference between this test system and the production system
> is the number of lines.  I need it to be able to run 80 extensions and
> therefor 80 lines (presented by SIP)
>
> How large should the processor, memory etc be - could anyone suggest a
> Dell / similar system that would be good for our needs.
>
> I don't need any zaptel hardware, as the places this is going to (its
> intended to be movable - not mobile per-se but movable) will only have
> outside internet connections, a local SIP provider is helping us which
> is why its SIP both sides.
>
> Thank you for your time on this matter
>
> David
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