[asterisk-biz] SIP to PSTN Hardware

[Steve Totaro]

Alex Balashov wrote:
> On Sun, 5 Aug 2007, Arya wrote:
>
>   
>> lets say if you want to start with 5 concurrent calls than grow to as 
>> 200 or more concurrent calls
>>     
>
>    A few Asterisk boxes supported by a proxy should be able to handle
> that in terms of transcoding and call volume alone.  Where the limitation
> is going to be terminating 200 TDM calls at once.  That would be 8.69 
> PRIs.  You might be able to swing it with Asterisk and PCs and quad-span
> T1 cards, provided they can actually operate concurrently in practice as
> well as advertised in theory, without lockups, etc.
>
>    Otherwise, some sort of outside media gateway like a Cisco AS would be 
> your ticket.  Cheaper solutions available also, but I do not have much
> familiarity with these.
>
>   

This is trivial.  Setup two front end servers (I like HP DL320s) each 
with a Sangoma quad port T1, then take the TDM<->SIP and send it to 
another Asterisk server running whatever apps you want. 

I have terminated a whole DS3 this way.

Thanks,
Steve Totaro