[asterisk-biz] Routing large call volumes (400 concurrent calls+)

[Jean-Michel Hiver]

Le Thu, 20 Sep 2007 06:22:10 +0400, Chris Bagnall <chris at minotaur.it> a  
écrit:

> Greetings list,
>
> I've recently been approached by a traditional PSTN LCR provider who are  
> trying to break into the VoIP market and set themselves up. They have  
> good contacts with some of the large carriers here in the UK who are  
> happy to install interconnects into the datacentre in which they have  
> rackspace.
>
> They're in a tricky position: they won't be generating anywhere near  
> enough volume to justify one of the big Cisco gateways (or similar), but  
> they will have sufficient volume to make PRI cards in asterisk servers  
> rather inefficient.
>
> Initially the main carrier offered to install 16 E1s as a starting point  
> for them, but:
> a) as their call volume increases, having lots of E1 ports and cables  
> seems incredibly inefficient compared to an E3 or TDMoE.
> b) I can't help but feel asterisk might not be the most appropriate  
> solution for something of this scale


I recommend not using Asterisk at all for this - it's H323 support isn't  
workable in the VoIP wholesale market. Asterisk is a good PBX but a  
terrible softswitch.


You should go for either:

1) something like a mediant 2000 16 E1 (which will handle all the  
transcoding) + 1 nextone SBC for soft-switching.


or


2) using a company like interroute who will do the interconnect and the  
teleport between TDM and VoIP for a relatively modest price.


Once you're setup with VoIP, please get back to me directly to see which  
routes we could exchange - our switch is in London too :-)


Cheers,
Jean-Michel.

-- 
Jean-Michel Hiver - YKOZ
+262 (0)692 828 070