[Asterisk-Dev] ZAP Channel Maxout.

[Kevin P. Fleming]

Christopher Dobbs wrote:
> We are planning on providing POTS termination for a local VoIP provider.
> Minimum required number of lines is 4000.  We also have to be able to 
> provide "operator" Service.

There are lots of ways to handle this, all of which are completely 
outside of Asterisk. Some options (in my personal order of preference):

Veraz Networks I-Gate 4000 PRO (up to 64,000 DS0, 100% redundancy (yes, 
100%), transcoding to your choice of codec, MGCP control)

Sentito IVG (up to 4,608 DS0, 100% redundancy (at the DS1 level, 
requires M13 muxing down from DS3), transcoding to your choice of codec, 
SIP control)

Nuera GX-21 (up to 1,632 DS0, nearly 100% redundancy, transcoding to 
your choice of codec, MGCP control)

All of these are carrier-grade products, designed for the type of system 
you are building. Digium's DS1 hardware, and Asterisk, are not (as much 
as I them <G>). Without building in redundancy, are you really willing 
to lose 12 DS1s just because a server stopped running properly (288 
channels)? If you want to make that system redundant, you have to 
purchase automatic A/B switches to route the DS1s to failover hardware, 
which is difficult to do well.

Another huge issue is that whatever telco you are working with is 
_really_, _really_ going to want this all to be a single trunk group. 
You can't split trunk groups across Asterisk servers, so at best your 
groups will be 12 DS1s each. That will be a large amount of management 
headache for your telco, and if you want to do origination with that 
configuration, it's nearly impossible to do right.

Another reply hit it on the head: if this is what you want to do, your 
equipment budget should be commensurate to the task. If you are wanting 
to build some racks full of 4U commodity servers and use TE410P cards 
for all of this, I have a feeling you are trying to save money on the 
wrong end of the equation :-)