Reliable solution for 600 analog extensions

I’m looking for a reliable solution for 600 analog extensions Trixbox configuration (single PRI connection to PSTN). I’d like to get a feedback from someone that has some experience (or ideas?) for this size / type of installation.

Thanks,

Can I just ask why you need 600 analogue extensions? It would be better to use IP phones (and cheaper).

Hi

How many concurrent calls ? as since trix uses AGIs to do every thing the box will grind to halt as every call will start an asterisk process.

Ian

to add to this a bit- with asterisk you always end up paying per channel (per extension) $50 to around $300, that is cheap ATAs to expensive IP phones. For a lot of analog phones you should consider all the options:

Voip gateway devices, like audiocodes etc. COnnect to * via SIP, gives you 24 ports or so
Channel bank- basically a T1 breakout box. Put a T1 card in the server and connect to channel bank, get 24 analog ports
ATAs- give each user an ATA. Harder to administer

High density analog cards like TDM2400 arent really an option…

And finally IP phones. I think this will be your best bet, for 600 units you can probably order direct from the manufacturer and get a much better deal. Try SNOM and AAstra… both have good remote provisioning tools and good documentation. For low end SNOMs or aastras you are paying around $100/phone, and when ordering in bulk you could probably get something like aastra 9133’s or snom 320’s for not much more than that. In return your users get a much nicer phone with useful buttons like hold/transfer/conference/etc.
The only problem with IP phones is that they require ethernet infrastructure. If you dont have 600 ethernet ports (remember an IP phone can passthru so you put it between the jack and the computer) then that increases the cost…

And lastly as was mentioned Trix uses agi’s for everything, which means if any significant portion of these users are on the phone at the same time or dialing at the same time your box will slow to a crawl. You should read the book asterisk: the future of telephony and write configs from scratch or use realtime… will work much faster.