Chris Shaw wrote:
* should have no problem keeping up with a setup like that, and VoIP is certainly capable.... However... What most interests you? Is it cost savings or audio quality. If it's cost savings, you could push 25 calls through a T1 using GSM encoding, but it would not sound quite the same as a regular line. If you use G.711 (mu/A-Law) then you would get toll quality audio but only be able to push about 17 calls through at once...
In my experience G.729a has better quality than GSM (higher latency, though). Plus it is supported on the Cisco 7960G so I can do passthrough from 7960G to 7960G and since NuFone also supports G729a, I can do passthrough there as well. Cut down on those pesky licenses and * CPU usage.
Also you must remember that the current RTP implementation in Asterisk is... somewhat... lacking, and with a 100% VoIP setup you will need a timing source like ztdummy (which requires a UHCI USB controller) or ZapRTC. Or if you're using linux 2.6, I don't think you need anything as the internal timer resolution is precise enough...
I would probably have a Wildcard TDM400 or something for POTS backup and basic analog anyways.
Our company is thinking of deploying a setup like this but a bit smaller, only 12 extensions and at most 8-9 simultaneous calls. I would certainly recommend a setup like this, it's a huge cost savings. I would also do plenty of homework and figure out how to do it before actually committing to it. Maybe even do a parallel setup where you have some POTS lines as a backup. I would also use some kind of failover where your IAX provider can forward your incoming calls to another IAX provider's number or a POTS number during downtimes...
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