Hi Monty, thanks for the reply. We are considering both, but have little knowledge of sip/analog gateways in production. There is a potential large cost saving for us to go the 3rd party sip/analog route.
I've done SIP/T1 to integrate another type of system that needed PSTN access through CM. It works well. Look at "implicit user rules" in System/Session Manager. If a gateway sends a SIP message from analog phone 1234 to SM, the implicit user rule triggers. Basically you match a range of numbers to a rule to just go to CM and use station programming there. That way you can get SIP messaging into CM to make your analog phone go off hook, use COR/COS/FRL/whatever other CM features and administration you need.
I'd think the largest concerns would be 1. some system to manage those gateways and ports - I'm sure Audiocodes/brandX has some product/suite to put all your Audiocodes/brandX gateways and ports into 1 web page and 2. determining what features you need, how to reconcile them, and doublechecking t.38, modem, whatever other voodoo they're doing with the analog and ensuring it syncs up with what your CM is doing, and potentially a SIP trunking provider is doing.
All to say, looks good on paper, but I don't typically like troubles involving 3-4 support vendors on the phone debugging a 14.4 kbps fax or modem connection
I do agree with Kyle ..maybe i should of elaborated my concern would be that you would have a unsupported solution by any one vendor that would take a lot of time to plan , configure, test, troubleshoot , when all that is taken into consideration the saving on capital outlay could soon become irrelevant.
It is the amount of devices that concerns , but that said you need to make an informed decision and i would get any prospective vendor to prove the solution works to you in a clear and concise way.
We do a ton of this using Audiocodes gateways. We to analog and DS1 to SIP with them. They are relatively inexpensive compared to Avaya and work just as well.
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