I’m interested in finding a phone system that can:
Allow a client to put in their vendor #
Enter a specific offer code and have the system announce if that particular code has been redeemed
If it is still valid, it will allow them to redeem it or mark it as ‘redeemed’
The data would be in a MySQL database that would have the “redeemed” or “Not redeemed” status’.
Also, I know next to nothing about PBX’s and phone systems. If I have 1 businees line with this system in place and one person is going through the process, what happens when other vendors call in? Do I need the 4 or 8 line PCI-E cards to route multiple calls from a single business line into a multiple call system?
I can atleast answer part of this… Which is yes… The tools and applications built in to asterisk give you the functionality to be able to accomplish what your trying to do. But if you only have 1 business line (or 1 channel) connected to the system then it can only handle 1 caller at a time. To be able to support multiple callers you will need multiple lines or trunk tied into the system. Whether that be multiple SIP trunks from a VOIP SIP provider, T1 etc…
There are several ways to accomplish that either by using some creative use of CURL against a web server to get the responses you need or using phpagi to write a full php application that would send and receive commands from the asterisk server based on responses from the caller.
Just when i think i’ve found the limits of asterisk I am suprised to find new things I can do that i didn’t know about before… It is quite flexible and powerful if you know what your doing.
Unfortunately i do not know much about the hardware end, yet, so I can’t speak much on that part of your question.
If you have one analogue business line, you are restricted to one caller at a time. If you have one primary rate digital line, you can have 23 or 24 (USA) or 30 callers, on a single PCI card.
You will have to ask your PSTN provider if they offer a queueing service. I suspect it is only possible as part of a Centrex offering.
You should also consider having a VoIP incoming trunk, rather than a circuit switched one, and letting someone else handle the PSTN interface.
Primary Rate digital line for 23 (PRI) / 24 (RBS) T1 or 30 for E1. Basic Rate gets you 1 per bearer - usually 2 per line provided.
Teminology slip, now corrected.