vicidial.org

Here is what I need a system to do, please tell me if Vicidial is right for me... and if not, any suggestions?

I would have 16 outbound analog POTS telephone lines on a couple of dialogic telephony cards.

This is an outgoing system only, and it only plays recorded messages.

I would send a text file of phone numbers, along with an mp3 file to the vicidial server. When the server sees these 2 files, it starts calling all the numbers in the text file and plays the mp3 file. It also logs who answered and who did not (based on dtmf keypress). Then, it deletes the text file and mp3 file, and waits for another one.

That's it. No inbound calling, no transfers, etc. It simply calls people, plays a message, and logs if they answered by way of dtmf keypress.

Thank you for any suggestions.

1) the functionality would require a bit of scripting, but vicidial can do it

2) the functionality is fairly basic and with some scripting ASTERISK (plain old asterisk) could handle it

3) dialogic is NOT supported by asterisk (proprietary systems), last I knew there were not even any negotiations to get a dialogic driver for asterisk (however, I honestly have not looked at that in a year and obviously "things change")

4) SIP trunks, however, could easily handle the same function or you ccould use sangoma or digium trunks or even a grandstream gateway to connect to POTS lines.

With VICIdial, you would then have the power of a full system in the future should you decide to use it. With asterisk and plain old scripting you'd have a system with very little overhead capable of a fairly large volume of calls (if you choose SIP and plain asterisk your system would be very scaled down and be very capable of maximizing potential for this single task).

Ok, I may look into that. So, asterisk doesn't support any multi-line analog cards, and I would pretty much be required to use a sip-to-analog gateway device to interface my POTS lines, correct? Thank you for your help!

digium and sangoma both make multiline analog cards. BUT a sip gateway can be used independently of your telephone server and often saves your tush when you have a problem with your server (since it can be independently reprogrammed to point to another server and you're "up" with no hardware in your actual telephone server, no "card" to move to a new box)