Question getting all started up

I understand that my questions are very basic. I have read as much as I can handle, but it seems as if everything that is documented is based on the belief that I have a bunch of VoIP clue. This is not the case… although I am doing my best to learn along the way.

Here is the nitty gritty: I want to make a system where I can use my existing VoIP phone (Cisco 7912) with my current VoIP carrier (RedWire Broadband of San Diego, CA) to be able to make automated calls and read user tone input. So, I want to be able to somehow make a quick application which uses a voip stack which will allow me to connect to my providers, PBX, register a phone number from a list of DIDs that were assigned to me, and then initiate a call to a specified phone number, wait for the user to answer the phone on the other end, and then play a wav file saying “Please enter your code”, the user then enters a code which was given to them, and if the code entered matches the code that is in the database, then phone will play the wave file of “Thank you for submitting the correct code”.

Now… how should I go about this? I do not need voice-to-text, I do not need unified voicemail, I just need a way to make the call, read input, and then play a message.

Do I need my own PBX? Do I need an array of other things? I’m assuming I can use some sort of softphone package or something. I have checked this page out ( … 20Software) which was very helpful to show me that what I want CAN be done, but I do not understand all this PBX stuff and I’m not sure that I need it all.

Can someone point me in the right direction? I understand that I do not know all there is to know about VoIP, which is why I am asking these questions here. I have tried doing all the reading I can on this subject but I have fallen short and am unable to decipher what I need to do to get going here.



First I am assuming that your voip provider is providing you with a SIP (sip is a voip protocol) login/password which your Cisco phone is connecting to. This is a gating item, you need to make sure this is correct. Asterisk will not work unless you can feed it the SIP info.

That said, if this is the case Asterisk will work for you and can do what you need. Here’s how it will work.

You register * to your voip provider. All calls to/from the provider go through your * box. When a call comes in Asterisk can handle your needs, asterisk can do just about anything you want to a call including ask for the code.
There are a number of ways to do this code entry thing, the simplest would be with the Authenticate() command and a file full of correct codes.

Your SIP phones (including your 7912) will also register to *. They will be extensions from Asterisk. You can then set up via extensions.conf when calls go to them, and what they can dial (each other/out/etc).

You only need a softphone if you want to make / recieve calls from a computer using a headset.

Hope that helps, feel free to ask more if/when you have more questions :smiley:

also check, great site :smile: