This is definitely doable, however, it seems that your specifications need to be modified a bit. Here are some things to consider:
- What type of PSTN connectivity will your remote Asterisk server have? (SIP / POTS / T1 / PRI / etc.)
- If your remote Asterisk server is going to be using a physical medium, do you have the connectors and hardware in place? EG: If you are using a T1 line, do you have a channel bank or T1 card?
- Are you comfortable with Asterisk dialplan / AGI / AMI, or are you going to use an Asterisk distribution like trixbox, AsteriskNOW, Elastix, etc?
- Will your client location (with the POTS line you wish to ring) have a PBX, or will it just be a typical POTS line hooked up to an analog handset?
My recommendation to you:
- Get a cheap server (any 1U with a dual core processor and 512MB of RAM will do), and put it at your remote location.
- Load Asterisk 1.6+ onto your server. I recommend 1.6+ as it can use the dahdi_dummy driver as a reliable software timing source (it will ensure that your audio quality is not choppy and broken).
- Get a SIP account with a reliable SIP provider. My personal favorites are: flowroute and voipms.
- Set up your new SIP account in Asterisk, and purchase a DID (phone number). This phone number will be your business phone number, the one that you give out to clients and put on business cards.
Configure your Asterisk dialplan to receive calls from your SIP account to your IVR menu.
Your IVR menu logic should be something like:
a. Play the IVR menu.
b. Wait for a keypress.
c. If the user dialed '1', then make an outgoing SIP call to the POTS line phone number you want to reach. If the user dialed '2', then playback the recorded message.
Now, if you are looking to save money, and have the most cost-effective setup for your remote IVR, I would recommend throwing up a second Asterisk server on site at your client location (where the POTS line comes in), and throw away the pots line and just setup an IAX2 trunk between your client location and your hosted server location. This way, when calls come in to your remote Asterisk server via your SIP provider, you can route the calls (when the user hits option 1) over your IAX2 trunk, directly to the client location for free!
Depending on your skill level, and comfort with Asterisk, this could be either a really fun learning experience or a confusing nightmare. If you would like to learn more about telephony and Asterisk, especially if you are going to use it for your business, you may want to use a simple (free) Asterisk distribution like: trixbox CE, Elastix, or AsteriskNOW.