You’ll need a lot more information before anyone can help you. Here’s why:
In order for FreePBX/Asterisk to receive a call, a trunk has to be established to send the call to the PBX. That trunk is (for purposes of this discussion) the only way to get a call into the system. In order to do that, you have to know where the trunk destination is so that you can program it. Note that the trunk identifies where the calls are coming from and what number is being called, so it’s going to have to meet that criterion at least.
After that, all calls that come in will be routed through all of your Incoming Routes until one matches. Once a match is found, the call will be processed through that Incoming Route to whatever the ‘next’ destination is. There is a default Incoming Route that you can implement that we call an “any/any” route. To configure this, simply leave the DID (your number) and the CID (the caller’s number) blank.
So having the phone number is good. Having a SIP account is good, but they aren’t sufficient to do what it sounds like you want to do.