PAP2T can't parse DIDs?

Question

I just switched to a SIP provider with whom I have two numbers, and bought a PAP2T to accommodate. One number is my home office biz number, the other is number is my private/personal number. These numbers are sent as DIDs when someone calls.
I have my biz phone in the "Phone 1" port and my personal phone plugged in the "Phone 2" port.
Upon portover to my SIP provider, I discovered the device is not sending DIDs for phone 2 to phone 2. Instead, calls for phone 2 are ringing on phone 1. 
If I call out from phone 1 to my cell, the CID is correct. If I call out from phone 2, the CID is also from phone 1. In other words, the device seems to think both phones are phone 1.
I called the SIP provider and we ran some test calls, They insisted they're doing what they're supposed to be doing, and its my device that seems to be mentally challenged. 
I have the Cisco ATA manual, and it does not address the capability of configuring the lines. It seems like its supposed to be automatic. There does not seem to be any settings that can be changed that would address the phenomenon.  
I spent 45 minutes on-line with a CISCO tech assist guy. He went through my settings remotely and could not find anything wrong.  He finally gave up and said he'd try to figure it out with the lab PAP2T.
Any ideas...anyone?
Tnx....Rich

Answer 1

The PAP2T does not have logic to "parse" an incoming call beyond the ip_address : port number.
Yes, I was referring to 5060 and 5061 the default sip port numbers.  An incoming call to a PAP2T is directed to an ip address and a specific port number.  If the call is to the userid and sip port number setup on Line 1, the phone on Line 1 will ring.  If the call is to the userid and sip port number setup on Line 2, the phone on Line 2 will ring.  There is an additional optional check that you have previously registered with the voip provider.  You can setup most any port numbers you want for line 1 and line 2.  Your registration requests will include the specified port number.  If you have the same account number and password setup on each line, either Broadvox will maintain multiple registrations or the effective registration will be the last port number to send a register request.  If they maintain multiple registrations they would more than likely ring all registrations for an incoming call.
Incoming calls use sip signalling.  The incoming call is called a sip invite.  An incoming sip invite is directed to userid @ ip_address: port.  The userid matches whatever you have setup as the userid on the line tab.  The ip address is your ip address.  The port number is the sip port number setup on the Line tab.  The To: line of the sip invite may reference something you have in the AuthID field if you are using that field. 
Maybe an asterisk PBX can separate incoming calls with the same userid beyond the uderid ip address : port number.  I don't know the answer to that, but a PAP2T or a SPA3012 use userid@ip_address : port.