The T-1 or DSX-1 port 0/2 is only used when connecting to a T-1 voice circuit. This would typically be a PBX using ISDN PRI signaling or a T-1 channel bank. It would not be needed for connecting to a SIP device (this uses Ethernet) or analog phones on the VOICE connector (these use the FXS ports).
If you want to test the physical port you can create a T-1 loopback plug by jumpering pins 1 to 4 and 2 to 5 on an 8-pin modular plug. This should cause the port to show an "UP" status. However it is not needed in your present setup as nothing in your configuration is capable of the T-1/DSX-1 protocol.
In your configuration, there are a few things you'll want to fix. Your default route isn't in your connected subnet. it should be in the 10.10.10.x range, where your router to the Internet is connected. Also your voice users only have a single digit as both the user identifier and SIP identity. This is their extension and is normally three or four digits. Unless you define that single digit in your dialplan it isn't going to work as expected. You also have a conflict between analog station 2 and SIP station 25. I'd use three digit extensions as you get started.
Thank you for your reply!
So i would change the config to
interface t1 0/1
And plug in the loop back?
And this wouldn't be a final config set up this was just for me to make sure the units work as they should before they get set up for real world use. So using the single digit extension was just a way to speed up the first set up on the FreePBX.
Since you have "a few" available, you could set up PRI trunks between them. You'll need a T1 crossover cable to link up like-models. The 900e series has DS1 ports instead of DSX, so those can hook up to a non-e model with a straight through cable.
Once you have the PRI interface configured on each side you send set up the dial plan on each to reach the extensions of the other. You'll definitely want to move to 3 or 4 digit extensions so each unit can have a unique range. Make sure one is using internal timing and the other is using the first as its timing source. You could get crazy and chain them all together over a sequence of PRI and SIP trunk hops!
Once it's all set up, turn up the debug and test various call scenarios to check for anything out of the ordinary.