Using Asterisk and FreePBX how can I map extensions to outbound routes

With SIP, the signaling is done via SIP and the digitized audio is sent via a different protocol, RTP. The SIP and RTP can and often are sent to different IP addresses. Thats normally not a problem, as long as the IP addresses are all reachable..

Whats happening in your situation, is something like this:

PBX2 sends a SIP INVITE to PBX1. Included in that INVITE is information about where to send the audio. PBX2 specifies its own IP address. Since its IP address is reachable from PBX1, calls between the two work.

Now, when the callee is an outside line, PBX1 sends its own INVITE to your provider, and passes on in that INVITE the information about where to send the audio (ie, the IP address of PBX2) If both PBXs were on public IPs, this would be fine. Since they are not both reachable from the outside, you need to modify the behavior of PBX1

On PBX1, in your sip.conf file, there should be a peer configuration for PBX2. In that peer configuration, you need to add the following line:

canreinvite=no

(On more recent versions of asterisk, you would use directmedia=no instead.)

This will cause PBX1 to stay in the media path whenever it is involved with a call with PBX2. In otherwords, when you call the outside world, PBX1 will give your provider its own IP address for where to send the audio, it will then proxy that audio, and send it on to PBX2.

Hope this helps!

If you want the 933 and 1 added the you'd need "Outbound Routes" like this:
NXXXXXX
NXXNXXXXXX

Then, in your Trunks, you want this in the diaplan rules.
933+NXXXXXX
1+NXXNXXXXXX

That would accept NXXXXXX as a valid number from your handsets but would dial 933NXXXXXX when the call goes out on the trunk. The same goes for the 1. It will be added to all calls that are dialed with 10 digits.