Assuming that you have Stage Method set to 1, and the Dial Plan permits and does not alter the number dialed, then whatever number is sent from the trunk should be dialed on the FXO port.
As a start, at the Asterisk command prompt, type
pjsip set logger on
make a failing outbound test call, paste the Asterisk log for the call at pastebin.freepbx.org and post the link here. It would also be useful to post screenshots of the FXO Port page.