Can anyone help with the following question?

I have a FreePBX instance setup with two softphone extensions, A and B, registered with it. I have a SIP trunk with an outbound route with a dial prefix of ‘.’ which is intended to match all calls. At the other end is another B2BUA system which will handle further processing.

If A dials a true external number, the call is routed to the SIP trunk. If A calls B, FreePBX does the smart thing and connects them directly.

I want all call signaling to be routed over the SIP trunk to the remote system, even when A calls B, so that the remote system can monitor and possibly modify the call before tromboning it back along the same trunk to FreePBX to terminate it to B. Can this be done in FreePBX, and if so how? Or should I be dropping down a layer and configuring Asterisk directly?