i have a elastix/freepbx box which has been upgraded to asterisk 1.6 for handling TCP based SIP Trunking. 

there are a bunch of extensions on this box, format 5XXX.
there is a SIP Trunk to/from callplus for PSTN connectivity
there are outbound routes that use the callplus sip trunk
there are inbound routes that accept inbound calls from the PSTN via the callplus sip trunk and send to ring groups or extensions

all the above works fine and as expected.

There is also a SIP trunk (using tcp as transport) to/from a microsoft OCS mediation server

Calls from OCS to number that matches an inbound route works fine and rings the correct extension or ring group.

call from OCS to number that is on the PSTN fails with the asterisk message "the number you have dialled is not in service".

how do i configure so that calls coming in to asterisk on the OCS SIP trunk can be routed out via the callplus SIP trunk?  I could set up an inbound route to catch all but I dont have an option (in the gui) to send calls to a trunk, only extension, ring groups, ivr, etc.

thanks
Regs