I happen to be a customer of TW Telecom and ended up using one of there SIP to Anaolg services that TW calles "ISDN Phone converged service"
This service is done over a PRI line, and in asterisk uses the DAHDI interface. Basically we have a ISDN line attached with 23 channels to talk analog over. This then goes into a ADTRAN box that converts the calls into sip.
This is what TW states is GR-1310 (Call Deflection for PRI),GR-853 (ISDN Call Forwarding).
I understand that there is in asterisk 1.8 the DAHDISendCallreroutingFacility that supports this. We are running 1.8.10 at this time.
My company has stated that after some searching by myself and another and coming up with no easy solution to support this, in FreePBX, that I would be allowed to put up a bounty and pay to have this support added and returned to FreePBX.
I am not sure how to go about doing this so if anyone could give me pointers on the process I would appreciate that.
If paying for support to have this added would be better please let me know.