Asterisk 1.8 SRTP/TLS with RTP/SIP Trunk

I am trying to figure out how to use SRTP/TLS for all of our of our internal extensions and soft phones and use a Vitelity SIP trunk to make and receive calls.

I actually have SRTP/TLS working fine for my extensions and I can call out via Vitelity. I sniffed the traffic at my soft phone and it was encrypted. The problem is when a call comes in the Vitelity SIP trunk it cannot connect to my extension. IT says my extension is busy.

Soft phone is Bria 2.4

Any thoughts or suggestions are greatly appreciated.