I'm having a very strange problem.  I have CUCM 7 and Asterisk setup in a lab environment.  I am trying to use asterisk as a SIP server to Bandwidth.com SIP trunk.  I created an extension using Xlite softphone in asterisk and am able to send and receive calls over the bandwidth.com SIP trunk to the PSTN.  I created another SIP trunk in Asterisk to CUCM.  I can call from the PSTN and the call goes through the asterisk box to a DN on CUCM just fine.  I can also dial a CUCM IP Phone from the Xlite softphone registered in asterisk.  However, even though I setup a specific route-pattern to call the asterisk extension I am unable to do so from a CUCM IP Phone.  I am also unable to route calls to the PSTN from an IP phone registered in CUCM.  I think the problem is for some reason CUCM is not actually sending the SIP traffic OUTBOUND.  When I try to call, I get the message "Your call can not be completed as dialed, please consult your directory and try again." and then a fast busy.  I ran the DNA and it shows everything as being fine, no issues there.  When I ran wireshark, I saw NO SIP traffic originating from CUCM.  I also have the asterisk console open in another window and that correlates with no traffic appearing to come outbound from CUCM.  I have the SIP trunk profile setup for UDP, my CSS setup is also correct.  Any ideas why I can get traffic going from the asterisk server to CUCM, but not the other way around?  Also, they are both on the same subnet, no NAT issues involved either.  This is very strange.  I noticed a similar issue on NetPro in regards to a ShoreTel SIP Trunk.  Thanks for any suggestions.