Most of the configuration is going to depend on your CUCM config/call routing/dial plan, without knowing that, there is no way to tell you if something's wrong, you should be able to tell as you know the rest of the config.
I suggest you use DNA to test some calls as if they were coming from that SIP trunk, and see if your call routing is correct.
I see you use the <none> CSS for inbound calls, that might be a problem, but that depends on your CSS/partitions configuration, and what you're dialing.
HTH
java
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