I have a sip trunk that is connected to the telephone service provider, and when someone calls to another user of the same cucm, but dialing the long number (the number providade by the telephone service provider + 0), the call is directed to the sip trunk, goes to the service provider, and comes back again. This calls last 2 minuts and 2 seconds and are dropped.
I have made some traces, and I see that if a call is made to an number that is not on the cucm, it goes normaly, but in the situation reported above, i see a lot o subscibes and notifys packets to the sip trunk, that eventualy expire and aparently drop the call.
The first question is why are you not implementing forced on net dialing? By this I mean all your DID assigned to your internal users should bot be going out of the SIP trunk to the PSTN. These calls should be routed internally. You can easily achieve this by using translation patterns. If you still want to route this calls out and then back in, then provide call traces.