The CUCM SRND states that the best way to implement a SIP trunk is to set the DMTF Preference on the Trunk in CUCM to "No Preference" and use dtmf-relay sip-kpml rtp-nte on the dial-peers pointing to the CUCM server(s). This allows CUCM to decide which method to use, depending if a phone supports out of band (e.g. KPML, SIP notify, etc.) or RFC2833 & thus avoid invoking an MTP to handle DTMF inter-working unless unavoidable.
Unfortunately this doesn't work with the old 7936 conference phones, you'll get no DTMF in calls via the SIP trunk, bug reference CSCuc80321 explains in detail. In short the workaround is to use DTMF Preference "RFC2833" on the Trunk, but this will always invoke an MTP. This in turn can cause a problem if you're using G729, as the MTPs running on the CUCM server(s) as part of the IP Voice Media Streaming App service only support G711. You'll need a router to be able to provide an MTP that can handle G729, as the example configuration snippet shows:
dspfarm profile 1 mtp
codec g729r8
codec g729ar8
maximum sessions software 60
associate application SCCP
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