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Outbound calls to Sip trunk through CUCM 8.2, Getting Dead Air

DTMF not working after the change to fast start.
Call flow: Phones---CUCM--H323--CUBE--SIPtrunk--ISP

When calling 8005201221 you are unable to hear ringback and the call would never connect, hence the report of dead air.
You would see the SIP Invites going out and you are getting the 100 trying and the 183 session progress. But you never got a 200 ok.
You enabled outbound faststart on the CUCM H323 GW config page and the call went through
You heard audio but your DTMF was not being recognized and you were still not seeing a 200 ok coming in from the ISTP.  You saw that the inbound dial-peer that was being matched for the leg between
the CUCM and the CUBE was a SIP dial-peer with dtmf-relay rtp-nte.
This dial-peer was being matched because of the generic incoming called-number . 

The above mentioned issue could happen because it was matching the wrong dial-peer.
The dial-peer it was matching has wrong DTMF type setup. 

1809 : 5119 17:13:43.208 EDT Tue Oct 30 2012.1 +3630 pid:30369 Answer 5742930581 active
 dur 00:00:03 tx:173/27680 rx:147/23520
IP SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw
TextRelay: off
 media inactive detected:n media contrl rcvd:n/a timestamp:n/a
 long duration call detected:n long duration call duration:n/a timestamp:n/a
The following dial-peer was setup to resolve the issue.
This dial-peer was setup for RTP-NTE.
dial-peer voice 30369 voip
no incoming called-number 1[2-9]..[2-9]......
 dial-peer voice 30371 voip
 description Long Distance DIALING FROM CUCM
 incoming called-number 1[2-9]..[2-9]......
 voice-class codec 1
 voice-class h323 1
 dtmf-relay h245-alphanumeric
 ip qos dscp cs3 signaling
 no vad
All inbound calls to CUCM endpoints had specific dialed strings.
It was not just an prefix with DNs in the last 4 positions.
This allowed to be a bit more broad with the dial-peers used for a call from CUCM to the ITSP.

dial-peer voice 1 voip
description LD OUT TO ITSP

destination-pattern 1[2-9]..[2-9]......
session protocol sipv2
dtmf-relay rtp-nte

dial-peer voice 2 voip
description LOCAL OUT TO ITSP
destination-pattern [2-9]......
session protocol sipv2
dtmf-relay rtp-nte
After this the outbound calls would set up normally and DTMF was recognized. 
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