DTMF Issue with SIP TRUNK [CUCM--SIP--CUCME--CUE]

I am facing an issue with dtmf-relay. Phone A registered to CUCM (SiteA) calls Phone D registered to CUCME (Site C). Between Regions G729 codec is negotiated. Phone D call-forward no answer to Voicemail. CUE integrated with CME. After leaving the Voicemail from Phone A to Phone D, when I press # key to send the Voicemail, it is not recognized.

Here is my scenario and the configuration.

(Phone A) -- CUCM ----SIP TRUNK ----CUCME (Phone D) ---> CUE.

voice service voip
 ip address trusted list
  ipv4 0.0.0.0 0.0.0.0
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 h323
 sip
  bind control source-interface Loopback0
  bind media source-interface Loopback0
  registrar server expires max 600 min 60
!
voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g729r8
!
telephony-service
 sdspfarm units 3
 sdspfarm transcode sessions 4
 max-ephones 5
 max-dn 10
 ip source-address 3.3.3.3 port 2000
 load 7945 term45.default.loads
 time-zone 28
 time-format 24
 date-format dd-mm-yy
 voicemail 3600
 max-conferences 8 gain -6
 call-forward pattern .T
 moh "music-on-hold.au"
 dn-webedit
 transfer-system full-consult
 transfer-pattern .T
 create cnf-files
!
sccp local Loopback0
sccp ccm 10.0.10.160 identifier 2 version 7.0
sccp ccm 3.3.3.3 identifier 1 version 7.0
sccp ip precedence 3
sccp
sccp ccm group 1
 associate ccm 1 priority 1
 associate ccm 2 priority 2
 associate profile 1 register BR2-IOS-XCODE
 associate profile 2 register BR2-IOS-CFB
 keepalive retries 5
 switchover method immediate
 switchback method immediate
 switchback interval 15
!
dspfarm profile 1 transcode 
 codec g729r8
 codec g711ulaw
 codec g711alaw
 codec g729ar8
 codec g729abr8
 maximum sessions 4
 associate application SCCP
!
dspfarm profile 2 conference 
 codec g711ulaw
 codec g711alaw
 codec g729ar8
 codec g729abr8
 codec g729r8
 codec g729br8
 maximum sessions 2
 associate application SCCP
!
dial-peer voice 1000 voip
 destination-pattern [15]...$
 session protocol sipv2
 session target ipv4:10.0.10.160
 incoming called-number .
 voice-class codec 1 
 dtmf-relay sip-notify
 no vad
!
dial-peer voice 3600 voip
 destination-pattern 3[16]00$
 session protocol sipv2
 session target ipv4:10.10.202.100
 incoming called-number 399[89]....
 dtmf-relay sip-notify
 codec g711ulaw
 no vad
!
On the CUCM, I did the following,

Media Termination Point Required (Checked)
MTP Preferred Originating CodecRequired Field: g711ulaw
DTMF Signaling MethodRequired Field: No preference
Non Secure SIP Trunk Profile: I am using TCP+UDP for INCOMING + Accept Unsolicited Notification (Checked).

Please let me know what I am missing.

Thanks,

Viki