Omantel use Pure in band DTMF with RFC 2833 disabled. We need to use LTI Transcoder on the CUBE to make this working. In this blog i have shared a working CUBE configurations for omantel SIP trunks

voice service voip
 ip address trusted list
  
  ipv4 212.72.15.217
  ipv4 212.72.15.220

 
 mode border-element
 allow-connections sip to sip
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 trace
 sip
!
!
voice class uri 1 sip
 host ipv4:IP
 host ipv4:IP

!
voice class uri 2 sip
 host ipv4:212.72.15.217
host ipv4:212.72.15.220






voice class codec 1
 codec preference 1 g711alaw
!
voice class codec 2
 codec preference 1 g711alaw
!
!
voice class sip-profiles 1
 request ANY sip-header Allow-Header modify ", UPDATE" ""
 request INVITE sip-header From copy "<sip:(.*)@.*>" u01 
 request INVITE sip-header From modify "From: \".*\" <" "From:\u01<" 
 request INVITE sip-header Remote-Party-ID copy "<sip:(.*)@.*>" u02 
 request INVITE sip-header Remote-Party-ID modify "Remote-Party-ID: \".*\" <" "Remote-Party-ID:\u02<" 
 request PRACK sip-header From copy "<sip:(.*)@.*>" u03 
 request PRACK sip-header From modify "From: \".*\" <" "From:\u03<" 
 request ACK sip-header From copy "<sip:(.*)@.*>" u04 
 request ACK sip-header From modify "From: \".*\" <" "From:\u04<" 
 request BYE sip-header From copy "<sip:(.*)@.*>" u05 
 request BYE sip-header From modify "From: \".*\" <" "From:\u05<" 
 response 180 sip-header Remote-Party-ID copy "<sip:(.*)@.*>" u06 
 response 180 sip-header Remote-Party-ID modify "From: \".*\" <" "From:\u06<" 
 response 180 sip-header Remote-Party-ID modify "Remote-Party-ID: \".*\" <" "From:\u06<" 
!
!
!
voice class server-group 1
description CUCM server Group
 ipv4 CUCM IP  preference 1
 ipv4 CUCM IP  preference 2
!
voice class server-group 2
description Omantel Server Group
 ipv4 212.72.15.217 preference 1
ipv4 212.72.15.220 preference 2




ip route 212.72.15.217 255.255.255.255 Omantel Interface IP
ip route 212.72.15.220 255.255.255.255 Omantel Interface IP

dspfarm profile 1 transcode  
 codec g711alaw
 codec g711ulaw
 codec pass-through
 codec g729abr8
 maximum sessions 96
 associate application CUBE
!

dial-peer voice 1 voip
 description incoming calls from PSTN
 translation-profile incoming IN
 session protocol sipv2
 incoming uri via 2
 voice-class codec 1  
 voice-class sip profiles 1
 dtmf-relay rtp-nte

dial-peer voice 2 voip
 description incoming calls from CUCM
 translation-profile incoming IN
 session protocol sipv2
 incoming uri via 1
 voice-class codec 1  
 voice-class sip profiles 1
 dtmf-relay rtp-nte




dial-peer voice 10 voip
 description Outgoing calls to CUCM
 destination-pattern ^....$
 session protocol sipv2
 session server-group 1
 voice-class codec 1  
 voice-class sip early-offer forced
 voice-class sip profiles 1
 dtmf-relay rtp-nte
 no dtmf-interworking
!


!
dial-peer voice 20 voip
 description Local calls to PSTN
 translation-profile outgoing OUT
 destination-pattern 2.......
 session protocol sipv2
 session server-group 2
 voice-class codec 1  
 voice-class sip early-offer forced
 voice-class sip profiles 1
 dtmf-relay rtp-nte
 no dtmf-interworking

dial-peer voice 21 voip
 description Mobile calls to PSTN
 translation-profile outgoing OUT
 destination-pattern [789].......
 session protocol sipv2
 session server-group 2
 voice-class codec 1  
 voice-class sip early-offer forced
 voice-class sip profiles 1
 dtmf-relay rtp-nte
 no dtmf-interworking
!
dial-peer voice 22 voip
 description International calls to PSTN
 translation-profile outgoing OUT
 destination-pattern 00T
 session protocol sipv2
 session server-group 2
 voice-class codec 1  
 voice-class sip early-offer forced
 voice-class sip profiles 1
 dtmf-relay rtp-nte
 no dtmf-interworking
!
dial-peer voice 23 voip
 description Service  calls to PSTN
 translation-profile outgoing OUT
 destination-pattern 1...
 session protocol sipv2
 session server-group 2
 voice-class codec 1  
 voice-class sip early-offer forced
 voice-class sip profiles 1
 dtmf-relay rtp-nte
 no dtmf-interworking
!
dial-peer voice 24 voip
 description ROP   calls to PSTN
 translation-profile outgoing OUT
 destination-pattern 9999
 session protocol sipv2
 session server-group 2
 voice-class codec 1  
 voice-class sip early-offer forced
 voice-class sip profiles 1
 dtmf-relay rtp-nte
 no dtmf-interworking