Apart from what you already shared can you please take screenshots of the entire SIP trunk configuration for both the Cube and the CUC, if you use SIP for that integration, from CM? I have a suspicion that you might be using a specific DTMF relay on either of these instead of the recommended No Preference.
I know that you already have solved your issues, but I took the time anyway to go through your configuration and clean-up the parts that are related to the voice service. There are more things that I'd likely clean-up if it was my configuration to maintain, but as I don't know any details on your system landscape I left that out.
This is what I would suggest you to do to get a cleaner looking configuration.
voice service voip
ip address trusted list
no ipv4 10.8.32.30 255.255.255.255 ! not needed as this is your CUC server, only CM needs to be added to list of trusted
no allow-connections h323 to h323 ! not needed for SIP to SIP calls
no allow-connections h323 to sip ! not needed for SIP to SIP calls
no allow-connections sip to h323 ! not needed for SIP to SIP calls
mode border-element licence capacity XX ! the number of sessions that you have licensed for Cube, if you use never HW and SW combo the command is just mode border-element
h323 ! As you do not use H323 this is redundant
no h450 h450-2 timeout T1 1000
no h450 h450-3 timeout T1 1000
no h225 timeout ntf 500
no h225 display-ie ccm-compatible
no h225 connect-passthru
no h245 passthru all
no h323
no voice class h323 1 ! As you do not use H323 this is redundant
no voice class sip-profiles 1 ! As you do not use this profile this is redundant
! As you do not use any of these it is redundant
voice register pool 11
no translation-profile outgoing PSTN_CallForwarding
no voice translation-rule 1 ! not user anywhere, so redundant
!
no voice translation-rule 2 ! not user anywhere, so redundant
!
!
no voice translation-profile PSTN_CallForwarding ! the referenced translation rules in this profile does not exist, so it is redundant
!
no voice translation-profile PSTN_Outgoing ! not user anywhere, so redundant
!
no voice translation-profile To_CUCM ! not user anywhere, so redundant
!
no voice translation-profile add-prefix-9-to-CID ! not user anywhere, so redundant
interface GigabitEthernet0/1.36
no h323-gateway voip bind srcaddr 10.8.36.2 ! As you do not use H323 this is redundant
no dspfarm profile 3 conference ! As you do not use this profile this is redundant
dspfarm profile 1 conference ! this is to remove any codec with a "b" in the name as you do not use VAD
shut
y
no max sess
no codec g729br8
no codec g729abr8
maximum sessions 4
no shut
!
voice translation-rule 40 ! added to remove prefix for outbound calls
rule 1 /^9\(.*\)/ /\1/
!
voice translation-profile PSTN-OUT ! added to remove prefix for outbound calls
translate called 40
!
dial-peer voice 100 voip
destination-pattern 9T !to stop the loop, make sure that you send the called number for outbound calls with the prefix (9) that I think that you might use for external calls
no session target sip-server ! not needed on this dial peer
translation-profile outgoing PSTN-OUT ! added to remove prefix for outbound calls
!
dial-peer voice 110 voip
no session target sip-server ! not needed on this dial peer
!
dial-peer voice 111 voip
no session target sip-server ! not needed on this dial peer
!
dial-peer voice 200 voip
no session target sip-server ! not needed on this dial peer
!
! As you do not use MGCP this is redundant/unneeded
no ccm-manager fallback-mgcp
no ccm-manager mgcp
no ccm-manager config server 10.8.32.20
no ccm-manager config
no mgcp call-agent 10.8.32.20 service-type mgcp version 0.1
no mgcp
sccp ccm group 1
associate profile 33 register <name of the MTP that you have created in CM under Media Resoures MTP> ! this is for the MTP resource that you have in the gateway to register to CM
If you have any question or wonder about parts please reach out to me.
By the look of your debug you use dial peer 92 as the inbound and 10 as the outbound. Based on your shared configuration that is not likely what you want or expect. I would advice you to follow @Vladimir Adutskevich advice and rework your dial peers so that you have these.
- Inbound from your ITSP
- Outbound to your ITSP
- Inbound from your CM(s)
- Outbound from your CM(s)
The recommend best practice these days are to use information in the VIA header to match the inbound dial peer and then for the outbound you should make sure that you do not have any overlap in the match. With your current configuration you have that on dial peer 10 as it matches .T. This is a very bad way of setting up your dial peer as it would match anything and that creates loops in your call plan.
I would recommend you to read up on the topic and this outstanding document is a good place to start In Depth Explanation of Cisco IOS and IOS-XE Call Routing
