Voice Gateways and Session Border Controllers

A CUBE running IOS XE receives an inbound INVITE from an ITSP. Two inbound dial-peers are configured: dial-peer 100 has 'incoming uri via 10.1.1.1' matching the provider's source IP, and dial-peer 200 has 'incoming called-number 5551234...'. The INVITE arrives from the ITSP source IP 10.1.1.1 with a called number of 5551234567. Both dial-peers appear to be valid candidates. Which dial-peer will CUBE select to process this inbound call leg, and why?

An engineer configures a CUBE to accept inbound SIP calls from an ITSP. The current inbound dial peer matches on 'incoming called-number', but the ITSP recently changed the format of the called number for each customer trunk, breaking inbound matching. The engineer wants a deterministic, format-independent way to bind incoming INVITEs from this ITSP to a specific inbound dial peer based on the host portion of the SIP Request-URI. Which configuration achieves this?

You are deploying media resources on an IOS XE CUBE that must transcode between G.711 and Opus for calls between a legacy PBX and a cloud calling provider. The gateway is a Cisco 4451-X ISR with a PVDM4 module installed. During design review, a colleague proposes using Local Transcoding Interface (LTI) rather than the traditional SCCP-controlled DSP farm profile. Which statement correctly describes a key characteristic of the LTI approach for provisioning these transcoding resources on the CUBE?

A CUBE connects an ITSP SIP trunk to CUCM. The provider sends RFC 2833 DTMF, but CUCM's SIP trunk is configured for out-of-band (SIP KPML/unsolicited NOTIFY), and calls otherwise complete with two-way audio. Digits pressed by PSTN callers into the CUCM auto-attendant are not recognized. The engineer wants CUBE to convert the DTMF signaling between the two legs while allowing media to bypass CUBE where possible. Which configuration approach correctly addresses this requirement?

A CUBE is sending calls to an ITSP that requires the called number in the format 91XXXXXXXXXX, but CUCM forwards the called number to CUBE as a 10-digit national number (XXXXXXXXXX). The engineer must prepend '91' to the called number only on the outbound dial peer facing the ITSP, without affecting inbound matching or other dial peers. Which configuration on the outbound VoIP dial peer accomplishes this?

A CUBE has two outbound dial peers to reach the PSTN through separate ITSPs. Dial peer 100 has 'preference 1' and points to ITSP-A; dial peer 200 has 'preference 2' and points to ITSP-B. When ITSP-A returns a SIP 503 Service Unavailable, calls fail immediately instead of rolling over to ITSP-B. The engineer confirms both dial peers match the destination pattern equally. What is the most likely cause and correct fix?

A CUBE has two outbound VoIP dial peers matching calls to the PSTN provider. Dial peer 100 is configured with 'preference 1' and points to the primary ITSP session target, while dial peer 200 has 'preference 0' pointing to a backup ITSP. Both dial peers use the identical 'destination-pattern 9T' and both session targets are reachable. When a user dials a PSTN number, which dial peer does CUBE select first, and why?

An engineer is configuring a CUBE to send outbound calls to an ITSP that provides two SIP proxy IP addresses (198.51.100.10 and 198.51.100.11) for redundancy. The requirement is that a single outbound dial peer must be able to try both proxies, distributing new call attempts across them, and failing over to the second proxy if the first is unreachable. Which configuration approach meets this requirement?

A CUBE connects CUCM to an ITSP. Calls forwarded by internal users are being rejected by the ITSP with a 400 Bad Request. A SIP debug shows that outbound INVITEs contain a Diversion header populated with an internal E.164 number that the ITSP does not recognize as a valid subscriber. The provider has confirmed they simply want the Diversion header removed entirely from all outbound INVITEs on the ITSP-facing trunk. Which configuration correctly accomplishes this on CUBE?

A CUBE connects an internal CUCM cluster to an ITSP SIP trunk. Voice calls work perfectly, but every T.38 fax call fails during the switchover to fax. A debug shows that when CUBE sends the re-INVITE to the ITSP, the SDP media line reads 'm=image 4000 udptl t38', while the ITSP's own documentation states its equipment only accepts the media transport token 'UDPTL' followed by 'TCP/UDPTL' capability and rejects lowercase 'udptl'. You must modify the outbound signaling toward the ITSP without affecting the internal leg. Which configuration approach resolves this?

A CUBE terminates SIP trunks for two different ITSPs. ITSP-A requires the outbound INVITE to use the domain 'itsp-a.com' in the SIP From and Contact headers, while ITSP-B requires 'itsp-b.net'. Both providers reach the internet through the same egress dial-peer configuration style, but they are separated using 'voice class tenant 100' and 'voice class tenant 200' referenced under their respective outbound dial peers. You have written two separate SIP profiles that rewrite the domain. Where must these SIP profiles be applied so that each ITSP receives only its own domain rewrite without affecting the other trunk?

A CUBE is configured to register to an ITSP on behalf of an on-premises PSTN pilot number using a SIP profile with outbound registration. After configuration, the ITSP reports that no REGISTER requests are being received, and 'show sip-ua register status' on the CUBE shows no entries. Debugging reveals the CUBE is generating no REGISTER traffic at all. Which configuration element is most likely missing?

A CUBE running IOS XE connects a G.729-only ITSP SIP trunk to an internal CUCM cluster where all IP phones are configured to use only G.711ulaw. Calls between the phones and the PSTN fail to establish media, and debugs show a codec negotiation failure. The gateway has spare DSP resources available. Which media resource must you configure on the CUBE to allow these calls to complete?

An engineer configures a CUBE to send calls to an ITSP that rejects any called number containing a leading plus sign. Inbound calls from CUCM arrive with called numbers in +E.164 format (e.g., +14085551234). The engineer creates the following: 'voice translation-rule 10 / rule 1 /^\+/ //' and 'voice translation-profile TO-ITSP / translate called 10'. Where must this translation profile be applied so that the plus is stripped before the digits are sent to the ITSP?