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Question 241. A Cisco Unified Communications Manager administrator wants a user to dial a service number using a short code, while the actual destination requires a completely different digit string. Which feature is most appropriate?

1. Route group
2. Translation pattern
3. Region configuration
4. Device pool

Correct Answer: 2. Translation pattern

Explanation :-

A translation pattern is designed to manipulate digits in Cisco Unified Communications Manager. It can transform a short code or other dialed number into a different destination format before the call continues through normal route selection. Route groups determine which gateways or trunks are used, regions influence codec relationships, and device pools provide common device configuration. Translation patterns are therefore useful when the dialed digits must be transformed without necessarily exposing the transformed destination directly to the user. The resulting number can then be routed according to the applicable dial-plan configuration and calling search space.

Question 242. A SIP trunk must use a particular certificate authority for TLS authentication. After a certificate replacement, the TLS handshake fails. Which configuration should be examined first?

1. Route list order
2. DTMF relay
3. Location bandwidth
4. Certificate trust store and certificate chain

Correct Answer: 4. Certificate trust store and certificate chain

Explanation :-

TLS communication depends on successful certificate validation and trust. When a SIP trunk certificate is replaced, the receiving system must trust the new certificate or its issuing certificate authority. The administrator should therefore examine the trust store, certificate chain, validity period, subject or SAN requirements, and whether the appropriate CA certificate is installed. Route lists affect call routing, DTMF relay affects digit transport, and location bandwidth affects CAC. A failed TLS handshake immediately following certificate replacement strongly points toward a certificate trust or validation problem.

Question 243. A CUBE receives an inbound INVITE and must select the correct inbound dial peer based on the SIP URI domain used by the provider. Which capability should be considered?

1. SIP URI-based dial-peer matching
2. Location CAC
3. Device Mobility
4. Region configuration

Correct Answer: 1. SIP URI-based dial-peer matching

Explanation :-

CUBE can use SIP URI information when matching dial peers, allowing routing policies to distinguish calls based on SIP domains or other URI components. This is particularly useful when an enterprise communicates with multiple providers or organizations using different SIP domains. Location CAC controls bandwidth admission, Device Mobility changes device-specific settings according to location, and region configuration controls codec relationships. When the inbound SIP request contains domain information that should determine which dial peer handles the call, URI-based dial-peer matching is the relevant capability to investigate.

Question 244. Two Cisco Unified Communications Manager sites use different bandwidth policies. Calls between them should use a specific codec, while calls within each site can use another codec. Which configuration establishes this relationship?

1. Route group
2. Translation pattern
3. Region configuration
4. Calling Search Space

Correct Answer: 3. Region configuration

Explanation :-

Regions in Cisco Unified Communications Manager define codec relationships between devices associated with different regions. Administrators can specify how calls within a region and between regions should be encoded, making regions useful for managing bandwidth across WAN connections. Route groups determine gateway or trunk selection, translation patterns manipulate digits, and Calling Search Spaces determine partition access. Therefore, when the requirement is to use a particular codec between two sites while allowing another codec for local calls, the administrator should configure the appropriate region relationships.

Question 245. A branch loses its WAN connection to centralized call control. Phones enter SRST mode, but users cannot dial external PSTN numbers. Which configuration is most important to verify?

1. Local SRST dial peers and PSTN gateway routing
2. Centralized Device Mobility settings
3. Unified Communications Manager region relationships
4. Centralized route-list distribution

Correct Answer: 1. Local SRST dial peers and PSTN gateway routing

Explanation :-

During SRST operation, the branch router provides local survivable call processing rather than depending on the unavailable centralized call-control system. External PSTN calls therefore require appropriate local dial peers and a reachable PSTN gateway or trunk. The administrator should verify destination patterns, preferences, voice-port or trunk connectivity, and the routing path used by the local SRST configuration. Centralized route lists and region relationships do not provide the local survivable PSTN routing function. Device Mobility also does not replace the local call-routing configuration required during an SRST outage.

Question 246. A SIP endpoint sends a reliable 183 Session Progress response and expects the remote endpoint to acknowledge it. Which SIP method provides that acknowledgment?

1. UPDATE
2. REFER
3. CANCEL
4. PRACK

Correct Answer: 4. PRACK

Explanation :-

PRACK, or Provisional Response Acknowledgment, is used to acknowledge reliable provisional SIP responses. This mechanism is particularly relevant when provisional responses such as 183 Session Progress are sent reliably. UPDATE has a different purpose and can modify session information within an existing dialog, REFER is commonly associated with referral and transfer operations, and CANCEL terminates a pending request. Therefore, when a SIP endpoint needs to explicitly acknowledge a reliable provisional response, PRACK is the appropriate SIP method.

Question 247. A CUBE deployment has enterprise and provider interfaces, but outbound SDP advertises an address that is unreachable from the provider network. Which area should be investigated?

1. Calling Search Space
2. Translation pattern
3. Media interface/address binding and NAT
4. Route pattern partition

Correct Answer: 3. Media interface/address binding and NAT

Explanation :-

SDP communicates the IP address and port where media should be sent. In a CUBE deployment with multiple interfaces or NAT boundaries, incorrect media bindings can cause CUBE to advertise an enterprise-side or private address that the provider cannot reach. The administrator should inspect CUBE media interface or address binding configuration and the associated NAT design. Calling Search Spaces and route patterns control call routing, while translation patterns manipulate digits. Correct media addressing is essential for establishing a usable RTP path across the enterprise-provider boundary.

Question 248. A company uses two SIP providers. Calls to one SIP domain must use Provider A, while calls to another domain must use Provider B. Which CUBE routing method is appropriate?

1. Region-based routing
2. SIP URI-based dial-peer matching
3. Location CAC
4. Device Mobility

Correct Answer: 2. SIP URI-based dial-peer matching

Explanation :-

SIP URI-based dial-peer matching allows CUBE to use information such as the destination SIP domain when selecting a dial peer. This makes it possible to associate different SIP domains with different providers or routing policies. Region configuration is primarily concerned with codec relationships, Location CAC controls bandwidth admission, and Device Mobility applies location-specific device settings. When the provider selection depends on the destination domain contained in a SIP URI, URI-based dial-peer matching provides the appropriate routing mechanism.

Question 249. An external provider requires CUBE to remove a proprietary SIP header before forwarding an INVITE. Which feature is designed for this type of signaling manipulation?

1. SIP profile or normalization
2. Route group
3. Location CAC
4. Device Mobility

Correct Answer: 4. SIP profile or normalization

Explanation :-

SIP profiles and normalization capabilities are intended to modify SIP signaling so that messages can interoperate with systems having different signaling requirements. Depending on the platform and configuration, they can manipulate supported SIP headers or other message elements. Route groups determine gateway or trunk selection, Location CAC controls bandwidth admission, and Device Mobility applies location-specific device settings. If a provider specifically requires an unwanted header to be removed from an outbound INVITE, the administrator should investigate the appropriate SIP profile or normalization configuration on the relevant signaling path.

Question 250. A SIP call consistently disconnects after the same negotiated interval. Packet capture shows that session refresh requests are not completing successfully. Which feature should be investigated?

1. Translation pattern
2. SIP session timers
3. Route group
4. Device Mobility

Correct Answer: 1. SIP session timers

Explanation :-

SIP session timers provide a mechanism for periodically refreshing an established session. If the negotiated refresh mechanism fails and the session reaches its expiration interval, the call can be terminated. A repeatable disconnect at approximately the same interval, combined with failed refresh transactions, is therefore a strong indicator of a session-timer issue. Translation patterns manipulate digits, route groups determine routing paths, and Device Mobility applies location-specific device settings. The administrator should inspect Session-Expires, Min-SE, refresher negotiation, and the corresponding refresh requests and responses.

Question 251. A user can reach internal destinations but cannot reach a PSTN route pattern because the route pattern is in a partition excluded from the user’s effective CSS. What should be changed?

1. The user’s effective Calling Search Space
2. The phone’s region
3. The SIP session timer
4. The route group’s codec

Correct Answer: 3. The user’s effective Calling Search Space

Explanation :-

A Calling Search Space determines which partitions a device or line can search when Cisco Unified Communications Manager performs digit analysis. If the desired PSTN route pattern is in a partition that is absent from the effective CSS, the route pattern is not available to that user even though it exists. The administrator should therefore adjust the effective CSS to include the appropriate partition while preserving the organization’s intended dial-plan restrictions. Regions, session timers, and codec configuration do not determine whether a route-pattern partition is visible to a user.

Question 252. A CUBE administrator wants to prevent unsolicited SIP requests from unknown external sources from reaching the enterprise call-routing environment. Which configuration provides the most direct protection?

1. Region codec configuration
2. Source-based access control and trusted-peer restrictions
3. Translation pattern
4. Route group distribution

Correct Answer: 2. Source-based access control and trusted-peer restrictions

Explanation :-

CUBE can restrict SIP signaling according to trusted sources and appropriate access-control policies. Source-based controls help prevent unsolicited or unauthorized devices from sending SIP requests into the enterprise call-routing environment. This is an important component of protecting externally exposed voice gateways against unauthorized calling and toll-fraud attempts. Region codecs, translation patterns, and route-group distribution address different aspects of collaboration configuration and do not directly establish source authorization. Therefore, trusted-peer and source-based access controls should be reviewed when the objective is to limit which external SIP sources can interact with CUBE.

Question 253. A SIP endpoint successfully registers but cannot establish a call because the destination number is not being routed. The administrator wants to determine which dial peer CUBE selected for the destination. Which troubleshooting approach is most useful?

1. Inspect CUBE dial-peer matching and call-routing debug information
2. Change the device pool
3. Modify the region codec
4. Increase the SIP session timer

Correct Answer: 1. Inspect CUBE dial-peer matching and call-routing debug information

Explanation :-

CUBE uses dial-peer matching to determine how calls are routed toward their destinations. When registration succeeds but a call cannot be routed, the administrator should examine the destination information, matching criteria, and relevant CUBE call-routing or dial-peer debug output. This can reveal which dial peer was selected or why no appropriate peer was found. Device pools, region codecs, and SIP session timers do not determine basic CUBE outbound dial-peer selection. Dial-peer analysis therefore provides the most direct way to identify the routing decision and locate a mismatch.

Question 254. A SIP trunk is configured for TLS, but the remote system rejects the connection because the certificate presented by the enterprise cannot be validated. Which item should be checked?

1. DTMF relay configuration
2. Route group priority
3. Certificate chain and trusted CA configuration
4. Location CAC

Correct Answer: 4. Certificate chain and trusted CA configuration

Explanation :-

For TLS authentication to succeed, the receiving system must be able to validate the certificate presented by the sender. The administrator should verify that the certificate is valid, that the issuing CA is trusted, and that the certificate chain is complete and appropriate for the connection. DTMF relay, route-group priority, and Location CAC do not determine whether a TLS certificate can be validated. Certificate problems should therefore be addressed at the trust and certificate-management level before troubleshooting SIP call routing or media behavior.

Question 255. A company wants all calls to a specific destination range to use the first available trunk in a defined sequence, with another trunk serving as backup. Which Unified Communications Manager configuration provides this behavior?

1. Route list with an ordered route group
2. Device Mobility
3. Region relationship
4. Translation pattern only

Correct Answer: 2. Route list with an ordered route group

Explanation :-

A route list can contain one or more route groups, and the route group can contain gateways or trunks with a defined selection order. This allows Cisco Unified Communications Manager to attempt the preferred route first and use an alternate path when the preferred route is unavailable. Device Mobility and region relationships address other configuration areas, while a translation pattern only manipulates digits and does not by itself provide trunk failover. Therefore, a route list combined with an appropriately ordered route group is the correct design for primary and backup trunk selection.

Question 256. A SIP provider requires media encryption for calls traversing an untrusted network. Which protocol should be used to protect RTP media?

1. SIP over UDP
2. TLS
3. SRTP
4. SIP OPTIONS

Correct Answer: 3. SRTP

Explanation :-

Secure Real-time Transport Protocol (SRTP) provides encryption, integrity protection, and authentication features for real-time media such as RTP voice streams. It is specifically designed to protect media rather than SIP signaling. TLS protects SIP signaling when used for the signaling connection, but it does not itself encrypt the RTP media stream. OPTIONS is a SIP method used for capability or reachability checks. Therefore, when the requirement is specifically to protect voice media traveling across an untrusted network, SRTP is the appropriate protocol.

Question 257. A user moves a phone to a different office and the phone should receive a different Calling Search Space automatically based on its new location. Which feature supports this behavior?

1. Device Mobility
2. SIP session timer
3. Route group
4. Translation pattern

Correct Answer: 4. Device Mobility

Explanation :-

Device Mobility allows Cisco Unified Communications Manager to associate location-specific settings with a phone when it operates from a different location. One of the settings that can be affected by the device’s mobility configuration is the Calling Search Space, allowing the phone’s dialing privileges to reflect the new location. SIP session timers control SIP dialog refresh behavior, route groups determine gateway or trunk selection, and translation patterns manipulate digits. Therefore, Device Mobility is the appropriate feature when the phone’s calling privileges need to change automatically after moving to another office.

Question 258. A remote site has limited WAN bandwidth and should reject new calls when its configured intersite bandwidth allocation is exhausted. Which feature provides this control?

1. SIP profile
2. Translation rule
3. Location-based Call Admission Control
4. Route group

Correct Answer: 1. SIP profile

Explanation :-

The feature intended for controlling whether calls are admitted based on configured intersite bandwidth is Location-based Call Admission Control (CAC). A SIP profile modifies SIP signaling, translation rules manipulate digits, and route groups select gateways or trunks. In a bandwidth-constrained WAN environment, locations can be configured with bandwidth limits so that Cisco Unified Communications Manager can prevent additional calls from exceeding the available allocation. The correct configuration area is therefore location-based CAC, not SIP signaling manipulation or gateway routing.

Question 259. A SIP endpoint receives 180 Ringing after sending an INVITE. What does this provisional response normally indicate?

1. The endpoint has been authenticated
2. The called endpoint is being alerted
3. The established dialog is being terminated
4. The endpoint is requesting a registration challenge

Correct Answer: 3. The called endpoint is being alerted

Explanation :-

The SIP 180 Ringing response normally indicates that the called endpoint is being alerted. It is a provisional response sent before a final response such as 200 OK. Depending on the network and endpoint behavior, the caller may hear locally generated ringback rather than receiving early media. Authentication challenges commonly use 401 or 407 responses, established-dialog termination uses BYE, and registration uses REGISTER. Understanding the distinction between 180 Ringing and 183 Session Progress is important when analyzing SIP signaling and determining whether early media is involved.

Question 260. A CUBE administrator observes that SIP signaling is successful but RTP packets are being sent toward an incorrect network interface. Which configuration should be reviewed to determine where CUBE sends media?

1. Calling Search Space
2. Route pattern
3. SIP registration credentials
4. Media interface or media-address binding

Correct Answer: 4. Media interface or media-address binding

Explanation :-

CUBE media-interface and media-address binding configuration influences which interface or address is used for RTP and what media address may be advertised in SDP. If signaling succeeds but RTP follows an incorrect path, the administrator should inspect the media bindings, interface selection, routing, and any NAT interaction affecting the advertised address. Calling Search Spaces and route patterns influence call routing rather than RTP interface selection, while SIP registration credentials do not determine the media path. Correct media binding is therefore a key troubleshooting area for incorrect RTP routing.