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Question 161. A Unified Communications Manager administrator wants to manipulate the called number before the call is routed to the PSTN. Which feature is specifically designed for this purpose?

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

Correct Answer: 3. Translation pattern

Explanation :-

A translation pattern is used by Cisco Unified Communications Manager to transform dialed digits during call processing. It can add, remove, or modify digits before the resulting number continues through the dial plan. This is useful when users dial an abbreviated or local format but the PSTN requires a different numbering format. Route groups determine which gateways or trunks are selected, regions control codec and bandwidth relationships, and device pools provide shared device configuration. The administrator should verify the translation pattern’s partition, CSS accessibility, transformation settings, and the resulting number before testing the downstream PSTN route.

Question 162. A route pattern is configured correctly, but only certain users can access it. The administrator wants to understand why. Which configuration relationship should be examined?

1. Partition and Calling Search Space
2. Region and codec preference
3. Device pool and NTP
4. SIP profile and DTMF relay

Correct Answer: 1. Partition and Calling Search Space

Explanation :-

A route pattern is placed in a partition, while a user’s Calling Search Space determines which partitions that user can access. Consequently, two users can dial the same number but receive different results if their effective CSS values differ. Region and codec settings affect media negotiation, device pools provide shared configuration, and SIP profiles manipulate SIP signaling. To troubleshoot selective route-pattern access, the administrator should identify the route pattern’s partition and compare it with the effective CSS applied to the calling line or device. This relationship is fundamental to Unified Communications Manager dial-plan access control.

Question 163. A CUBE router receives an INVITE from a trusted SIP provider, but the called number must be transformed before the call is sent to an internal SIP trunk. Which configuration can perform this transformation?

1. SIP OPTIONS
2. Region configuration
3. Device Mobility
4. Voice translation rule and profile

Correct Answer: 4. Voice translation rule and profile

Explanation :-

Voice translation rules and profiles provide digit manipulation on Cisco voice gateways and CUBE. They can transform the called number received from the provider into a format recognized by the internal dial plan. The translation profile determines how the rule is applied, while the translation rule defines the matching and replacement behavior. SIP OPTIONS provides reachability or capability checks, region configuration controls codec and bandwidth relationships, and Device Mobility applies location-specific endpoint settings. The engineer should verify the incoming number, translation direction, rule matching, and final destination before troubleshooting the internal SIP trunk.

Question 164. A company wants calls between two Unified Communications Manager regions to prefer a low-bandwidth codec. Where should the codec relationship be configured?

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

Correct Answer: 2. Region configuration

Explanation :-

Region configuration defines codec preferences and bandwidth relationships between Unified Communications Manager regions. By configuring the appropriate codec relationship, the administrator can influence media negotiation for calls crossing the WAN between those regions. Route patterns perform destination matching, Calling Search Spaces control dial-plan access, and translation patterns manipulate digits. When selecting a low-bandwidth codec, the administrator should also consider endpoint support and the overall media design. Verification should be performed using call statistics or signaling to confirm the codec that was actually negotiated rather than relying solely on the configured preference.

Question 165. A branch loses WAN connectivity to centralized call control. Phones register to the branch router and internal calls work, but external calls do not. Which configuration is most important for PSTN routing in this condition?

1. Central CUCM route list
2. Remote device mobility settings
3. Unified Communications Manager region
4. Local SRST dial peers and PSTN gateway

Correct Answer: 4. Local SRST dial peers and PSTN gateway

Explanation :-

During SRST operation, the branch router provides local survivable call processing, so local dial peers and gateway configuration become essential for external calls. The engineer should verify destination patterns, outbound dial-peer matching, gateway availability, and connectivity to the PSTN service. Centralized route lists and regions are not the primary call-routing mechanisms while the branch is operating independently from Unified Communications Manager. Device Mobility is also unrelated to external call routing in SRST. A useful troubleshooting sequence is to confirm phone registration, test dial-peer matching, verify the gateway, and then inspect the PSTN signaling path.

Question 166. A SIP endpoint receives a reliable provisional response and the network requires explicit acknowledgment of that response. Which SIP method should be expected?

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

Correct Answer: 2. PRACK

Explanation :-

PRACK is the SIP method used to acknowledge reliable provisional responses. Reliable provisional-response behavior can be important when early-media or other provisional signaling must be delivered reliably between SIP systems. REFER is associated with referred operations such as call transfer, BYE terminates an established dialog, and CANCEL terminates a pending request such as an unanswered INVITE. When troubleshooting PRACK, the engineer should inspect the provisional response, the reliability-related headers, the PRACK transaction, and the final response to the PRACK. Interoperability issues can occur when one system expects reliable provisional responses while the other does not support the same behavior.

Question 167. A SIP call is established successfully, but the caller cannot hear the remote party. The remote party can hear the caller. Which investigation is most appropriate?

1. Calling Search Space only
2. RTP media path, SDP addresses, and NAT/firewall behavior
3. Route-pattern partition only
4. Device Mobility group

Correct Answer: 1. RTP media path, SDP addresses, and NAT/firewall behavior

Explanation :-

One-way audio generally indicates a media-path problem rather than a basic SIP call-routing problem. The engineer should inspect SDP addresses and ports, RTP packet flow, routing, NAT translations, firewall policies, and CUBE media-interface configuration where applicable. Calling Search Spaces and route-pattern partitions determine whether destinations can be selected, while Device Mobility controls location-specific endpoint settings. Packet captures are particularly useful because they can show whether RTP is leaving one endpoint, arriving at the other side, or being blocked somewhere in between. The direction of the missing audio can provide an important clue about the failing media path.

Question 168. A CUBE router has two SIP providers and should use a different provider depending on the destination SIP domain. Which capability can support this routing design?

1. DHCP option 150
2. Location CAC
3. SIP URI-based dial-peer matching
4. SRST fallback

Correct Answer: 3. SIP URI-based dial-peer matching

Explanation :-

SIP URI-based dial-peer matching allows CUBE to use SIP URI information, such as the destination domain, when selecting an outbound dial peer. This makes it possible to route calls for different SIP domains toward different providers or destinations. DHCP option 150 supports Cisco IP phone provisioning, location CAC manages media admission, and SRST provides survivable call processing. The engineer should inspect the actual Request-URI received by CUBE and ensure that the dial-peer matching criteria correspond to the provider domains. Session targets and outbound interface behavior should also be verified.

Question 169. A SIP provider requires the enterprise to send a specific value in the P-Asserted-Identity header. Which feature can manipulate that SIP header?

1. Route list
2. SIP profile or normalization
3. Region
4. Location

Correct Answer: 2. SIP profile or normalization

Explanation :-

SIP profiles or normalization mechanisms can modify selected SIP headers to meet interoperability requirements. P-Asserted-Identity is a SIP identity-related header that may need to be changed to satisfy provider-specific expectations. Route lists control gateway or trunk selection, regions influence codec and bandwidth relationships, and locations participate in call admission control. Before implementing a normalization rule, the engineer should capture the current SIP message and identify the exact required header format. After applying the change, another SIP trace should verify that the desired value is inserted or modified without unintentionally changing other signaling information.

Question 170. A SIP call consistently disconnects after the same negotiated interval when session refreshes are not exchanged successfully. Which mechanism should be investigated?

1. DTMF relay
2. Device Mobility
3. Route-group priority
4. SIP session timers

Correct Answer: 4. SIP session timers

Explanation :-

SIP session timers establish a negotiated interval during which the session must be refreshed. If the expected refresh does not arrive, the session can eventually be terminated. A consistent disconnect interval is therefore an important clue that session-timer negotiation or refresh traffic should be investigated. The engineer should inspect Session-Expires, Min-SE, refresher selection, UPDATE or re-INVITE messages, and any intermediary device that could block the refresh. DTMF relay, Device Mobility, and route-group priority address different aspects of collaboration operation and do not normally explain predictable SIP session expiration.

Question 171. An administrator needs a route list to use Gateway-A first and Gateway-B only when the preferred path is unavailable. How should the gateways be organized?

1. In a route group with the desired selection order
2. In separate partitions
3. In different regions
4. In separate device pools

Correct Answer: 1. In a route group with the desired selection order

Explanation :-

A route group organizes gateways or trunks that can be used as part of Unified Communications Manager routing. The administrator can configure the route group so that the preferred gateway is selected first and an alternate gateway can be considered according to the configured selection method. The route group can then be referenced by a route list. Partitions control dial-plan access, regions manage codec and bandwidth relationships, and device pools provide common device settings. Gateway redundancy should be tested by simulating the appropriate failure condition and confirming that call processing selects the intended alternate path.

Question 172. A user moves a phone to a different office, but the phone continues using the original site’s region and location settings. Which feature should be checked?

1. SIP OPTIONS
2. Device Mobility
3. Route pattern
4. DTMF relay

Correct Answer: 3. Device Mobility

Explanation :-

Device Mobility is designed to provide location-specific configuration when a supported phone moves between physical locations. If the phone continues using settings from the previous office, the administrator should inspect Device Mobility configuration, device mobility groups, physical locations, and the device pools associated with those locations. SIP OPTIONS is used for reachability or capability checks, route patterns perform destination matching, and DTMF relay controls keypad signaling. The engineer should also verify how the system determines the phone’s current location and whether device- or line-level settings are overriding the expected mobility-derived configuration.

Question 173. A CUBE router receives a SIP INVITE from an unknown Internet address and attempts to route it toward the PSTN. Which security measure can help reduce unauthorized call attempts?

1. Increase the SIP session timer
2. Disable RTP
3. Restrict inbound SIP sources with access controls and trusted-peer policies
4. Change the region codec

Correct Answer: 4. Restrict inbound SIP sources with access controls and trusted-peer policies

Explanation :-

CUBE deployments exposed to external networks should restrict which SIP sources are permitted to establish signaling relationships. Access-control policies and trusted-peer restrictions can help prevent unauthorized systems from sending calls through the enterprise and reduce toll-fraud exposure. Increasing session timers does not provide source authorization, disabling RTP would prevent normal media operation, and changing codec preferences does not address unauthorized signaling. The engineer should combine source restrictions with appropriate dial-peer policies, authentication where required, monitoring, and least-privilege routing. Security controls should be tested to confirm that legitimate provider traffic continues to operate.

Question 174. A service provider sends a SIP INVITE with a national-format called number. The enterprise dial plan expects E.164 numbers. Which operation should occur before normal destination routing?

1. Digit translation
2. Region selection
3. Location CAC
4. DTMF negotiation

Correct Answer: 2. Digit translation

Explanation :-

When the incoming number format does not match the enterprise dial plan, digit translation can convert the number into the expected format before routing continues. On Cisco voice gateways and CUBE, voice translation rules and profiles can add, remove, or modify digits to achieve the required E.164 representation. Region selection controls media characteristics, location CAC manages bandwidth admission, and DTMF negotiation concerns keypad signaling. The engineer should verify the exact incoming number, apply the translation in the correct direction, and confirm that the resulting E.164 number matches the intended outbound dial peer or Unified Communications Manager route pattern.

Question 175. A Unified Communications Manager administrator wants to prevent a group of users from accessing expensive international destinations while preserving access to domestic PSTN calls. Which dial-plan design is appropriate?

1. Use separate partitions and assign CSS values according to authorization
2. Change the RTP port range
3. Configure a different SIP codec
4. Enable SIP OPTIONS

Correct Answer: 1. Use separate partitions and assign CSS values according to authorization

Explanation :-

Separate partitions and carefully designed Calling Search Spaces allow administrators to control access to different classes of destinations. Domestic PSTN route patterns can be placed in a partition available to the users, while international patterns can reside in a restricted partition that their CSS does not include. RTP ports and codecs affect media transport rather than destination authorization, and SIP OPTIONS does not provide dial-plan access control. The engineer should test both permitted and restricted destinations and verify the effective CSS at the line and device levels to ensure the policy is applied consistently.

Question 176. A SIP provider requires encrypted media in addition to encrypted SIP signaling. Which combination addresses both requirements?

1. UDP and RTP-NTE
2. TCP and ordinary RTP
3. TLS and SRTP
4. SIP OPTIONS and SDP

Correct Answer: 3. TLS and SRTP

Explanation :-

TLS provides protection for SIP signaling, while Secure Real-time Transport Protocol, or SRTP, protects the media stream. Together they address the two separate security requirements of encrypted call-control signaling and protected audio or video media. RTP-NTE is a DTMF signaling method and does not encrypt SIP signaling or provide general media confidentiality. OPTIONS is a SIP capability or reachability method, while SDP describes session parameters rather than providing encryption itself. The engineer should verify certificate and trust configuration for TLS and confirm compatible SRTP negotiation and keying behavior for the media path.

Question 177. A SIP endpoint sends an OPTIONS request to a remote voice system while no active call exists. What is the most likely purpose?

1. To terminate a call
2. To check peer availability or capabilities
3. To transfer a call
4. To modify an established media session

Correct Answer: 2. To check peer availability or capabilities

Explanation :-

SIP OPTIONS can be used to query a remote SIP system for availability or supported capabilities without establishing a normal call. Periodic OPTIONS exchanges are commonly used as keepalive or reachability mechanisms for SIP trunks. BYE terminates an established dialog, REFER is commonly associated with call transfer, and UPDATE can modify session information when supported. Engineers troubleshooting SIP trunk availability should inspect OPTIONS response codes, timing, routing, firewall behavior, and whether the remote system responds consistently. A successful OPTIONS response indicates SIP-level responsiveness but does not guarantee that every call-routing or media function is working.

Question 178. A SIP endpoint is registered successfully but cannot place calls to a particular external destination. Other destinations work. Which troubleshooting area should be examined first?

1. Phone display settings
2. NTP synchronization only
3. Destination digit analysis and route-pattern matching
4. RTP encryption

Correct Answer: 4. Destination digit analysis and route-pattern matching

Explanation :-

If registration succeeds and other destinations work, a destination-specific routing issue should be investigated first. The engineer should examine the digits being dialed, route-pattern matching, partitions, Calling Search Spaces, translation patterns, and the route list or route group selected for the destination. Phone display settings do not determine routing, NTP synchronization is generally unrelated to a single destination failure, and RTP encryption becomes relevant after call signaling has established the session. A digit-analysis trace can help identify which routing object is selected or why no appropriate route pattern is accessible.

Question 179. A CUBE router must select a SIP dial peer based on a destination URI rather than only a numeric destination pattern. Which feature is relevant?

1. SIP URI matching
2. Location CAC
3. Device pool
4. Region configuration

Correct Answer: 3. SIP URI matching

Explanation :-

SIP URI matching enables CUBE to use URI information such as a user, host, or domain when selecting an appropriate dial peer. This is useful for SIP-to-SIP routing where destinations may be expressed as user@domain rather than traditional telephone numbers. Location CAC controls bandwidth admission, device pools provide shared endpoint settings, and region configuration controls codec and bandwidth relationships. When configuring URI-based routing, the engineer should examine the actual SIP Request-URI and ensure that the matching criteria correspond to the received syntax. The selected dial peer and session target should then be verified.

Question 180. During troubleshooting, a SIP INVITE is followed by a 180 Ringing response but no 183 response with SDP. What does the 180 response normally indicate?

1. The call has been terminated
2. The endpoint is being alerted
3. The endpoint has registered
4. The call has entered SRST

Correct Answer: 1. The endpoint is being alerted

Explanation :-

A 180 Ringing response is a provisional SIP response indicating that the called endpoint is being alerted. It does not by itself mean that early media has been established. A 183 Session Progress response can carry SDP and is commonly associated with early-media scenarios. BYE terminates an established dialog, REGISTER handles registration, and SRST is a survivability mechanism rather than a SIP response. When troubleshooting ringback, engineers should distinguish between locally generated ringback and network-provided early media, then inspect the SIP responses and SDP to determine whether media is being established before the final call answer.