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Question 341. A Cisco Unified Communications Manager administrator needs to route an external call through a gateway only after the dialed number matches a specific pattern. Which dial-plan element should be configured?

1. Line Group
2. Device Pool
3. Hunt Pilot
4. Route Pattern

Correct Answer: 4. Route Pattern

Explanation :-

A Route Pattern in Cisco Unified Communications Manager defines a pattern that UCM uses to match a dialed destination and determine how the call should be routed. It can be associated with a Route List, gateway, or trunk depending on the deployment. Route Patterns are commonly used for PSTN access, intersite routing, and other destinations that require specific digit-pattern matching. A Line Group contains hunt members, a Device Pool provides common device configuration, and a Hunt Pilot represents a number used to reach a hunt group. When the requirement is to identify an external destination based on the digits dialed and then select an outbound route, the Route Pattern is the appropriate configuration element.

Question 342. A UCM deployment has a common PSTN route pattern used by several sites. Each site must send calls through its own local gateway. Which feature allows the same routing configuration to use different local gateways?

1. Local Route Group
2. SIP PRACK
3. Call Park
4. Global Dial Plan Replication

Correct Answer: 1. Local Route Group

Explanation :-

Local Route Group enables Cisco Unified Communications Manager to use a route group associated with the originating device’s location while allowing common route patterns to be shared across sites. This is particularly useful in centralized call-processing architectures where each branch has its own PSTN gateway. Instead of creating separate route patterns for every site, administrators can use common routing logic and associate devices with the appropriate local route group. SIP PRACK handles reliable provisional responses, Call Park temporarily holds calls, and Global Dial Plan Replication distributes dial-plan information between clusters. Local Route Group therefore provides a scalable method for combining centralized dial-plan administration with local PSTN access.

Question 343. A Cisco IOS voice gateway has several outbound dial peers that match the same destination pattern. The administrator wants to establish a primary and secondary path. Which dial-peer parameter can be used to influence selection preference?

1. Session target
2. Codec class
3. Preference
4. Voice translation profile

Correct Answer: 3. Preference

Explanation :-

The preference value on Cisco IOS voice dial peers can influence which matching dial peer is selected when multiple candidates are available. A lower preference value represents a more preferred path when the applicable dial peers otherwise meet the matching requirements. This can be used to create primary and backup routing paths. The session target identifies where the call should be sent after a dial peer is selected, while codec configuration controls media negotiation. Voice translation profiles manipulate calling or called numbers. When troubleshooting unexpected dial-peer selection, administrators should examine destination-pattern matching, preference, dial-peer type, and the overall dial-peer selection process rather than assuming that the first configured dial peer will always be selected.

Question 344. A CUBE administrator needs to modify the From header in outbound SIP messages to satisfy a service-provider interoperability requirement. Which capability should be investigated?

1. SRST
2. SIP profile
3. Call Admission Control
4. Local Route Group

Correct Answer: 2. SIP profile

Explanation :-

SIP profiles on Cisco CUBE can modify selected elements of SIP messages to address interoperability requirements between different SIP implementations. If a service provider expects a particular value or format in the From header, the administrator can use an appropriate SIP profile to manipulate the signaling before it is sent toward the provider. SRST provides local survivability during loss of UCM connectivity, CAC controls network-resource admission, and Local Route Group is a UCM routing feature. SIP traces should be reviewed before implementing a profile so that the administrator understands the exact header content being received and transmitted. This helps ensure that only the necessary signaling elements are changed.

Question 345. A remote branch loses its WAN connection to UCM. Phones register with the local gateway, but users cannot make internal calls between branch phones. Which area should be checked first?

1. SRST configuration and local dial plan
2. UCM SIP Route Patterns
3. Global Dial Plan Replication
4. CUBE tenant configuration

Correct Answer: 1. SRST configuration and local dial plan

Explanation :-

When Cisco IP phones enter SRST mode, the local gateway becomes responsible for providing survivable call processing. If phones are registered locally but cannot call one another, the administrator should verify SRST configuration, local directory-number assignments, dial peers, destination patterns, and the local call-routing behavior. Central UCM route patterns and Global Dial Plan Replication may not be available during the WAN outage, so they are not the first place to troubleshoot local SRST calls. CUBE tenants are associated with border-element trunk separation and are unrelated to basic local SRST phone-to-phone routing. A working SRST registration state does not automatically guarantee that every desired dial-plan feature has been configured locally.

Question 346. A SIP trunk sends an INVITE, receives 100 Trying, and then receives no further response. The administrator wants to determine whether the transaction is retransmitting and eventually timing out. Which information is most useful?

1. Device Mobility settings
2. SIP transaction timers and packet captures
3. UCM Line Group membership
4. Route List names

Correct Answer: 2. SIP transaction timers and packet captures

Explanation :-

SIP transaction timers control retransmission and timeout behavior for SIP transactions. When an INVITE receives an initial provisional response such as 100 Trying but no subsequent response, examining SIP traces and packet captures can reveal whether the request is being retransmitted, whether responses are returning, and where the transaction eventually times out. Device Mobility, Line Group membership, and Route List names do not directly explain SIP transaction timing. Administrators should correlate timestamps in traces with the relevant SIP transaction behavior and verify network connectivity between the SIP peers. This approach can help distinguish a remote application problem from packet loss, routing issues, firewall behavior, or SIP transaction timeout.

Question 347. A SIP endpoint receives a 180 Ringing response but the caller hears no ringback. The administrator wants the caller to receive media generated by the remote network before answer. Which signaling behavior should be investigated?

1. SIP REGISTER
2. BYE processing
3. Early media using a provisional response with SDP
4. Extension Mobility

Correct Answer: 3. Early media using a provisional response with SDP

Explanation :-

A 180 Ringing response indicates that the destination is being alerted, but it does not necessarily establish an RTP media path. If the remote network generates ringback or an announcement that should be heard before the call is answered, early media may be established through a provisional response such as 183 Session Progress containing SDP. Administrators should examine the SIP signaling and SDP as well as the RTP path between the endpoints or CUBE. REGISTER handles endpoint registration, BYE terminates an established dialog, and Extension Mobility concerns user profiles. Therefore, early-media signaling and media-path configuration are the relevant areas when no pre-answer audio is heard.

Question 348. A UCM administrator wants to convert a user’s locally dialed PSTN number into a standardized +E.164 representation before routing. Which concept is most directly related to this requirement?

1. DTMF relay
2. SIP session timer
3. Number normalization
4. RTP transcoding

Correct Answer: 3. Number normalization

Explanation :-

Number normalization converts telephone numbers from local or site-specific formats into a standardized representation. In Cisco collaboration environments, +E.164-style globalized numbers can provide a consistent identity for routing and call processing across locations. A normalized number can then be used by routing rules, applications, and other collaboration components without requiring every site to maintain a completely different representation. DTMF relay transports keypad digits, SIP session timers maintain established SIP sessions, and RTP transcoding changes media codecs. When a design requires consistent global telephone-number representation regardless of the user’s dialing method, number normalization should be considered as part of the dial-plan strategy.

Question 349. A CUBE receives a SIP call from a provider, but the call is unexpectedly sent to an internal destination instead of the intended external routing path. Which area should be examined first?

1. SIP registration expiration
2. Incoming and outgoing dial-peer matching
3. Extension Mobility
4. UCM Device Mobility

Correct Answer: 2. Incoming and outgoing dial-peer matching

Explanation :-

CUBE call routing depends heavily on correct dial-peer matching. When a call arrives from a provider and is unexpectedly routed toward an internal destination, administrators should inspect the incoming dial peer selected for the call and the outgoing dial peer selected afterward. The received called number, calling number, URI information, destination patterns, preferences, and session targets should be compared with the configured dial peers. SIP registration expiration and Extension Mobility are not normally responsible for CUBE dial-peer selection, while Device Mobility is a UCM endpoint-location feature. Debugging commands and SIP traces can help confirm which dial peers are selected and what signaling information is being used for the decision.

Question 350. A service provider requires calls from the enterprise to present a specific ten-digit calling number. The internal user has only a four-digit extension. Which configuration should be reviewed?

1. Calling-party transformation
2. Call Admission Control
3. SIP PRACK
4. Hunt Pilot

Correct Answer: 1. Calling-party transformation

Explanation :-

Calling-party transformation allows Cisco Unified Communications Manager to modify the calling number presented toward an external destination. An internal four-digit extension can be transformed into the organization’s full ten-digit number when required by a PSTN carrier. The exact configuration may involve route patterns, route lists, gateways, or calling-party transformation settings depending on the UCM design. CAC controls bandwidth resources, PRACK acknowledges reliable provisional responses, and Hunt Pilots distribute calls among hunt members. When an external provider requires a specific calling-number format, administrators should verify both the transformation rules and the resulting SIP or ISDN signaling to confirm that the expected number is actually being presented.

Question 351. A UCM administrator creates a Route List containing two Route Groups. The first group contains a primary PSTN gateway and the second contains a backup gateway. What is the main purpose of this configuration?

1. URI normalization
2. Call hunting among users
3. Alternate outbound routing
4. SIP registration

Correct Answer: 3. Alternate outbound routing

Explanation :-

A Route List can contain multiple Route Groups and can provide an ordered routing structure for outbound calls. By placing a primary gateway in one Route Group and a backup gateway in another, an administrator can design alternate routing when the primary path cannot complete the call under the applicable routing conditions. This is different from user call hunting, which uses Hunt Pilots, Hunt Lists, and Line Groups. URI normalization changes SIP addressing formats, while SIP registration establishes endpoint registration. Route Lists and Route Groups are therefore important tools for building resilient PSTN and intersite routing designs in Cisco Unified Communications Manager.

Question 352. A UCM administrator needs to route SIP URI calls based on a specific domain such as example.com. Which configuration should be reviewed?

1. Device Mobility
2. SIP Route Pattern
3. Call Park
4. SRST

Correct Answer: 2. SIP Route Pattern

Explanation :-

SIP Route Patterns allow Cisco Unified Communications Manager to match SIP URI destinations and route calls toward the appropriate trunk or routing resource. A pattern can be designed to match a specific URI structure or domain, such as destinations associated with example.com. This makes SIP Route Patterns useful in URI-based dialing and intercluster or interorganization SIP deployments. Device Mobility changes device settings according to location, Call Park provides temporary call holding, and SRST provides survivable local call processing. When URI calls are not reaching the intended trunk, administrators should verify the URI format, SIP Route Pattern, SIP trunk configuration, and any normalization rules that modify the destination before routing.

Question 353. A branch phone is configured for Device Mobility. When the phone is moved to another office, it continues using the original location’s calling-search space. Which area should be investigated?

1. Device Mobility information and roaming-device configuration
2. CUBE tenant configuration
3. SIP DTMF negotiation
4. Route Group preference

Correct Answer: 1. Device Mobility information and roaming-device configuration

Explanation :-

Device Mobility allows Cisco Unified Communications Manager to apply location-specific settings when a device operates from a different physical location. If a phone continues using its original calling-search space after being moved, administrators should verify the Device Mobility configuration, physical location information, device pool relationships, roaming-sensitive settings, and whether the phone has correctly identified its new location. CUBE tenants control border-element separation, DTMF negotiation concerns keypad signaling, and Route Group preference affects outbound routing. The problem is therefore most directly related to Device Mobility processing. Reviewing the device’s current location and the configuration associated with that location can reveal why the expected settings were not applied.

Question 354. A UCM deployment uses a centralized call-processing model with multiple remote sites. The administrator wants to prevent calls from consuming more WAN bandwidth than is available between sites. Which configuration should be verified?

1. SIP profile
2. Hunt List
3. Call Admission Control and location bandwidth
4. Extension Mobility

Correct Answer: 3. Call Admission Control and location bandwidth

Explanation :-

Call Admission Control and location bandwidth settings work together to control voice-resource usage across constrained network links. Cisco Unified Communications Manager can use configured bandwidth relationships between locations to determine whether another call should be admitted. This prevents excessive voice traffic from overwhelming a WAN connection and potentially degrading active calls. SIP Profiles manipulate selected SIP signaling, Hunt Lists distribute calls among hunt groups, and Extension Mobility manages user profiles. When the requirement is to enforce a limit on the number of calls or amount of voice bandwidth crossing a WAN path, administrators should examine the CAC design, location assignments, bandwidth limits, and any relevant RSVP or network-based call-admission mechanisms.

Question 355. A CUBE administrator observes repeated SIP requests from an unknown external source attempting to place calls through the enterprise gateway. Which risk should be considered most seriously?

1. Toll fraud
2. Device Mobility failure
3. Number normalization failure
4. Hunt-group overflow

Correct Answer: 1. Toll fraud

Explanation :-

Unauthorized SIP requests against a voice gateway can create a significant toll-fraud risk if attackers discover a way to originate calls through the enterprise. CUBE deployments should therefore restrict signaling sources, use appropriate access controls, configure restrictive dial peers, and avoid allowing untrusted callers to reach unauthorized outbound destinations. Repeated requests from unknown sources should be investigated using SIP logs, access-control mechanisms, and call-routing policies. Device Mobility, number normalization, and hunt-group overflow do not explain the security risk associated with unauthorized external call attempts. Voice gateways should be treated as security-sensitive infrastructure because successful abuse can result in unauthorized calls and potentially substantial carrier charges.

Question 356. A SIP endpoint sends an INVITE, then the user hangs up before the destination answers. Which SIP request normally informs the remote side that the pending call attempt should be stopped?

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

Correct Answer: 4. CANCEL

Explanation :-

CANCEL is used to terminate a pending SIP transaction before it receives a final response. For example, if a caller sends an INVITE and then hangs up while the destination is still ringing, the caller’s system can send CANCEL to stop the outstanding call attempt. BYE is normally used after a SIP dialog has been established and is therefore not the appropriate method for canceling an unanswered INVITE transaction. PRACK acknowledges reliable provisional responses, while UPDATE can modify session parameters in applicable SIP dialogs. Understanding the distinction between CANCEL and BYE is especially important when analyzing SIP traces because the correct method depends on whether the call has reached an established dialog.

Question 357. A SIP call connects, but one endpoint cannot hear audio from the other. SIP signaling is bidirectional and the SDP shows different private addresses on each side. Which troubleshooting approach is most appropriate?

1. Check the media path, SDP addresses, NAT, and firewall behavior
2. Reconfigure the UCM Hunt List
3. Delete the user’s Extension Mobility profile
4. Change the SIP session timer

Correct Answer: 1. Check the media path, SDP addresses, NAT, and firewall behavior

Explanation :-

One-way audio commonly indicates a media-path problem rather than a basic SIP signaling problem. SDP communicates the IP addresses and ports that should be used for RTP, so incorrect private addresses across NAT boundaries or routed networks can prevent one side from reaching the other. Administrators should compare the SDP offer and answer, verify routing and NAT behavior, inspect firewall policies, and use packet captures to determine whether RTP is actually being transmitted and received. Hunt Lists and Extension Mobility do not control RTP reachability, while changing SIP session timers generally does not correct a direct media-address problem. Media troubleshooting should therefore focus on the actual RTP path.

Question 358. A CUBE deployment must connect two SIP networks that support different codecs. Calls establish signaling successfully, but no common codec is available. Which media resource may be required?

1. SIP registrar
2. Transcoder
3. Hunt Pilot
4. Route List

Correct Answer: 2. Transcoder

Explanation :-

A transcoder can convert media from one supported codec to another when two endpoints or networks do not share a common codec. In a CUBE or UCM environment, transcoding may be required when a service provider supports codecs that differ from those used internally. Signaling can complete successfully even when media negotiation cannot identify a mutually supported codec, so codec compatibility should be examined in SIP and SDP traces. A SIP registrar manages endpoint registration, a Hunt Pilot handles call distribution, and a Route List selects routing resources. When the fundamental problem is incompatible media codecs, the availability and placement of an appropriate transcoding resource should be investigated.

Question 359. A UCM administrator wants a user to log into any supported shared phone and receive the user’s configured device profile. Which feature provides this capability?

1. Local Route Group
2. Extension Mobility
3. Call Admission Control
4. CUBE tenant

Correct Answer: 2. Extension Mobility

Explanation :-

Cisco Unified Communications Manager Extension Mobility allows users to log into supported phones and load their configured device profile. This enables users to access their personal phone settings, directory numbers, speed dials, and other profile-based configuration from compatible shared devices. Extension Mobility is different from Device Mobility, which applies location-specific configuration when a device moves between physical locations. Local Route Group is used for location-aware call routing, CAC controls network-resource admission, and CUBE tenants separate border-element configurations. When users need to move between shared phones while retaining their personalized phone configuration, Extension Mobility is the appropriate UCM feature.

Question 360. A branch’s WAN connection to UCM fails. The administrator wants local phones to continue registering and making basic calls through the branch gateway. Which technology should be configured and validated?

1. Global Dial Plan Replication
2. SIP Route Pattern
3. Cisco SRST
4. UCM Extension Mobility

Correct Answer: 3. Cisco SRST

Explanation :-

Cisco Unified Survivable Remote Site Telephony, or SRST, allows a branch gateway to provide local call-processing capabilities when phones can no longer reach the centralized Cisco Unified Communications Manager. During the outage, supported phones can register with the local gateway and continue making basic calls according to the available SRST configuration. Administrators should validate SRST registration, local directory numbers, dial peers, PSTN routing, and any required voice ports or trunks. Global Dial Plan Replication is used for distributing dial-plan information between clusters, SIP Route Patterns handle URI routing, and Extension Mobility provides user-profile functionality. SRST is therefore the technology specifically designed for branch voice survivability during WAN or UCM connectivity loss.