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Cisco Implementing Cisco Collaboration Core Technologies (CLCOR, 350-801, v2.0 blueprint, CCNP Collaboration) (CLCOR) (CLCOR) — Questions 1–75

259 questions total · 4 pages · All types, answers revealed

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1
MCQhard

A Cisco Unified Communications Manager (CUCM) cluster is configured with a SIP trunk to a Cisco Unified Border Element (CUBE) for PSTN connectivity. Users report that outbound calls to a specific area code fail, but all other calls succeed. The CUCM route pattern for that area code is configured with a route list that includes the SIP trunk. The CUBE shows that it receives the INVITE but rejects it with a 488 Not Acceptable Here. Which CUCM configuration should be checked first to resolve this issue?

A.The SIP trunk's Inbound Calls > Significant Digits setting.
B.The region assigned to the SIP trunk, which defines the codec used for calls.
C.The route group's distribution algorithm.
D.The SIP trunk's Outbound Calls > Calling Party Selection setting.
AnswerB

A 488 Not Acceptable Here often results from a codec mismatch between CUCM and CUBE. The region assigned to the SIP trunk determines which codec is offered. If the region is set to a codec that CUBE does not support or that is not allowed on the dial peer, CUBE rejects the call with 488. Checking the region and associated codec is the first step.

Why this answer

A 488 Not Acceptable Here response from CUBE indicates that the media description (SDP) in the INVITE is not acceptable. The region assigned to the SIP trunk in CUCM defines the codec that will be offered for calls traversing that trunk. If the region forces a codec that CUBE cannot support or that is not configured on the dial peer, CUBE rejects the call.

Therefore, the region configuration should be verified first.

Exam trap

The trap here is assuming that a 488 error is always a CUBE dial-peer issue, when it can also originate from a CUCM region codec mismatch that forces an unsupported codec toward CUBE.

Full explanation →
2
Multi-Selectmedium

Which TWO of the following are valid methods for providing high availability for IP phones in a CUCM design?

Select 2 answers
A.Configuring multiple TFTP servers in Option 150
B.IP address redundancy via DHCP
C.SRST deployment
D.Load balancing via round-robin
E.CUCM Group configuration
AnswersC, E

SRST lets a branch router assume call control for its local IP phones when WAN connectivity to the CUCM cluster is lost. Phones fall back to the gateway and retain basic calling, satisfying the branch survivability requirement.

Why this answer

Option C (SRST deployment) is correct because Survivable Remote Site Telephony lets remote-site IP phones register with a local router/gateway when WAN connectivity to the CUCM cluster is lost, keeping basic call processing available and thus providing high availability for those phones. Option E (CUCM Group configuration) is correct because a CUCM Group defines an ordered list of CUCM nodes that a device pool's phones use for registration; if the primary subscriber fails, phones fail over to the next CUCM node in the group, which is the standard CUCM redundancy mechanism. Option A is not a high-availability method for phone registration: multiple TFTP servers in Option 150 only provide redundancy for configuration file download, not for call processing/registration.

Option B is invalid because DHCP cannot provide IP address redundancy for phones; DHCP only leases addresses and does not offer a failover address mechanism for the phone itself. Option D is not a CUCM phone high-availability method; CUCM uses a primary/secondary/tertiary node failover model via CUCM Groups and device pools, not round-robin load balancing of phone registrations.

Exam trap

CLCOR often tests the confusion between redundancy for configuration files (TFTP) and redundancy for call control (CUCM Groups, SRST).

Full explanation →
3
MCQeasy

What is the first step to enable Webex Hybrid Services in a new organization?

A.Register the organization in Control Hub.
B.Purchase a PSTN plan.
C.Install the Expressway-C software.
D.Assign licenses to all users.
AnswerA

Control Hub is the management portal for Webex services, and an organisation must first be registered there before any Hybrid Services connector or resource can be provisioned. Registration establishes the org identity and licensing context required for subsequent hybrid configuration.

Why this answer

The first step to enable Webex Hybrid Services is to register the organization in Control Hub (A), because Control Hub is the management portal where the organization's Webex identity, licenses, and hybrid service configuration live. Without an organization registered in Control Hub, there is no anchor point for connecting on-premises components like Expressway-C or Expressway-E. Only after registration can you download the connector software, generate certificates, and configure hybrid services such as Hybrid Calendar, Hybrid Call Service, or Hybrid Message Service.

Exam trap

CLCOR often tests ordering of operations — candidates jump to installing Expressway-C because it is the visible on-premises component, forgetting that Control Hub organization registration must happen first to generate the connector and token.

How to eliminate wrong answers

Option B (Purchase a PSTN plan) is wrong because PSTN connectivity is a calling feature unrelated to enabling hybrid services; hybrid services connect on-premises infrastructure to Webex cloud, not to the PSTN. Option C (Install the Expressway-C software) is wrong because Expressway-C cannot be registered to Webex until the organization exists in Control Hub and the connector has been downloaded and configured from there — installation is a later step. Option D (Assign licenses to all users) is wrong because user licensing is a post-registration administrative task and is not required to enable hybrid services at the organizational level.

Full explanation →
4
MCQmedium

Which deployment model should be chosen for Cloud Connected PSTN (CCP) if the organization wants to leverage their existing carrier contracts?

A.Local Gateway PSTN
B.Webex Calling PSTN via Cisco
C.Cloud Connected PSTN (CCP)
D.Webex Calling Dedicated Instance
AnswerC

Cloud Connected PSTN (CCP) lets an organisation pair Webex Calling with its existing PSTN carrier contracts and numbers, rather than mandating a Cisco-provided calling plan. This directly satisfies the requirement to leverage current carrier agreements.

Why this answer

Cloud Connected PSTN (CCP) lets an organization bring its own PSTN carrier into Webex Calling by peering the carrier's network with the Webex Calling cloud, so existing carrier contracts, rates, and DIDs are preserved. This is the deployment model explicitly designed for 'use my existing carrier' scenarios, as opposed to Cisco-provided PSTN or on-premises gateways. The carrier must be a Webex Calling CCP partner.

Exam trap

CLCOR often tests the confusion between 'bring your own carrier' (CCP) and 'Cisco-provided PSTN' or 'Local Gateway,' so candidates who pick Local Gateway miss that CCP is the cloud-peered, contract-preserving option.

How to eliminate wrong answers

Option A is wrong because Local Gateway PSTN requires an on-premises Cisco gateway (e.g., CUBE) to connect to the PSTN, which does not leverage a cloud-peered carrier contract and adds on-prem hardware. Option B is wrong because Webex Calling PSTN via Cisco means Cisco provides the PSTN connectivity and DIDs, which replaces rather than leverages the customer's existing carrier contract. Option D is wrong because Webex Calling Dedicated Instance is a deployment topology for dedicated (non-shared) Webex Calling infrastructure, not a PSTN connectivity model, and does not by itself address carrier contract reuse.

Full explanation →
5
MCQhard

A Cisco IOS gateway is connected to a WAN circuit where the physical line rate is 10 Mbps but the contracted CIR is only 4 Mbps. Voice quality suffers during bursts because the provider polices traffic above the CIR and drops excess packets. The engineer wants the gateway to smooth outbound traffic so the provider does not discard bursts. Which configuration element should be applied to the WAN interface?

A.A policy-map using the shape average 4000000 command applied outbound on the WAN interface.
B.A class-map matching voice RTP ports with a set dscp ef action applied inbound on the WAN interface.
C.A policy-map using the police 4000000 command applied outbound on the WAN interface.
D.The bandwidth 4000 command under the physical WAN interface configuration.
AnswerA

Traffic shaping buffers and delays excess packets to conform to a configured average rate, which matches a provider policing at the CIR. Setting shape average to 4000000 bits per second on the outbound WAN interface smooths bursts so the policer sees a rate within contract, reducing drops and preserving voice quality across the link.

Why this answer

When a provider polices at a CIR lower than the physical line rate, the enterprise side must pace its own transmissions. Traffic shaping with shape average set to the CIR buffers excess packets and releases them at a conforming rate, so the provider policer does not see bursts. Applying this outbound on the WAN interface protects voice and data from being discarded.

Exam trap

The trap here is confusing policing, which drops excess traffic, with shaping, which delays it to conform to the contracted rate.

Full explanation →
6
Multi-Selectmedium

Which TWO of the following are valid deployment models for the Webex Calling PSTN connectivity? (Choose two)

Select 2 answers
A.Webex Calling dedicated ISP.
B.Hybrid Cloud PSTN.
C.Local Gateway.
D.Peer-to-Peer PSTN.
E.Cloud Connected PSTN (CCP).
AnswersC, E

Local Gateway is a valid on-premises deployment model, connecting Webex Calling to your existing PSTN provider through a registered gateway at your site. It satisfies the scenario's requirement for a supported PSTN connectivity option, alongside Cloud Connected PSTN and Cisco PSTN, giving organisations direct control over voice trunks.

Why this answer

Webex Calling supports two valid PSTN connectivity deployment models: Local Gateway (C) and Cloud Connected PSTN (E). Local Gateway is correct because it uses an on-premises gateway (such as a Cisco IOS XE router or CUBE) registered to Webex Calling via the Local Gateway registration process, allowing the enterprise to connect its own PSTN trunks to the Webex Calling cloud. Cloud Connected PSTN (CCP) is correct because it lets a certified CCP provider deliver PSTN service directly to Webex Calling through the Cisco Cloud Connectivity platform, with no on-premises gateway required.

The remaining options are not valid Webex Calling PSTN deployment models: 'Webex Calling dedicated ISP' (A) is not a recognized model, 'Hybrid Cloud PSTN' (B) is not a Webex Calling PSTN option (hybrid refers to other deployment aspects), and 'Peer-to-Peer PSTN' (D) is not a supported architecture for Webex Calling PSTN connectivity.

Exam trap

CLCOR often tests the valid PSTN connectivity options, and candidates may be misled by plausible-sounding but non-existent models like 'Hybrid Cloud PSTN' or 'Peer-to-Peer PSTN'. The trap is selecting options that sound technical but are not official Webex Calling deployment models.

Full explanation →
7
MCQmedium

A network administrator is troubleshooting one-way audio issues between a Cisco IP Phone and a Cisco TelePresence endpoint. The path traverses a router with a configured class-map. Which command would you use to verify the packet markings on the ingress interface?

A.show class-map configuration
B.show ip rtp statistics
C.show qos interface
D.show policy-map interface
AnswerD

The service-policy is applied on the ingress interface, so show policy-map interface displays per-class matched and marked packet counters there. This confirms whether the class-map's marking (for example DSCP EF for voice) is actually being applied, revealing why one-way audio occurs.

Why this answer

The command `show policy-map interface` displays the actual packet markings and QoS statistics applied to an interface by a policy-map. Since the class-map is referenced within a policy-map, this command shows the number of packets matching each class and the marking actions (e.g., DSCP, CoS) that were applied. This is the only option that directly verifies the markings on the ingress interface.

Exam trap

CLCOR often tests the confusion between configuration verification commands (like `show class-map`) and operational verification commands (like `show policy-map interface`), so candidates must remember that only the latter shows actual packet markings and statistics.

How to eliminate wrong answers

Option A is wrong because `show class-map` only displays the class-map configuration (match criteria) and does not show any packet markings or interface statistics. Option B is wrong because `show ip rtp statistics` provides RTP stream statistics such as jitter and packet loss, but does not show QoS markings. Option C is wrong because `show qos interface` is not a valid Cisco IOS command for verifying packet markings; the correct command is `show policy-map interface`.

Full explanation →
8
MCQeasy

Which service must be running for a phone to receive its configuration file?

A.Cisco Database Layer Monitor
B.Cisco CallManager
C.Cisco TFTP
D.Cisco CTI Manager
AnswerC

Cisco TFTP delivers the XML configuration file to the phone during provisioning; the phone requests it by name from the TFTP server address supplied via DHCP option 150. Without this service running, the phone cannot download its configuration and remains unregistered.

Why this answer

The Cisco TFTP service must be running for a phone to receive its configuration file. When a phone boots, it requests its configuration file (e.g., SEP<MAC>.cnf.xml) from the TFTP server, which is a core CUCM service. Without TFTP, the phone cannot download its configuration, firmware, or dial plan, and will fail to register.

Exam trap

CLCOR often tests the specific role of each CUCM service, so the trap is confusing TFTP (configuration file delivery) with CallManager (call processing) or CTI Manager (application integration).

How to eliminate wrong answers

Option A is wrong because Cisco Database Layer Monitor monitors the database replication and health, not file transfer to phones. Option B is wrong because Cisco CallManager handles call processing and registration after the phone has its configuration, but it does not serve the configuration file itself. Option D is wrong because Cisco CTI Manager provides computer telephony integration for applications like Jabber and contact centers, not phone configuration file delivery.

Full explanation →
9
MCQmedium

A network engineer is deploying a new branch office with a Cisco IOS XE gateway that connects to the service provider via a 100-Mbps Ethernet WAN link. The engineer needs to ensure that interactive voice traffic always receives priority treatment during congestion, while preventing non-real-time traffic from being starved. Which QoS mechanism should be configured on the WAN interface to guarantee strict priority for voice while still servicing other queues?

A.Enable traffic shaping on the WAN interface to smooth out bursts and reduce congestion for all traffic types.
B.Configure weighted random early detection (WRED) on the WAN interface to drop non-voice packets before congestion occurs.
C.Configure a low-latency queue with the priority command inside a policy-map, and apply it to the WAN interface with service-policy output.
D.Apply class-based weighted fair queueing (CBWFQ) with a bandwidth guarantee for the voice class, and rely on the scheduler to prioritize voice.
AnswerC

The priority command in a policy-map creates a low-latency queue (LLQ) that provides strict priority scheduling for voice traffic while policing the priority class to a configured rate. This ensures voice gets absolute priority during congestion, and the policer prevents voice from starving other queues. Applying the policy-map to the WAN interface with service-policy output activates this behavior.

Why this answer

A low-latency queue created with the priority command inside a policy-map provides strict priority scheduling for voice traffic, ensuring that voice packets are always transmitted first when congestion occurs. The priority policer limits the amount of voice traffic that can enter the priority queue, preventing starvation of other queues. This mechanism is the standard way to guarantee voice quality on a WAN interface.

Exam trap

The trap here is assuming that a bandwidth guarantee alone ensures priority treatment, when in fact only the priority command creates a strict priority queue.

Full explanation →
10
MCQmedium

You are troubleshooting a scenario where internal calls between two sites are failing. The call fails immediately when dialed. What is the most effective way to verify that the Calling Search Space (CSS) has access to the destination partition?

A.Check the 'Route Filter' list.
B.Use the 'Route Pattern' report in Cisco Unified Reporting.
C.Reset the SIP trunk.
D.Use the Dialed Number Analyzer (DNA) tool.
AnswerD

Dialed Number Analyzer simulates the call routing path using the actual dial plan, showing precisely which partitions the CSS can reach and where the immediate failure occurs. It isolates CSS-to-partition mismatches faster than reading configuration or capturing traces.

Why this answer

The Dialed Number Analyzer (DNA) tool in Cisco Unified Communications Manager simulates a call from a specified calling device/DN to a dialed number, showing exactly which partitions, CSS, translation patterns, route patterns, and route lists are matched. It is the definitive tool for verifying whether a CSS has access to the destination partition and for pinpointing where dial plan processing fails.

Exam trap

CLCOR often tests the misconception that reports or trunk resets can diagnose CSS/partition access issues, when only DNA actually simulates the dial plan decision path.

How to eliminate wrong answers

Option A is wrong because Route Filters are used with route lists to filter route groups based on dialed digits or time-of-day, not to verify CSS-to-partition access. Option B is wrong because the Route Pattern report in Cisco Unified Reporting lists route patterns and their associated partitions/route lists but does not simulate a call or evaluate CSS access. Option C is wrong because resetting a SIP trunk is a disruptive action that does not diagnose dial plan/CSS issues and would not reveal partition access problems.

Full explanation →
11
MCQhard

A collaboration designer must ensure that inter-cluster calls between two CUCM clusters remain secure. Which mechanism must be implemented to ensure media remains encrypted from end to end?

A.Disable SIP inspection on the firewalls.
B.Configure a VPN between the two clusters.
C.Ensure SRTP is enabled on the SIP Trunk Security Profile.
D.Enable MTP (Media Termination Point) for all calls.
AnswerC

SRTP encrypts the media payload itself, so enabling it on the SIP Trunk Security Profile ensures calls traversing the inter-cluster trunk carry encrypted RTP end to end, satisfying the requirement that media stays protected between clusters.

Why this answer

To ensure media remains encrypted end-to-end between two CUCM clusters, SRTP must be enabled on the SIP Trunk Security Profile. This setting ensures that the media streams (RTP) are encrypted using SRTP, providing confidentiality and integrity for the call. Without SRTP, even if signaling is encrypted, the media could be sent in clear text.

Exam trap

CLCOR often tests the misconception that a VPN or firewall configuration alone can secure media, but the exam expects knowledge that SRTP must be explicitly enabled on the SIP trunk security profile for end-to-end media encryption.

How to eliminate wrong answers

Option A is wrong because disabling SIP inspection on firewalls does not encrypt media; it may allow SIP traffic to pass but does not provide encryption. Option B is wrong because a VPN between clusters encrypts all traffic at the network layer, but it does not ensure that the media itself is encrypted end-to-end; it only secures the tunnel, and media could still be unencrypted within the tunnel. Option D is wrong because enabling MTP for all calls does not encrypt media; MTPs are used for transcoding or DTMF conversion and can actually prevent encryption if not configured properly.

Full explanation →
12
MCQeasy

Which network condition is the most significant contributor to poor voice quality in a VoIP environment?

A.Low CPU usage on the router
B.Static IP addresses
C.Jitter
D.High bandwidth usage
AnswerC

Jitter, the variation in packet arrival delay, causes gaps and reordering that voice codecs cannot conceal, producing choppy, distorted audio. Consistent latency is tolerable, but variable delay is the dominant contributor to poor VoIP voice quality.

Why this answer

Jitter—variation in packet arrival delay—is the most significant contributor to poor voice quality because VoIP codecs rely on a steady packet stream. Excessive jitter causes the jitter buffer to underrun or overflow, resulting in clipped, robotic, or dropped audio that is far more noticeable than raw latency.

Exam trap

CLCOR often tests the misconception that bandwidth alone determines voice quality, leading candidates to pick 'high bandwidth usage' instead of the correct latency/jitter/loss triad.

How to eliminate wrong answers

Option A is wrong because low CPU usage on the router is a healthy condition and does not degrade voice quality; high CPU would be the concern. Option B is wrong because static IP addressing has no direct impact on voice quality; DHCP vs static is an addressing choice, not a QoS factor. Option D is wrong because high bandwidth usage alone does not necessarily degrade voice if QoS is properly configured to prioritize RTP; bandwidth is a capacity issue, whereas jitter directly disrupts playout.

Full explanation →
13
MCQhard

You have a requirement to use secure signaling for SIP endpoints. Which protocol and port combination must be used in the SIP Security Profile?

A.SIP TLS 5061
B.HTTP 443
C.SIP TCP 5060
D.SRTP 5061
AnswerA

SIP TLS on port 5061 encrypts signalling between SIP endpoints, satisfying the requirement for secure signalling. TLS protects headers and credentials that plain SIP on 5060 exposes, and 5061 is the registered port for SIP over TLS. The SIP Security Profile in Cisco Unified Communications Manager expects this pairing.

Why this answer

SIP TLS on port 5061 is the standard secure signaling transport for SIP endpoints in Cisco Unified Communications Manager. The SIP Security Profile in CUCM requires selecting TLS as the transport with port 5061 so that SIP signaling is encrypted with certificates between the endpoint and the call agent.

Exam trap

CLCOR often tests the confusion between signaling encryption (SIP TLS/5061) and media encryption (SRTP), causing candidates to pick SRTP as the 'secure signaling' answer.

How to eliminate wrong answers

Option B is wrong because HTTP 443 is used for web/HTTPS management and XML services, not for SIP signaling. Option C is wrong because SIP TCP 5060 is the unencrypted signaling port and provides no confidentiality or integrity for SIP messages. Option D is wrong because SRTP is the media encryption protocol (typically on even-numbered RTP ports), not a signaling protocol, and it does not run on 5061.

Full explanation →
14
MCQeasy

In the context of collaboration design, what is the purpose of the 'Location' in CUCM?

A.Defining physical building layouts
B.Device registration grouping
C.Call Admission Control
D.TFTP file distribution
AnswerC

Location in CUCM defines a logical grouping of devices sharing bandwidth constraints, enabling Call Admission Control to deny or reroute new calls when WAN links are saturated. It protects voice quality across sites, rather than performing dial-plan translation or device registration.

Why this answer

In Cisco Unified Communications Manager (CUCM), a 'Location' is the configuration object used to implement Call Admission Control (CAC). It defines the total bandwidth available between groups of devices, typically representing a WAN link between sites. When a call is placed between devices in different Locations, CUCM checks whether sufficient bandwidth exists in both Locations before allowing the call to proceed.

Exam trap

CLCOR often tests the confusion between Locations (CAC/bandwidth) and Regions (codec selection), so candidates who pick 'device registration grouping' or 'physical layout' are conflating Locations with Device Pools or ERL.

How to eliminate wrong answers

Option A is wrong because physical building layouts are documented via separate constructs (like ERL/ELF in emergency responder location, or simply site documentation) — CUCM Locations do not model floor plans or building geometry. Option B is wrong because device registration grouping is handled by Device Pools, not Locations; Device Pools define region, location, softkey template, and other device-level defaults. Option D is wrong because TFTP file distribution is a function of the TFTP service and device configuration file requests, not Locations.

Full explanation →
15
MCQeasy

An administrator needs to verify that the Webex Hybrid Calendar Service connector is running and healthy. Where in Webex Control Hub should the administrator check the status of the connector?

A.Users > Manage Users
B.Organization Settings > Hybrid
C.Services > Hybrid Services > Calendar
D.Devices > List Devices
AnswerC

In Webex Control Hub, the status of hybrid service connectors is found under Services > Hybrid Services. Selecting Calendar shows the connector status, including whether it is running, stopped, or in error. This is the central place to monitor and manage all hybrid services, including Calendar, Call, and Message.

Why this answer

The correct location to verify the health of the Webex Hybrid Calendar Service connector is Services > Hybrid Services > Calendar in Webex Control Hub. This page provides the connector status, version, and any alerts. Administrators can also restart or troubleshoot connectors from here, making it the central monitoring point.

Exam trap

The trap here is assuming connector status is under Organization Settings or Devices, but it is specifically under Services > Hybrid Services.

Full explanation →
16
MCQhard

You have a cluster with multiple nodes. You want to ensure that phones in the 'New York' office prioritize registration to a 'New York' subscriber. How do you implement this?

A.Create a CM Group with the local subscribers and assign it to the Device Pool.
B.Use a Route Group to point to the subscriber.
C.Assign the phones to a specific Region.
D.Configure the TFTP server address in the phone.
AnswerA

Assigning a CM Group containing the New York subscriber to the device pool makes that subscriber the primary registration target for phones in that pool, with others acting as backups. This satisfies the requirement to prioritise local registration for the New York office.

Why this answer

A Cisco CallManager Group (CM Group) defines an ordered list of subscribers that devices in an associated Device Pool will register to. By creating a CM Group containing the New York subscribers and assigning it to the Device Pool used by New York phones, those phones will prefer the local subscriber for registration, keeping signaling local and reducing WAN dependency. This is the standard CUCM mechanism for call processing redundancy and locality.

Exam trap

CLCOR often tests the confusion between CM Groups (registration priority) and Route Groups (call routing), so candidates pick Route Group thinking it directs phones to a subscriber.

How to eliminate wrong answers

Option B is wrong because a Route Group is used for outbound call routing to gateways/trunks, not for phone registration. Option C is wrong because Regions control codec selection between sites, not registration priority. Option D is wrong because the TFTP server address only tells the phone where to download its config; it does not control which subscriber the phone registers to.

Full explanation →
17
MCQmedium

You are troubleshooting intermittent audio quality. You notice that RTCP packets are arriving with high variance. What does this indicate?

A.Latency
B.Codec Mismatch
C.Packet Loss
D.Jitter
AnswerD

RTCP receiver reports carry interarrival jitter statistics; high variance in packet arrival times relative to the RTP timestamp indicates jitter, the variation in packet spacing. This is distinct from loss or latency, and it explains the intermittent audio degradation being troubleshot.

Why this answer

RTCP (RTP Control Protocol) provides reception quality feedback, including interarrival jitter, which is the statistical variance in packet arrival times. High variance in RTCP-reported jitter indicates network jitter — packets are arriving with uneven spacing, causing audio quality issues like choppy or robotic voice. Jitter is distinct from latency (delay) and packet loss (missing packets).

Exam trap

The trap is confusing jitter with latency or packet loss — candidates see 'high variance' and think 'delay' (latency) or 'missing packets' (loss), but RTCP jitter specifically measures interarrival variance, which is the definition of jitter.

How to eliminate wrong answers

Option A is wrong because latency is the absolute delay in packet delivery, not the variance — RTCP reports jitter as a variance metric, not a delay metric. Option B is wrong because codec mismatch would cause negotiation failures or garbled audio, not high RTCP jitter variance; RTCP jitter is a network-layer metric independent of codec. Option C is wrong because packet loss is reported separately in RTCP (fraction lost, cumulative lost) and would not manifest as high jitter variance — loss and jitter are distinct RTCP metrics.

Full explanation →
18
Multi-Selectmedium

A collaboration engineer is designing QoS for a Cisco Collaboration deployment where IP phones connect to Cisco Catalyst switches and the WAN edge is a Cisco IOS XE router. The engineer wants to ensure that voice bearer traffic is properly marked and prioritized end to end. Which two actions should the engineer take to establish a consistent QoS trust boundary? (Choose two.)

Select 2 answers
A.Configure a priority queue for voice bearer traffic on the WAN egress interface.
B.Configure the access switch port to trust the DSCP or CoS marking from the IP phone.
C.Configure the WAN router to re-mark all traffic entering from the LAN to DSCP 0.
D.Configure the access switch to trust CoS only on the WAN router uplink.
E.Configure the IP phones to mark all traffic, including PC data, as EF.
AnswersA, B

Even with correct markings, voice needs strict priority treatment during congestion to keep delay and jitter low. A low-latency queue on the WAN egress interface, typically matching DSCP EF with a bandwidth guarantee and policer, ensures voice packets are serviced ahead of data so call quality remains acceptable when the link is saturated.

Why this answer

A consistent trust boundary requires the access switch to accept the phone's DSCP or CoS marking so voice keeps its EF value, and the WAN egress interface to provide a strict priority queue for that marked bearer traffic. Together these steps preserve marking from the endpoint and guarantee low latency across the constrained WAN link, which is the foundation of an end-to-end collaboration QoS design.

Exam trap

The trap here is focusing only on marking or only on queuing, when end-to-end voice quality requires both a trusted marking at the access edge and a priority queue at the congestion point.

Full explanation →
19
Multi-Selectmedium

Which TWO of the following are valid types of Media Resources in CUCM?

Select 2 answers
A.Music on Hold Server
B.Conference Bridge
C.Calling Search Space
D.Route Group
E.Translation Pattern
AnswersA, B

Music on Hold Server is a distinct media resource type configured in CUCM, streaming audio to callers placed on hold. It satisfies the stem's requirement for valid Media Resource types, alongside other resources such as conference bridges, transcoders, and media termination points.

Why this answer

In Cisco Unified Communications Manager, Media Resources are the software or hardware components that provide media-processing functions for calls, and the two valid types listed here are the Music on Hold (MOH) Server (option A) and the Conference Bridge (option B). A Music on Hold Server is a media resource because it streams audio to callers placed on hold, and it is configured under Media Resources > Music On Hold Server in CUCM. A Conference Bridge is likewise a media resource because it mixes the audio streams of multiple participants in a conference call, and it appears under Media Resources > Conference Bridge.

The other options are not media resources: a Calling Search Space (C) is a dial-plan construct that defines which partitions a device or line can search for call routing, a Route Group (D) is a list of gateways or trunks used for route-list selection in call routing, and a Translation Pattern (E) is a dial-plan element that manipulates the called-party number before routing. None of C, D, or E perform any media processing, so they do not qualify as Media Resources in CUCM.

Exam trap

CLCOR often tests the distinction between dial-plan objects (CSS, Partition, Translation Pattern, Route Group, Route List) and media-processing objects (MOH, Conference Bridge, Transcoder, MTP, Annunciator), so candidates who see 'Route Group' and 'Translation Pattern' as infrastructure components may incorrectly select them.

Full explanation →
20
MCQmedium

What is the primary function of the Cisco Webex Edge Audio service?

A.To allow calling into Webex meetings via the corporate PSTN network.
B.To allow users to join meetings without an internet connection.
C.To encrypt all meeting audio traffic.
D.To provide high-definition video conferencing.
AnswerA

Webex Edge Audio routes meeting call-ins through the organisation's existing corporate PSTN infrastructure rather than Cisco's cloud telephony. This satisfies the requirement to leverage on-premises voice gateways for dial-in, reducing toll charges and keeping call traffic within the enterprise network.

Why this answer

Webex Edge Audio is a service that lets organizations route Webex meeting audio through their existing corporate PSTN/telephony infrastructure instead of the Webex cloud PSTN. This allows participants to dial into meetings using corporate phone lines, reducing toll costs and leveraging existing carrier contracts. It bridges on-premises voice gateways (CUBE/Expressway) with the Webex cloud for inbound and outbound audio.

Exam trap

CLCOR often tests the misconception that Edge Audio is about encryption or offline access — it is specifically about routing meeting audio through the corporate PSTN via SIP trunking.

How to eliminate wrong answers

Option B is wrong because joining without an internet connection is not the purpose of Edge Audio — participants still need connectivity to the Webex cloud or corporate network; Edge Audio is about PSTN routing, not offline access. Option C is wrong because encrypting meeting audio is handled by Webex's standard media encryption (SRTP/TLS), not by Edge Audio, which is a PSTN routing feature. Option D is wrong because high-definition video conferencing is a general Webex Meetings capability unrelated to Edge Audio, which specifically addresses audio PSTN integration.

Full explanation →
21
MCQmedium

You need to enforce a specific codec for calls traversing a CUBE. Where is the most appropriate place to apply this configuration?

A.Global voice service
B.SIP profile
C.Under the interface
D.Voice class codec applied to dial-peer
AnswerD

A voice class codec defines the codec preference list, and binding it to the dial-peer governs codec negotiation for calls matching that peer. This satisfies the requirement to enforce a specific codec on CUBE traffic without affecting unrelated peers.

Why this answer

Codec enforcement on CUBE is applied via a voice class codec that is referenced by the dial-peer handling the call. The dial-peer is where media capabilities are negotiated for that call leg, so binding the codec class there gives per-destination control. Global voice service, SIP profiles, and interface configs do not enforce codecs.

Exam trap

CLCOR often tests whether candidates place codec enforcement on the dial-peer via voice class codec rather than globally or on the SIP profile — codecs are per-call-leg media decisions.

How to eliminate wrong answers

Option A is wrong because global voice service configures signaling and media parameters (e.g., SIP, media flow-around) but does not select codecs per call. Option B is wrong because SIP profiles manipulate signaling headers/SDP text, not codec negotiation. Option C is wrong because interface configuration handles IP transport, not media codec selection.

Full explanation →
22
MCQmedium

What happens to a user's Webex account if they are removed from the synchronized Active Directory group?

A.The account remains active indefinitely.
B.The account is moved to a guest group.
C.The account becomes a free account.
D.The account is deactivated in Webex.
AnswerD

Directory synchronisation treats the Active Directory group as the source of truth, so removing a user from the synchronised group removes them from the Webex directory sync, and Control Hub deactivates the associated Webex account rather than deleting it outright.

Why this answer

When Webex Directory Synchronization is configured with Active Directory, the sync engine continuously reconciles Webex Control Hub users against the AD source of truth. If a user is removed from the synchronized AD group, the next sync cycle detects that the user is no longer in scope and deactivates the corresponding Webex account. Deactivation preserves the user's data (messages, files, settings) but blocks sign-in and license consumption until the account is either re-synced or manually reactivated.

Exam trap

CLCOR often tests the distinction between deactivation and deletion in Webex directory sync — candidates assume removal from AD either does nothing or fully deletes the account, when the actual behavior is a reversible deactivation that preserves data.

How to eliminate wrong answers

Option A is wrong because Webex directory sync is authoritative — leaving the account active indefinitely would defeat the purpose of AD-driven lifecycle management and would keep consuming licenses for users who no longer exist in the source group. Option B is wrong because guest accounts in Webex are created through a separate invitation workflow (or via specific guest-related sync rules), not as an automatic fallback when a synced user drops out of the AD group. Option C is wrong because Webex does not convert a deactivated enterprise user into a free consumer account; free accounts are a distinct account type tied to a personal sign-up, and enterprise-managed identities cannot silently downgrade to that state.

Full explanation →
23
MCQmedium

Which field in a Route Pattern can be used to discard the first few digits of a dialed number before routing?

A.Calling Search Space
B.Called Party Transform Mask
C.Discard Digits
D.Route Filter
AnswerC

The Discard Digits field applies a pre-dot or post-dot digit-discard pattern, stripping leading digits from the dialled string before Cisco Unified Communications Manager matches the pattern and routes the call. This satisfies the requirement to remove initial digits.

Why this answer

The 'Discard Digits' field in a Cisco Unified Communications Manager Route Pattern is specifically designed to strip leading digits from the dialed number before the call is routed. It uses predefined discard digit instructions (such as PreDot, PreAt, or NoDigits) to remove unwanted prefixes like access codes or area codes. This transformation occurs before the number is presented to the destination, ensuring the correct digits are sent.

Exam trap

CLCOR often tests the confusion between digit manipulation fields (Discard Digits, Called Party Transform Mask) and routing/authorization fields (Calling Search Space, Route Filter). Candidates may incorrectly choose Called Party Transform Mask because it also manipulates digits, but it does not discard them.

How to eliminate wrong answers

Option A is wrong because Calling Search Space is used for call routing authorization and class of service, not for digit manipulation. Option B is wrong because Called Party Transform Mask is used to modify the called party number by adding or replacing digits, not discarding them. Option D is wrong because Route Filter is used to filter route patterns based on dialed digits for route selection, not to discard digits from the dialed number.

Full explanation →
24
MCQhard

You need to ensure that the Webex App can perform 'Click-to-Call' from a browser. What must be installed on the user's machine?

A.The Hybrid Calendar connector.
B.A local PSTN gateway.
C.Cisco Jabber for Windows.
D.Webex browser extension.
AnswerD

Click-to-Call from a browser requires the Webex browser extension, which integrates the Webex App with the browser to parse telephone numbers on web pages and hand them to the client. Without this extension, the browser cannot invoke the Webex App's calling functions.

Why this answer

Click-to-Call from a browser requires the Webex browser extension, which registers a URI handler (such as tel: or webex:) so the browser can hand off the click to the Webex App for call initiation. Without the extension, the browser has no mechanism to invoke the Webex App's dialer from a web page. The extension is available for Chrome, Edge, and Firefox and must be installed and enabled per user profile.

Exam trap

CLCOR often tests whether candidates confuse client-side browser integration (the Webex extension) with server-side services (Hybrid Calendar connector) or alternate clients (Jabber) — the extension is the only component that enables browser click-to-call.

How to eliminate wrong answers

Option A is wrong because the Hybrid Calendar connector is a server-side service on Expressway-C that syncs calendar data with Webex — it has nothing to do with browser click-to-call. Option B is wrong because a local PSTN gateway provides voice trunking to the PSTN and is unrelated to browser integration. Option C is wrong because Cisco Jabber is a separate softphone client; installing Jabber does not enable click-to-call within the Webex App and is not the required component.

Full explanation →
25
MCQmedium

When designing for high availability of Cisco Unified Communications Manager (CUCM) services, which specific configuration ensures that a secondary node takes over call processing if the primary node fails?

A.Configure a CUCM Group in the Device Pool with multiple servers.
B.Set the 'Call Processing Load' value in the System settings.
C.Enable Auto-registration for all phones.
D.Configure the cluster-wide failover parameter in the Service Parameter page.
AnswerA

A CUCM Group lists an ordered set of servers, and assigning it to the Device Pool lets endpoints register with the primary node while retaining the secondary as a fallback. If the primary fails, call processing shifts to the next listed server, delivering the required high availability.

Why this answer

A CUCM Group is an ordered list of up to three CUCM servers (primary, secondary, tertiary) that a device uses for call processing. When you assign a CUCM Group to a Device Pool, every device referencing that pool will attempt registration with the first server in the list and, if it becomes unreachable, will fail over to the next server in the group. This is the standard, supported mechanism for call-processing redundancy in a CUCM cluster.

Exam trap

CLCOR often tests the misconception that failover is controlled by a global service parameter or a system-wide setting, when in fact it is enforced per-device through the CUCM Group assigned in the Device Pool.

How to eliminate wrong answers

Option B is wrong because 'Call Processing Load' is not a real CUCM System setting; there is no such parameter that controls failover behavior. Option C is wrong because Auto-registration is a feature that allows unprovisioned phones to automatically receive a DN from a defined range — it has nothing to do with server redundancy or failover. Option D is wrong because there is no cluster-wide 'failover parameter' on the Service Parameter page; failover is controlled per-device via the CUCM Group assigned through the Device Pool, not by a global service parameter.

Full explanation →
26
MCQmedium

What is the function of 'Cisco Unified Reporting' in a CUCM environment?

A.To generate cluster and system reports.
B.To monitor network bandwidth.
C.To manage real-time call logs.
D.To perform bulk configuration.
AnswerA

Cisco Unified Reporting provides prebuilt cluster-wide and system-level reports, such as cluster overview and service status, without needing external tools. This directly fulfils the requirement to generate cluster and system reports in a CUCM environment.

Why this answer

Cisco Unified Reporting is a CUCM administration interface that generates on-demand reports about the cluster, such as device counts, license usage, dial plan summaries, and system health snapshots. It is designed for reporting and troubleshooting, not real-time monitoring or configuration. It pulls data from the CUCM database and presents it in downloadable formats like PDF or CSV.

Exam trap

CLCOR often tests the distinction between Unified Reporting (static reports), RTMT (real-time monitoring), and BAT (bulk configuration), causing candidates to confuse reporting with monitoring or configuration.

How to eliminate wrong answers

Option B is wrong because Cisco Unified Reporting does not monitor network bandwidth; bandwidth monitoring is handled by tools like Cisco Prime Collaboration or SNMP-based monitoring, not Unified Reporting. Option C is wrong because real-time call logs and active call monitoring are provided by Cisco Unified Real-Time Monitoring Tool (RTMT) and CDR Analysis and Reporting (CAR), not Unified Reporting. Option D is wrong because bulk configuration is performed using the Bulk Administration Tool (BAT), not Unified Reporting, which is read-only reporting.

Full explanation →
27
MCQhard

A Cisco Unified CM cluster is configured with a SIP trunk to a Cisco Unified Border Element (CUBE). Calls from the PSTN arrive at CUBE and are sent to CUCM. The ITSP requires that the From header contain the original calling number in E.164 format, but CUCM is sending it in a 10-digit national format. You need to ensure the From header is rewritten to include the +1 prefix. Which configuration on CUCM will accomplish this?

A.Modify the SIP Profile assigned to the trunk to enable 'Use Fully Qualified E.164 Number'.
B.Configure an Outbound Calling Party Transformation Pattern on the SIP trunk that prepends +1.
C.Create a Translation Pattern that matches the 10-digit number and adds the +1 prefix, then apply it to the trunk's inbound CSS.
D.Create a SIP Normalization Script that adds the +1 to the From header on the SIP trunk.
AnswerB

Outbound Calling Party Transformation Patterns are applied to the calling number before the SIP message is sent out the trunk. By creating a transformation pattern that matches the 10-digit number and prepends +1, CUCM will rewrite the calling number in the From header to E.164 format. This is the native, supported method for manipulating the calling party number on a SIP trunk.

Why this answer

The calling party number in the From header of an outbound SIP message can be manipulated using an Outbound Calling Party Transformation Pattern on the SIP trunk. This pattern matches the current calling number and can prepend +1, ensuring the ITSP receives the number in E.164 format. This approach is native to CUCM and does not require scripting.

Exam trap

The trap here is confusing translation patterns with transformation patterns; translation patterns affect dialed digits during call routing, while transformation patterns specifically manipulate calling and called numbers on trunk interfaces.

Full explanation →
28
MCQeasy

When configuring a Device Pool in CUCM, which parameter is responsible for defining the set of codecs used between devices in different regions?

A.Calling Search Space
B.Date/Time Group
C.Location
D.Region
AnswerD

The Region configuration defines which codec set is used for calls between devices in different regions, controlling the maximum audio codec bandwidth permitted across those regions. Device Pools reference a Region, so this parameter governs inter-region codec selection rather than the pool's own codec preference.

Why this answer

In Cisco Unified Communications Manager (CUCM), a Region defines the set of codecs used for calls between devices within that region and between different regions. Regions are used to manage bandwidth by specifying which codecs are allowed for intra-region and inter-region communication. The Region parameter directly controls the codec selection for calls between devices in different regions.

Exam trap

CLCOR often tests the distinction between Regions and Locations; candidates may confuse the two, thinking that Location defines codecs, when in fact Location defines bandwidth and Region defines codecs.

How to eliminate wrong answers

Option A is wrong because a Calling Search Space (CSS) is used for call routing and class of service, not codec selection. Option B is wrong because a Date/Time Group defines time zones and formats for devices, not codecs. Option C is wrong because a Location defines the bandwidth available for calls to and from a particular location, but it does not define the codecs; it works in conjunction with Regions to enforce bandwidth limits.

Full explanation →
29
MCQhard

When troubleshooting CUBE SIP normalization, which tool allows you to view the normalized SIP messages in real-time?

A.show voice class sip-profiles
B.debug ccsip messages
C.debug voip ccapi inout
D.debug sip-ua
E.show sip-profile
AnswerB

CUBE applies SIP normalization profiles on inbound and outbound dial peers, rewriting headers and SDP. The debug ccsip messages command traces these SIP messages as they are parsed and normalised in real time, exposing the before-and-after header transformations needed to verify the normalisation rules are firing correctly.

Why this answer

The command `debug ccsip messages` (B) enables real-time debugging of SIP messages on Cisco Unified Border Element (CUBE). It displays the actual SIP messages sent and received, including the normalized messages after SIP normalization profiles are applied, allowing you to verify the normalization logic.

Exam trap

CLCOR often tests the confusion between configuration verification commands (show) and real-time debugging commands (debug), and between different debug commands for SIP versus call control.

How to eliminate wrong answers

Option A is wrong because `show voice class sip-profiles` displays the configuration of SIP profiles but does not show real-time normalized messages. Option C is wrong because `debug voip ccapi inout` traces call control API events, not SIP message normalization. Option D is wrong because `debug sip-ua` debugs SIP user agent events on CUCM, not CUBE normalization.

Option E is wrong because `show sip-profile` is not a valid command; the correct command is `show voice class sip-profiles`.

Full explanation →
30
MCQmedium

When a SIP phone initiates a call, which message does the CUCM send to acknowledge the initial INVITE?

A.401 Unauthorized
B.100 Trying
C.180 Ringing
D.200 OK
AnswerB

CUCM replies with 100 Trying to the initial INVITE, a provisional response confirming the request was received and is being processed, which stops the phone retransmitting the INVITE. It precedes any 180 Ringing or 200 OK, so it is the acknowledgement the SIP phone expects first.

Why this answer

When a SIP phone sends an INVITE to CUCM, CUCM immediately returns a 100 Trying provisional response to stop the phone from retransmitting the INVITE while CUCM processes the request. This is defined in RFC 3261 as a provisional (1xx) response indicating the request has been received and is being processed. It is not a final response and does not indicate the call is ringing or answered.

Exam trap

CLCOR often tests SIP response-code sequencing, and candidates confuse the immediate 100 Trying acknowledgement with later provisional responses like 180 Ringing or final responses like 200 OK.

How to eliminate wrong answers

Option A is wrong because 401 Unauthorized is a final 4xx response used for SIP digest authentication challenges, not the immediate acknowledgement of an INVITE. Option C is wrong because 180 Ringing is a provisional response sent later, once the called party's device is actually alerting — it is not the first acknowledgement. Option D is wrong because 200 OK is the final success response sent when the call is answered, which occurs after ringing and is not an acknowledgement of the initial INVITE.

Full explanation →
31
MCQhard

A collaboration engineer is implementing a dial plan that requires all calls to emergency services (911) to be routed over a specific SIP trunk to a dedicated PSAP gateway, regardless of the calling device's location. Which CUCM configuration element should be used to ensure this routing?

A.A Route Pattern for 911 in a partition that is included in the calling search space of all devices, with the route pattern pointing to a route list containing only the PSAP gateway SIP trunk
B.A Route Pattern for 911 with the 'Route This Pattern' option and a 'Calling Search Space' that includes the PSAP partition
C.A dedicated Route Pattern for 911 that points to a Route List containing only the PSAP gateway SIP trunk
D.A Transformation Pattern that modifies the dialed number to a unique prefix and routes based on that prefix
AnswerA

This approach ensures that all devices can dial 911 because the partition containing the 911 route pattern is in every device's calling search space. The route pattern then directs the call to a route list that contains only the PSAP gateway SIP trunk, guaranteeing that emergency calls use that specific trunk. This is the standard best practice for emergency call routing in CUCM.

Why this answer

To ensure that all emergency calls route over a specific SIP trunk, the 911 route pattern must be placed in a partition that is included in the calling search space of every device. The route pattern should point to a route list that contains only the PSAP gateway SIP trunk. This guarantees that any device dialing 911 will match the pattern and be routed to the dedicated trunk.

The other options either do not guarantee universal access or misuse configuration elements.

Exam trap

The trap here is placing the calling search space on the route pattern instead of on the devices; the route pattern's CSS does not control which devices can use it.

Full explanation →
32
MCQmedium

A collaboration engineer is preparing a Cisco Unified Communications Manager cluster for a new site that requires the Cisco 8845 IP Phone to operate in a secure environment. The security team mandates that all signaling and media between the phone and CUCM be encrypted, and that the phone authenticate to CUCM using a locally significant certificate (LSC). Which action must the engineer take first to ensure the phone can obtain an LSC and register securely?

A.Upload a publicly signed certificate to the CUCM Tomcat trust store and restart the Cisco Tomcat service.
B.Enable 'Mixed Mode' on the CUCM cluster and restart the Cisco CallManager service.
C.Configure a Phone Security Profile with 'Device Security Mode' set to Encrypted and assign it to the phone.
D.Install a Certificate Authority Proxy Function (CAPF) certificate on the phone manually via the phone's web interface.
AnswerB

Enabling Mixed Mode activates the Cisco Certificate Trust List (CTL) and allows the cluster to issue LSCs to phones. Without Mixed Mode, the phone can only use a Manufacturer Installed Certificate (MIC) or no certificate, and secure signaling cannot be established. This is the foundational step before any secure registration can occur.

Why this answer

To enable secure signaling and media with LSCs, the CUCM cluster must first be switched to Mixed Mode. This activates the Cisco CTL and allows the CAPF service to issue LSCs to supported endpoints. Only after Mixed Mode is enabled can a Phone Security Profile with Encrypted mode be effective, and the phone can then register securely using its LSC.

Exam trap

The trap here is assuming that configuring a Phone Security Profile alone enables secure registration, when Mixed Mode must be enabled first to allow LSC issuance.

Full explanation →
33
MCQeasy

Where do you configure the time zone for a specific set of devices, such as those in a specific branch office?

A.Region
B.Enterprise Parameters
C.Location
D.Date/Time Group
AnswerD

A Date/Time Group bundles a time zone, NTP servers and date/time settings, then is assigned to device pools. Devices in the branch office inherit that group, so their clocks reflect local time without per-device configuration.

Why this answer

The time zone for a specific set of devices, such as those in a branch office, is configured using a Date/Time Group in CUCM. A Date/Time Group defines the time zone and date/time format, and it is assigned to devices via device pools. This allows different groups of devices to have different time zones based on their location.

Exam trap

CLCOR often tests the confusion between Date/Time Group and other location-related settings like Region or Location, and candidates may incorrectly select Location for time zone configuration when it is actually used for bandwidth management.

How to eliminate wrong answers

Option A is wrong because a Region in CUCM is used to define codec selection between sites for bandwidth management, not time zone settings. Option B is wrong because Enterprise Parameters are cluster-wide settings that apply to all devices, not a specific set of devices. Option C is wrong because a Location in CUCM is used for call admission control and bandwidth allocation, not for time zone configuration.

Full explanation →
34
MCQeasy

Where in Webex Control Hub do you verify the health status of Hybrid Service connectors?

A.Troubleshooting tab
B.Devices tab
C.Services tab
D.Users tab
AnswerC

The Services tab in Webex Control Hub provides the dedicated health view for Hybrid Service connectors, showing connector status, alerts and connectivity to on-premises resources. This directly satisfies the stem's requirement to verify connector health, since connector monitoring lives under Services rather than general organisation or user settings.

Why this answer

In Webex Control Hub, the health status of Hybrid Service connectors is verified under the Services tab. Navigating to Services > Hybrid Services (or the specific service like Call/Calendar) shows connector status, registration state, and any alerts for each Expressway-C node running the connector.

Exam trap

CLCOR often tests the location of Hybrid Service connector status in Control Hub — candidates pick Troubleshooting because it sounds diagnostic, but connector health lives under Services.

How to eliminate wrong answers

Option A is wrong because the Troubleshooting tab focuses on call-quality diagnostics, meeting issues, and user-level troubleshooting — not connector health. Option B is wrong because the Devices tab lists Webex devices (phones, room systems) and their status, not Hybrid Service connectors. Option D is wrong because the Users tab manages user accounts, licenses, and assignments, not connector health.

Full explanation →
35
MCQeasy

An engineer needs to verify the current active calls on a Cisco IOS XE voice gateway and see details such as call legs, codecs, and durations. Which command should be used?

A.show sip-ua calls
B.show voice call status
C.show call active voice brief
D.show dial-peer voice summary
AnswerC

The 'show call active voice brief' command provides a concise summary of all active voice calls, including call legs, calling and called numbers, codec, and call duration. It is the standard command to verify current active calls on a Cisco IOS XE voice gateway, making it the correct choice for this scenario.

Why this answer

The 'show call active voice brief' command is the correct tool to verify active calls on a Cisco IOS XE voice gateway. It lists each call leg with critical details such as calling/called numbers, codec, and duration, allowing an engineer to quickly assess current call activity. Other commands either focus on configuration or are protocol-specific.

Exam trap

The trap here is selecting a command that shows dial-peer or port status instead of actual active call details, which requires the 'show call active voice brief' command.

Full explanation →
36
MCQhard

You are migrating to a new SIP provider. You have a requirement to use 'Early Offer' for all calls on this trunk. Where is this configured?

A.SIP Trunk Security Profile
B.Device Pool
C.Route List
D.SIP Profile
AnswerD

Early Offer behaviour for SIP signalling is controlled by the SIP Profile, where the header and offer settings are defined and then applied to the trunk. Configuring it there satisfies the requirement that all calls on the new provider trunk use Early Offer.

Why this answer

Early Offer for a SIP trunk is configured in the SIP Profile, which contains parameters such as 'Early Offer support for voice and video calls' and 'MTP Required'. The SIP Profile is assigned to the SIP trunk and controls whether CUCM sends an INVITE with SDP (Early Offer) or without SDP (Delayed Offer). This is the correct location for this setting.

Exam trap

CLCOR often tests the distinction between SIP Profile, SIP Trunk Security Profile, and Device Pool settings, and candidates pick the security profile or device pool when the question is about signaling behavior like Early Offer.

How to eliminate wrong answers

Option A is wrong because the SIP Trunk Security Profile controls security-related settings like authentication, encryption, and digest credentials, not offer/answer behavior. Option B is wrong because the Device Pool groups devices for common settings like region, date/time, and SRST reference, not SIP offer mode. Option C is wrong because the Route List defines the order of route groups for call routing, not SIP signaling behavior.

Full explanation →
37
Multi-Selecthard

A company is deploying Webex Hybrid Data Security (HDS) to enhance privacy for Webex messages and files. Which two components must be deployed on-premises to support this service? (Choose two.)

Select 2 answers
A.Hybrid Data Security management connector
B.Hybrid Data Security node
C.Cisco Unified Communications Manager IM and Presence Service
D.Webex Meetings Server
E.Expressway-C and Expressway-E pair
AnswersA, B

The Hybrid Data Security management connector is an on-premises component that provides management and monitoring capabilities for the HDS nodes. It communicates with the Webex cloud to synchronize configuration and status. It is required in addition to the HDS node to fully deploy the service.

Why this answer

Hybrid Data Security requires two on-premises components: the HDS node, which stores encryption keys and performs cryptographic operations, and the HDS management connector, which handles management and communication with the Webex cloud. Together, they ensure that encryption keys remain on-premises while enabling secure messaging and file sharing in Webex.

Exam trap

The trap here is confusing Hybrid Data Security with other hybrid services that use Expressway or IM&P, leading to selection of components that are not part of HDS.

Full explanation →
38
MCQmedium

A collaboration engineer is deploying a new branch where WAN congestion causes jitter on voice calls. The carrier marks all traffic with DSCP 0 on ingress. The engineer wants the branch Cisco IOS router to reclassify voice bearer traffic into the EF class and apply priority queueing before the traffic enters the MPLS WAN. Which configuration approach accomplishes this goal?

A.Configure RSVP between the branch router and the WAN headend to reserve bandwidth for voice bearer traffic.
B.Configure a class-map matching RTP media ports, then a policy-map that uses the priority command, and apply it outbound on the WAN interface with service-policy output.
C.Configure the WAN interface with mls qos trust dscp and enable auto-qos on the interface.
D.Configure a policy-map with a bandwidth guarantee for voice and apply it inbound on the LAN interface.
AnswerB

This is correct because trust is broken at ingress (DSCP 0), so the router must reclassify. A class-map matching the RTP port range identifies voice bearer, the priority command gives it strict-priority LLQ treatment, and applying the policy-map outbound on the WAN interface shapes and queues the traffic in the correct direction toward the congestion point.

Why this answer

Because the carrier resets all packets to DSCP 0, the branch router cannot rely on trust. The engineer must classify voice bearer by RTP port range, mark or treat it as EF, and give it strict-priority queueing with the priority command in a policy-map applied outbound on the WAN interface. This restores low-latency treatment toward the congested MPLS path.

Exam trap

The trap here is assuming that trusting the carrier's DSCP markings is sufficient when the provider has already zeroed them.

Full explanation →
39
MCQeasy

Which of the following is a key advantage of utilizing Cloud Connected PSTN (CCP)?

A.It allows for legacy analog phone support without an ATA.
B.It provides free long-distance calling globally.
C.It requires no internet connection.
D.It eliminates the need for on-premises PSTN gateways.
AnswerD

Cloud Connected PSTN delivers PSTN connectivity through the Cisco cloud, so calls route over the internet to the provider rather than through local gateways. This removes the requirement to deploy and maintain on-premises PSTN gateways at each site.

Why this answer

Cloud Connected PSTN (CCP) is a Cisco Webex Calling feature that connects your cloud phone system directly to a service provider's PSTN network in the cloud, bypassing the need for local gateways. This eliminates the hardware, maintenance, and configuration overhead of on-premises PSTN gateways, allowing calls to be routed entirely over the internet. It simplifies deployment and scales easily, which is the key advantage.

Exam trap

CLCOR often tests the misconception that CCP provides free calling or requires no internet, when in fact it still depends on an internet connection and incurs PSTN charges.

How to eliminate wrong answers

Option A is wrong because CCP does not inherently support legacy analog phones without an ATA; analog devices still require an ATA to convert signals to IP, regardless of PSTN connection type. Option B is wrong because CCP does not provide free long-distance calling; PSTN usage charges still apply based on the service provider's rates. Option C is wrong because CCP relies on an internet connection to reach the cloud PSTN service; without internet, calls cannot be made.

Full explanation →
40
MCQeasy

Which CUCM service must be activated in Serviceability to allow for the use of the 'Cisco Extension Mobility' feature?

A.Cisco Extension Mobility
B.Cisco DB Replicator
C.Cisco TFTP
D.Cisco CallManager
AnswerA

Activating the Cisco Extension Mobility service in Cisco Unified Serviceability enables the EM application on the node, allowing users to log in to shared phones and download their device profile. This directly satisfies the stem's requirement, since the feature cannot function until this specific service is running.

Why this answer

The 'Cisco Extension Mobility' service is a dedicated feature service in Cisco Unified Serviceability that must be activated on the Cisco Unified Communications Manager (CUCM) cluster. Activating this service enables the Extension Mobility feature, allowing users to log in to any phone and have their personal directory number, speed dials, and services applied. Without this service running, the Extension Mobility login and logout functions will not work, even if the feature is configured in CUCM Administration.

Exam trap

CLCOR often tests the distinction between core CUCM services (like Cisco CallManager and Cisco TFTP) and feature-specific services (like Cisco Extension Mobility), causing candidates to overlook that Extension Mobility requires its own dedicated service activation.

How to eliminate wrong answers

Option B is wrong because Cisco DB Replicator is responsible for replicating the database across cluster nodes and is not specific to Extension Mobility; it is a core service that should always be running. Option C is wrong because Cisco TFTP provides configuration files to phones and firmware updates, but it does not enable the Extension Mobility feature itself. Option D is wrong because Cisco CallManager is the core call processing service that handles call routing and signaling; while it is essential for overall telephony, it does not specifically enable Extension Mobility.

Full explanation →
41
MCQmedium

When using Webex Calling with the Local Gateway, which protocol is used for the SIP trunk between the CUBE and the Webex Calling cloud?

A.H.323
B.SCCP
C.MGCP
D.SIP
AnswerD

SIP is the signalling protocol carried over the trunk between the CUBE and Webex Calling cloud, satisfying the stem's requirement for the Local Gateway trunk protocol. Webex Calling registers the CUBE as a local gateway using SIP, not a proprietary alternative, so the trunk transports SIP messages directly.

Why this answer

The SIP trunk between the CUBE (Cisco Unified Border Element) and the Webex Calling cloud uses SIP (Session Initiation Protocol) for signaling. Webex Calling's Local Gateway architecture relies on SIP over TLS to establish and manage calls between on-premises CUBE and the Webex cloud. This is the standard protocol for Webex Calling PSTN connectivity.

Exam trap

CLCOR often tests protocol confusion — candidates pick H.323 or MGCP because they associate Cisco voice with those protocols, but Webex Calling cloud trunking is strictly SIP.

How to eliminate wrong answers

Option A is wrong because H.323 is a legacy ITU protocol used in older Cisco voice deployments; Webex Calling Local Gateway does not use H.323 for its cloud trunk. Option B is wrong because SCCP (Skinny Client Control Protocol) is a Cisco proprietary protocol for phone-to-call-manager signaling, not for cloud SIP trunking. Option C is wrong because MGCP is a gateway control protocol used between a call agent and media gateways, not for the CUBE-to-Webex cloud trunk.

Full explanation →
42
MCQmedium

You need to ensure that the Webex Hybrid Calendar service can access an Exchange 2019 environment. Which authentication method is recommended by Cisco for this deployment?

A.Digest Authentication
B.OAuth 2.0
C.Basic Authentication
D.Kerberos Constrained Delegation
AnswerB

OAuth 2.0 satisfies Cisco's requirement for Webex Hybrid Calendar against Exchange 2019, since on-premises deployments must use the Modern Authentication path rather than basic authentication, which Microsoft has deprecated. OAuth 2.0 delegates access via tokens issued by Microsoft Entra ID, enabling the calendar connector to read and write mailboxes without storing user credentials.

Why this answer

Cisco recommends OAuth 2.0 for Hybrid Calendar with Exchange 2019 because Microsoft has deprecated Basic Authentication for EWS and modern security baselines require token-based auth. OAuth 2.0 allows the Calendar Connector to authenticate via Azure AD app registration without storing user credentials. This is the supported and future-proof method for Exchange 2019 deployments.

Exam trap

CLCOR often tests authentication method recommendations, and candidates pick Basic or Kerberos because they are familiar with legacy Exchange setups, missing that Cisco explicitly recommends OAuth 2.0 for Exchange 2019.

How to eliminate wrong answers

Option A is wrong because Digest Authentication is a legacy HTTP auth mechanism not supported by Cisco for Hybrid Calendar and is not used with EWS in modern Exchange. Option C is wrong because Basic Authentication is deprecated by Microsoft for Exchange Online and strongly discouraged for Exchange 2019; Cisco no longer recommends it. Option D is wrong because Kerberos Constrained Delegation is an on-premises Windows auth model, not the method Cisco recommends for the Calendar Connector's cloud-mediated access to Exchange.

Full explanation →
43
MCQmedium

What is the primary function of a 'Location' object in CUCM?

A.To define the phone's time zone.
B.To define the phone's CSS.
C.To define the phone's codec.
D.To perform Call Admission Control (CAC).
AnswerD

A Location object groups devices with shared bandwidth characteristics, such as a branch site. CUCM uses these locations to perform Call Admission Control, limiting the number of concurrent calls across a WAN link so voice quality is preserved.

Why this answer

The primary function of a Location object in CUCM is to perform Call Admission Control (CAC), which limits the number of simultaneous calls (audio/video bandwidth) between locations to prevent oversubscription of WAN links. Locations define the available bandwidth and track usage for calls between them. Other settings like time zone, CSS, and codec are handled by other objects.

Exam trap

CLCOR often tests the confusion between Location (CAC/bandwidth) and Region (codec selection) — candidates mix up which object controls bandwidth versus codec.

How to eliminate wrong answers

Option A is wrong because time zone is defined by the Date/Time Group assigned via the Device Pool, not by Location. Option B is wrong because the Calling Search Space is assigned directly to the device or line, not through Location. Option C is wrong because codec selection is determined by Regions (and the device's region setting in the Device Pool), not by Location.

Full explanation →
44
MCQmedium

A company is deploying a centralized Cisco Unified Communications Manager (CUCM) cluster with a single publisher and two subscribers. The publisher is located in the data center, and the subscribers are in the same data center. Branch offices connect via a WAN. The design team wants to ensure that IP phones at the branches can register and make calls even if the WAN link fails. Which CUCM feature should be implemented to achieve this?

A.Cisco Expressway-C and Expressway-E for Mobile and Remote Access (MRA)
B.Cisco Unified Border Element (CUBE) at the branch
C.Cisco Unified Communications Manager Express (CME) on the branch router
D.Cisco Unified Survivable Remote Site Telephony (SRST) on the branch router
AnswerD

SRST runs on the branch router and provides call processing fallback when the WAN fails and IP phones lose registration to the centralized CUCM. It allows the phones to register to the router and make basic calls, including PSTN access, ensuring survivability. This directly addresses the requirement for branch offices to maintain service during WAN outages.

Why this answer

SRST is the correct solution because it enables branch routers to provide call processing when the centralized CUCM becomes unreachable due to WAN failure. It allows IP phones to fall back to the router for basic calling, including PSTN access. Other options either provide different functions (CME, CUBE) or are for remote access (Expressway) and do not address WAN survivability.

Exam trap

The trap here is confusing branch survivability with remote access solutions, such as Expressway MRA, which require WAN connectivity and do not provide local call processing during outages.

Full explanation →
45
MCQmedium

Which configuration in Control Hub allows you to define the emergency address for a user in a Webex Calling environment?

A.User Profile Settings.
B.Device Settings.
C.Organization Settings.
D.Location Settings.
AnswerD

Location Settings in Control Hub define the civic address and emergency (E911) details assigned to users and numbers. Configuring the location satisfies the requirement by binding each Webex Calling user to a validated emergency address.

Why this answer

In Webex Calling, emergency addresses are defined at the Location level in Control Hub. Each location has an associated emergency address that is used for emergency calls (e.g., 911). User profiles, device settings, and organization settings do not directly define emergency addresses; they inherit from the location.

The correct answer is D.

Exam trap

CLCOR often tests the hierarchy of settings in Control Hub, and candidates may incorrectly assume that emergency addresses are set at the user or device level. The trap is forgetting that emergency addresses are tied to locations, not individual users or devices.

How to eliminate wrong answers

Option A is wrong because user profile settings do not contain emergency address configuration; emergency addresses are assigned based on the user's location. Option B is wrong because device settings do not define emergency addresses; devices inherit the location's emergency address. Option C is wrong because organization settings are too broad and do not specify emergency addresses per user or location.

Full explanation →
46
MCQmedium

You have a requirement to play a custom announcement when a call is placed on hold. Where do you configure this?

A.Device Pool configuration.
B.Music on Hold Audio Source.
C.Route Group configuration.
D.SIP Profile.
AnswerB

Uploading the custom audio file as a Music on Hold Audio Source lets Cisco Unified Communications Manager stream it to the held party, satisfying the requirement to play a bespoke announcement rather than the default tone.

Why this answer

Music on Hold (MOH) audio sources are configured under the Music On Hold Audio Source configuration in CUCM, where you define the audio file (e.g., a custom .wav or .au) and its parameters. This audio source is then referenced by a Music On Hold Server and assigned to device pools or other entities. To play a custom announcement on hold, you create a custom audio source and assign it appropriately.

Exam trap

CLCOR often tests the MOH hierarchy, and candidates mistakenly choose Device Pool or SIP Profile, not realizing the actual audio file is defined in the Music On Hold Audio Source.

How to eliminate wrong answers

Option A is wrong because the Device Pool configuration does not define audio files; it references a Music On Hold Server, which in turn references audio sources, but the custom announcement itself is defined in the MOH Audio Source. Option C is wrong because a Route Group is a list of gateways for call routing and has nothing to do with hold audio. Option D is wrong because a SIP Profile contains SIP signaling parameters (timers, headers) and does not define hold music or announcements.

Full explanation →
47
MCQeasy

Which file does an IP phone download first to obtain its configuration from the TFTP server?

A.Dialplan.xml
B.SEP[MAC].cnf.xml
C.Ringlist.xml
D.CTLFile.tlv
AnswerB

The phone first retrieves its device-specific SEP[MAC].cnf.xml, which contains its directory number, firmware load and cluster settings. This file satisfies the stem's requirement for the initial configuration download, since the default XML file is only fetched later as a fallback when no MAC-specific file exists.

Why this answer

An IP phone booting on a CUCM network first obtains its network parameters via DHCP, then contacts the TFTP server and downloads its device-specific configuration file named SEP[MAC].cnf.xml, where [MAC] is the phone's MAC address in uppercase hex. This file contains the phone's DN, cluster TFTP addresses, and other settings. It is the primary per-device configuration file and is downloaded before any shared files like dialplan.xml or ringlist.xml.

Exam trap

CLCOR often tests the boot order, and candidates mistakenly choose shared files like dialplan.xml or CTLFile.tlv as the first download, when the device-specific SEP[MAC].cnf.xml is always first.

How to eliminate wrong answers

Option A is wrong because Dialplan.xml is a shared file containing dial plan digit analysis rules; it is downloaded after the device-specific SEP file and is not the first configuration file. Option C is wrong because Ringlist.xml is a shared file listing available ring tones, downloaded later and not device-specific. Option D is wrong because CTLFile.tlv is the Certificate Trust List file used for secure signaling (ITL/CTL), downloaded as part of the secure boot process but not the first configuration file the phone retrieves.

Full explanation →
48
MCQeasy

A company is deploying Cisco Unified Communications Manager in a centralized call processing model with remote branches connected via a WAN. The network team wants to ensure that if the WAN link fails, IP phones at the branch can still make calls to the PSTN. Which solution should be implemented at the branch?

A.Cisco Expressway-C and Expressway-E for Mobile and Remote Access (MRA)
B.A second CUCM subscriber at the branch office
C.Cisco Unified Survivable Remote Site Telephony (SRST) on the branch router
D.Cisco Unified Communications Manager Express (CME) on the branch router
AnswerC

SRST provides call control redundancy for remote branches. When the WAN link to the central CUCM fails, the branch router with SRST takes over call processing, allowing IP phones to register with it and make calls to the PSTN via local gateways. This ensures business continuity for voice services.

Why this answer

SRST is the designed solution for branch office survivability in a centralized CUCM deployment. When the WAN fails, the branch router with SRST assumes call control, enabling IP phones to register and make PSTN calls via local gateways. This maintains voice services during outages without requiring a full CUCM cluster at the branch.

Exam trap

The trap here is confusing SRST with CME; SRST is a fallback mechanism for CUCM, while CME is a standalone call agent that does not automatically provide redundancy.

Full explanation →
49
MCQhard

In a multi-site deployment, you have a requirement to transcode between G.711 and G.729. If the transcoder resources are exhausted, what is the expected outcome for a new call requiring this conversion?

A.The call will use G.711 regardless
B.The call will be rejected
C.The call will proceed with no audio
D.The call will fall back to an MTP
AnswerB

Exhausted transcoder resources leave no Media Termination Point capable of converting G.711 to G.729, so Cisco Unified Communications Manager cannot allocate the required resource for the new call and rejects it with a resource-unavailable cause code. The stem's transcoding constraint is therefore unsatisfiable, making rejection the expected outcome.

Why this answer

When transcoder resources are exhausted, Cisco Unified Communications Manager (CUCM) cannot allocate the necessary DSP resources to convert between G.711 and G.729 codecs. Without a transcoder, the two endpoints cannot negotiate a common codec, so the call is rejected with a cause code indicating resource unavailability (typically 'No Bandwidth' or 'Unavailable').

Exam trap

CLCOR often tests whether candidates confuse transcoders (codec conversion) with MTPs (DTMF/media termination) — they are distinct DSP resources with different functions, and MTPs cannot perform codec transcoding.

How to eliminate wrong answers

Option A is wrong because the call cannot simply 'use G.711' — the G.729-only endpoint does not support G.711, so no common codec exists without transcoding; the call cannot proceed. Option C is wrong because proceeding with no audio would only happen if the codec mismatch were ignored, but CUCM rejects the call before it is established rather than allowing a silent failure. Option D is wrong because an MTP (Media Termination Point) handles DTMF conversion, media flow-through, and supplementary services — it does not perform codec transcoding between G.711 and G.729; that requires a hardware transcoder (DSP) resource.

Full explanation →
50
MCQmedium

Which design principle must be followed when implementing a centralized call processing deployment with remote branch offices connected via a WAN?

A.Enable SRST on the local branch gateway to handle call processing during a WAN outage.
B.Deploy a dedicated CUCM node in every branch office.
C.Configure H.323 gateways at each site to communicate directly with all remote phones.
D.Use RSVP to guarantee bandwidth across the WAN for signaling.
AnswerA

SRST lets the branch gateway register local phones and provide call processing if the WAN to the central cluster fails. This satisfies the resilience constraint of a centralised deployment, where remote sites lose call control when connectivity to the central Cisco Unified Communications Manager cluster drops.

Why this answer

In a centralized call processing deployment, branch offices rely on the central CUCM for call control. If the WAN link fails, branch phones lose registration and cannot make or receive calls. SRST on the local branch gateway provides fallback call processing, allowing basic PSTN calling during WAN outages.

This is a fundamental design requirement for centralized deployments with remote sites.

Exam trap

CLCOR often tests the misconception that QoS or additional CUCM nodes can replace SRST for branch survivability, but SRST is the specific required design element for WAN outage resilience.

How to eliminate wrong answers

Option B is wrong because deploying a dedicated CUCM node in every branch office defeats the purpose of centralized call processing and increases cost and complexity. Option C is wrong because H.323 gateways do not provide call processing for IP phones; they only provide PSTN gateway functionality. Option D is wrong because RSVP is a QoS mechanism for bandwidth reservation, not a call processing redundancy solution; it does not address WAN outage survivability.

Full explanation →
51
MCQhard

A Cisco Unified CM administrator is troubleshooting a call that fails with a 'No Route to Destination' cause code. The called number is 2001, which is an internal extension on a remote cluster. The local cluster has a SIP trunk to the remote cluster and a route pattern of 2XXX pointing to that trunk. The SIP trunk is up, but the call fails. What is the most likely cause?

A.The remote cluster is not configured with a route pattern for the local cluster, so the call is rejected.
B.The SIP trunk is not configured with a calling search space that includes the partition of the route pattern.
C.The route pattern is not assigned to a partition that is included in the calling search space of the dialing device.
D.The SIP trunk is not registered to the remote cluster, so the call cannot be routed.
AnswerC

For a call to match a route pattern, the pattern must be in a partition that is included in the calling search space of the originating device. If the partition is not in the CSS, the pattern is not considered during digit analysis, resulting in 'No Route to Destination'. This is a common oversight in dial plan configuration.

Why this answer

The most likely cause is that the route pattern 2XXX is in a partition that is not included in the calling search space of the dialing device. Without the partition in the CSS, Unified CM cannot match the pattern, resulting in 'No Route to Destination'. Verifying the device's CSS and partition assignments resolves this issue.

Exam trap

The trap here is focusing on the SIP trunk's configuration or remote cluster settings, when the failure is actually due to the local device's calling search space not including the route pattern's partition.

Full explanation →
52
MCQeasy

Where do you define the default dial plan for a new CUCM cluster?

A.System > Dial Plan Configuration
B.Service Parameters
C.You create partitions, CSS, and route patterns manually.
D.Enterprise Parameters
AnswerC

Cisco Unified Communications Manager ships no preconfigured dial plan; administrators must manually create partitions, calling search spaces and route patterns to define dialling behaviour. This manual construction is precisely how the cluster's default dial plan is established during initial configuration.

Why this answer

In Cisco Unified Communications Manager (CUCM), there is no single 'default dial plan' configuration page. A dial plan is built from three core components: partitions (which group route patterns), calling search spaces (CSS) (which define the order of partitions a device can search), and route patterns (which match dialed digits and determine the next hop). These must be manually created and configured by the administrator.

CUCM does not ship with a pre-built default dial plan, so option C correctly describes the required manual process.

Exam trap

CLCOR often tests the misconception that CUCM includes a pre-configured default dial plan or that dial plan settings are found under a single menu like 'System > Dial Plan Configuration', when in fact all dial plan elements must be manually created under 'Call Routing'.

How to eliminate wrong answers

Option A is wrong because 'System > Dial Plan Configuration' does not exist in CUCM; the actual menu path for dial plan components is 'Call Routing', not a single 'Dial Plan Configuration' page. Option B is wrong because Service Parameters control cluster-wide service settings (e.g., timers, thresholds, trace levels) for individual services, not dial plan definitions. Option D is wrong because Enterprise Parameters define cluster-wide settings such as auto-registration ranges, phone URL authentication, and corporate directory URLs, but they do not define dial plans.

Full explanation →
53
MCQmedium

Which service must be active for CTI applications (like Cisco Finesse) to control phones?

A.Cisco CallManager
B.Cisco CTIManager
C.Cisco TFTP
D.Cisco Database Layer Monitor
AnswerB

Cisco CTIManager provides the computer telephony integration interface that Finesse and other CTI applications use to monitor and control devices. Without this service active, CTI clients cannot issue phone control commands, so it must be running for Finesse to operate.

Why this answer

Cisco CTIManager is the service that provides the Computer Telephony Integration (CTI) interface, allowing applications like Cisco Finesse, Jabber, and third-party CTI apps to monitor and control phones and lines. Without CTIManager active, CTI applications cannot control devices.

Exam trap

CLCOR often tests which service enables CTI control — candidates pick Cisco CallManager because it handles calls, but CTIManager is the specific CTI interface service.

How to eliminate wrong answers

Option A is wrong because Cisco CallManager handles core call processing and signaling, not the CTI interface used by applications. Option C is wrong because Cisco TFTP provides configuration files and firmware to phones, not CTI control. Option D is wrong because Cisco Database Layer Monitor monitors the CUCM database replication and health, not CTI.

Full explanation →
54
MCQeasy

A network engineer is troubleshooting a Cisco IP Phone 7841 that fails to register with CUCM. The phone displays 'Configuring IP' and then 'Registering'. The engineer verifies that the phone has obtained an IP address and can ping the TFTP server. Which protocol does the phone use to download its configuration file from the TFTP server?

A.FTP
B.HTTPS
C.HTTP
D.TFTP
AnswerD

Cisco IP Phones use TFTP to download their configuration files from the CUCM TFTP server. After obtaining an IP address and discovering the TFTP server via DHCP Option 150, the phone requests its configuration file (e.g., SEP<MAC>.cnf.xml) using TFTP. This file contains the CUCM server addresses and other settings needed for registration. Without successful TFTP download, the phone cannot proceed to register.

Why this answer

Cisco IP Phones use TFTP to download their configuration files from the CUCM TFTP server. This process occurs after the phone obtains an IP address and discovers the TFTP server via DHCP Option 150. The configuration file contains the necessary information for the phone to register with CUCM, including the CUCM server list.

TFTP is the standard protocol for this step.

Exam trap

The trap here is confusing TFTP with HTTP or HTTPS, which are used for other phone functions like web access or firmware updates, but not for initial configuration download.

Full explanation →
55
MCQeasy

A user dials a number that matches a route pattern configured with the 'Urgent Priority' check box selected. What is the effect of this setting on the dialing behavior?

A.The call is placed immediately without waiting for the interdigit timeout.
B.The call is blocked if the destination is busy.
C.The call is queued until a gateway becomes available.
D.The route pattern is placed at the top of the dial plan for all users.
AnswerA

When Urgent Priority is enabled on a route pattern, CUCM routes the call as soon as the dialed digits match the pattern, bypassing the interdigit timeout. This is useful for patterns that overlap with others, ensuring immediate routing. In this scenario, the user's dialed number matches the urgent pattern, so the call proceeds without delay, which is the intended behavior of this setting.

Why this answer

The Urgent Priority setting on a route pattern instructs CUCM to route the call immediately when the dialed digits match the pattern, without waiting for the interdigit timeout. This is particularly useful when patterns overlap, such as a pattern for 911 and another for 9.1.1, ensuring emergency calls are placed without delay. It does not affect queuing, pattern order, or busy handling.

Exam trap

The trap here is thinking that Urgent Priority changes the order of route patterns or blocks calls, when it actually only affects the timing of digit analysis after a match.

Full explanation →
56
MCQhard

When designing a secure collaboration network, what is the best practice for handling certificates in a CUCM and Expressway cluster?

A.Use self-signed certificates on all nodes.
B.Disable certificate verification on all endpoints.
C.Use the Publisher as the CA for the entire network.
D.Use a trusted third-party CA for all servers.
AnswerD

A trusted third-party CA lets every CUCM and Expressway node present certificates that all peers already trust, eliminating self-signed trust bundles between clusters. This satisfies the secure collaboration requirement by enabling TLS verification across the cluster without manual certificate exchange.

Why this answer

The best practice is to use a trusted third-party Certificate Authority (CA) for all servers in a CUCM and Expressway cluster. Public CA-signed certificates are trusted by default on endpoints, browsers, and other servers, eliminating the need to distribute and manually trust self-signed or internal CA certificates. This is especially critical for Expressway-E, which faces external endpoints that will not trust an internal CA.

Exam trap

CLCOR often tests certificate trust models, and candidates mistakenly believe an internal CA or the Publisher can serve as a universal trust anchor, not realizing external endpoints will reject certificates from untrusted roots.

How to eliminate wrong answers

Option A is wrong because self-signed certificates are not trusted by external endpoints and browsers, causing security warnings and failed TLS negotiations in MRA and B2B scenarios. Option B is wrong because disabling certificate verification defeats the purpose of TLS, exposes the deployment to man-in-the-middle attacks, and is explicitly discouraged by Cisco. Option C is wrong because using the CUCM Publisher as a CA for the entire network creates a single point of trust that external endpoints will not recognize, and it is not a supported or scalable PKI design.

Full explanation →
57
MCQeasy

A collaboration administrator is deploying a new Cisco Unified Communications Manager (CUCM) cluster and needs to ensure that the publisher database is protected against corruption. The administrator plans to use the Disaster Recovery System (DRS) for backups. Which statement accurately describes a DRS backup requirement?

A.A DRS backup can only be restored to a cluster with the exact same IP addresses and hostnames.
B.DRS requires an external SFTP server to store the backup files, and the backup can include the publisher and all subscribers.
C.DRS backups must be scheduled on each node individually, and the backup files are stored locally on each node.
D.DRS backup is only supported for the publisher node and cannot include subscriber nodes.
AnswerB

DRS uses a configured SFTP server as the backup destination. The backup can be scheduled from the publisher and can include the publisher and any subscribers. This centralizes backup management and ensures that all cluster data is protected. The SFTP server must be reachable from all nodes being backed up.

Why this answer

The Disaster Recovery System (DRS) in CUCM uses an external SFTP server to store backups. Backups are scheduled from the publisher and can include the publisher and all subscribers. This ensures that the entire cluster's configuration and data are protected.

The SFTP server must be configured with sufficient storage and network access from all nodes.

Exam trap

The trap here is assuming that DRS backups are stored locally or that only the publisher can be backed up, when in fact an external SFTP server is required and subscribers can be included.

Full explanation →
58
MCQhard

A branch office has a limited bandwidth link. You need to ensure that no more than 5 calls can be made across the WAN at once. What should you configure?

A.Define a Location and set the Audio Bandwidth limit.
B.Modify the Region setting.
C.Create a CSS to block calls.
D.Configure a Route Filter.
AnswerA

Cisco Unified Communications Manager applies the Location's Audio Bandwidth limit to cap concurrent calls across the WAN, since each call consumes a fixed bandwidth increment. Setting the limit to the value matching five calls enforces the constraint on the constrained branch link.

Why this answer

CUCM Locations implement Call Admission Control (CAC) by defining the total audio/video bandwidth available between locations. Setting the Audio Bandwidth limit on a Location (e.g., to a value that supports only 5 G.711 calls, roughly 400 kbps plus overhead) ensures CUCM denies the 6th call across the WAN. Locations are assigned to devices via Device Pools, and the bandwidth between any two locations is calculated based on the configured limits.

Exam trap

CLCOR often tests the confusion between Regions (codec policy) and Locations (bandwidth/CAC), so candidates who pick Region for a call-count limit miss that only Locations enforce admission control.

How to eliminate wrong answers

Option B is wrong because Regions define which codecs are allowed between locations (e.g., G.729 between branch and HQ) but do not enforce a call-count or bandwidth limit. Option C is wrong because a Calling Search Space controls dialing permissions, not bandwidth or call admission — it cannot limit the number of concurrent calls. Option D is wrong because a Route Filter is a dial-plan object that filters route patterns based on dialed digits, time of day, or calling party, not a bandwidth or call-count mechanism.

Full explanation →
59
MCQmedium

When using 'RSVP' for Call Admission Control (CAC), what happens if the available bandwidth on the path is less than the requested amount?

A.The call is placed but with a lower quality codec
B.The call is put on hold until bandwidth is available
C.The call proceeds, but marks the packets with Best Effort
D.The call is blocked
AnswerD

RSVP signals the requested bandwidth along the path; when a router finds insufficient unreserved capacity, it returns a PathErr and the admission decision rejects the flow, so the call setup is blocked rather than rerouted or downgraded.

Why this answer

With RSVP-based Call Admission Control, when a call's RSVP PATH/RESV signaling requests more bandwidth than is available on any link along the path, the router rejects the reservation and the call is blocked (denied). RSVP CAC is a deterministic admission control mechanism: if the requested bandwidth cannot be reserved, the call setup fails rather than proceeding with degraded quality. This ensures that admitted calls receive their guaranteed quality of service.

Exam trap

The trap is assuming RSVP CAC degrades or queues calls when bandwidth is insufficient — candidates may confuse it with QoS marking/dropping mechanisms, but RSVP CAC is binary: it either admits or blocks the call.

How to eliminate wrong answers

Option A is wrong because RSVP CAC does not dynamically downgrade the codec — codec selection happens before or during call setup via other mechanisms, and RSVP either admits or rejects the reservation as requested. Option B is wrong because RSVP does not queue or hold calls for later admission; it immediately rejects the reservation request, and the call fails rather than waiting. Option C is wrong because RSVP CAC does not admit the call with Best Effort marking — that would defeat the purpose of admission control; if bandwidth is insufficient, the reservation is rejected outright, not downgraded to Best Effort.

Full explanation →
60
Multi-Selectmedium

Which TWO of the following media resource types are supported in CUCM?

Select 2 answers
A.IP Phone
B.Dial-Peer
C.Media Termination Point (MTP)
D.Voice Gateway
E.Transcoder
AnswersC, E

CUCM registers MTP as a media resource for inserting or removing media streams, enabling supplementary services such as DTMF relay and RSVP. It satisfies the stem's requirement for a supported media resource type alongside transcoders, conference bridges and annunciators.

Why this answer

In Cisco Unified Communications Manager (CUCM), media resource types are configured under Media Resources and include annunciators, conference bridges, Media Termination Points (MTPs), music on hold (MOH) servers, and transcoders. Option C, Media Termination Point (MTP), is correct because an MTP is a software or hardware media resource registered to CUCM that provides supplementary services such as DTMF conversion, RSVP agent functionality, and H.323-to-SIP interworking by terminating and reoriginating media streams. Option E, Transcoder, is correct because a transcoder is a media resource that converts one codec (for example, G.711) to another (for example, G.729) so endpoints with mismatched codecs can communicate.

Option A, IP Phone, is incorrect because a phone is an endpoint device, not a media resource type configured under Media Resources. Option B, Dial-Peer, is incorrect because dial peers are configuration elements on IOS voice gateways (used for call routing), not a CUCM media resource type. Option D, Voice Gateway, is incorrect because a voice gateway is a network device that connects CUCM to external networks such as PSTN or other PBXs, not a media resource type within CUCM.

Exam trap

CLCOR often tests whether candidates can distinguish CUCM media resources (MTP, transcoder, conference bridge, annunciator) from device types (phones, gateways) and IOS constructs (dial-peers).

Full explanation →
61
MCQhard

A company has deployed Cisco Webex Hybrid Calendar Service with Exchange 2019 on-premises. Users report that scheduled Webex meetings are not appearing on their calendars, and the connector shows a 'Connected' status but no calendar events are being processed. Which action should be taken to resolve this issue?

A.Reinstall the Hybrid Calendar Service connector on the Expressway-C because the certificate may have expired.
B.Verify that the impersonation account has the required permissions to access user mailboxes and that the Exchange Web Services (EWS) URL is correctly configured.
C.Restart the Cisco Webex Hybrid Calendar Service connector service on the Expressway-C.
D.Change the Webex site URL in Control Hub to match the on-premises Exchange domain.
AnswerB

For Hybrid Calendar Service to process calendar events, the impersonation account must have the 'ApplicationImpersonation' role in Exchange and the EWS URL must be reachable and correctly configured in Control Hub. If these are misconfigured, the connector may show connected but fail to read or write calendar events. Checking these is the first step.

Why this answer

Hybrid Calendar Service relies on an impersonation account with ApplicationImpersonation rights and a correctly configured EWS URL. Even if the connector shows 'Connected', missing permissions or an incorrect EWS URL can prevent calendar processing. Verifying these settings ensures the connector can access mailboxes and process meeting invitations.

Exam trap

The trap here is assuming that a 'Connected' status means the service is fully functional, overlooking that underlying permissions or URL misconfigurations can cause silent failures.

Full explanation →
62
MCQeasy

A network administrator is configuring QoS on a Cisco IOS router for a collaboration deployment. The administrator needs to ensure that voice control traffic, such as H.225 and H.245 signaling, is marked with the appropriate DSCP value. According to Cisco best practices, which DSCP value should be used for call signaling traffic?

A.CS3 (24)
B.AF31 (26)
C.EF (46)
D.AF41 (34)
AnswerA

CS3 (Class Selector 3) is the recommended DSCP for call signaling traffic in Cisco collaboration networks. It provides a higher priority than best effort but lower than voice bearer, which is appropriate for signaling. This marking ensures that call setup and teardown messages are prioritized over data traffic but do not compete with voice media. Cisco documentation consistently recommends CS3 for call signaling.

Why this answer

The correct DSCP for call signaling is CS3 (24). This marking ensures that signaling traffic is prioritized above data but below voice bearer, which is essential for maintaining call quality. Voice bearer uses EF (46), while signaling uses CS3.

This separation allows the network to treat real-time voice media with the highest priority and signaling with the next highest, ensuring that call setup and control messages are not delayed during congestion.

Exam trap

The trap here is confusing voice bearer marking (EF) with signaling marking (CS3), or using outdated values like AF31.

Full explanation →
63
MCQmedium

A Cisco Unified Communications Manager cluster is using a SIP trunk to an ITSP. The ITSP sends calls to the enterprise with a +1 prefix, but the enterprise dial plan uses 10-digit numbers without the country code. You need to strip the +1 from the incoming calling and called numbers before they are presented to the dial plan. Where should you configure this translation?

A.Create a Translation Pattern in the Calling Search Space applied to the SIP trunk's incoming calls.
B.Modify the Route Group's distribution algorithm to remove the prefix during digit analysis.
C.Configure Called Party Transformation and Calling Party Transformation patterns on the SIP trunk and assign them to the trunk's inbound settings.
D.Create a SIP Normalization Script that rewrites the SIP INVITE headers on the trunk.
AnswerC

Called Party Transformation and Calling Party Transformation patterns are designed to manipulate incoming and outgoing numbers on a SIP trunk. By creating patterns that strip the leading +1 and applying them to the trunk's inbound transformation settings, the cluster modifies the numbers before they hit the dial plan. This is the native, granular method for per-trunk digit manipulation in CUCM.

Why this answer

Incoming calls from a SIP trunk can be modified using Calling Party Transformation and Called Party Transformation patterns, which are applied to the trunk's inbound settings. These patterns strip the +1 prefix before the numbers are processed by the dial plan, ensuring the 10-digit format is used. This approach is native to CUCM and keeps digit manipulation scoped to the specific trunk.

Exam trap

The trap here is assuming that a Translation Pattern is the only way to manipulate incoming digits, when trunk-level transformation patterns are the more precise and intended mechanism for SIP trunk inbound calls.

Full explanation →
64
MCQhard

You are implementing a QoS policy on a WAN edge router. You have a requirement to prioritize video traffic. Which queueing mechanism is recommended by Cisco to minimize jitter and latency for video streams?

A.Weighted Fair Queuing (WFQ)
B.Priority Queuing (PQ) only
C.Random Early Detection (RED)
D.Low Latency Queuing (LLQ)
AnswerD

LLQ assigns video traffic to a strict-priority queue with a configured bandwidth guarantee, so latency-sensitive packets are drained ahead of other classes. This directly minimises jitter and latency, satisfying the stem's video prioritisation requirement, while the policed priority limit prevents starvation of remaining traffic classes.

Why this answer

Low Latency Queuing (LLQ) is the Cisco-recommended mechanism because it combines a strict priority queue for delay-sensitive traffic (voice/video) with CBWFQ for all other classes, ensuring voice gets absolute priority while preventing starvation of other traffic. This minimizes jitter and latency for real-time streams.

Exam trap

CLCOR often tests the distinction between fair-queuing mechanisms (WFQ, CBWFQ) and strict-priority mechanisms (PQ, LLQ) — candidates must remember that only LLQ provides both strict priority for voice and bandwidth guarantees for other classes without starvation.

How to eliminate wrong answers

Option A is wrong because WFQ provides fair bandwidth sharing based on flow weights but offers no strict priority — it cannot guarantee the low jitter voice requires. Option B is wrong because Priority Queuing alone gives strict priority but can starve lower-priority traffic and lacks the class-based bandwidth guarantees of LLQ. Option C is wrong because RED is a congestion-avoidance mechanism that drops packets probabilistically to prevent tail drop — it does not prioritize or queue voice traffic.

Full explanation →
65
MCQmedium

You are configuring a Cisco IOS XE voice gateway to route inbound SIP calls to different dial peers based on the Request-URI domain. You want to match calls destined to 'example.com' to a specific dial peer. Which configuration element should you use to achieve this?

A.translation-profile
B.voice class uri 10 sip
C.dial-peer voice 100 voip
D.voice-class codec
AnswerB

A 'voice class uri' defines a URI pattern that can be referenced in an inbound dial peer with 'incoming uri via' or 'incoming uri request'. This allows matching based on the Request-URI domain, such as 'example.com', so the gateway can route calls appropriately. It is the standard method for URI-based dial-peer matching on Cisco IOS XE gateways.

Why this answer

The correct approach is to define a voice class URI that specifies the domain pattern, then reference it in an inbound dial peer using 'incoming uri request' or 'incoming uri via'. This allows the gateway to inspect the SIP Request-URI and select the appropriate dial peer. Other options like translation profiles or codec classes operate after dial-peer matching and cannot influence the selection.

Exam trap

The trap here is confusing URI matching with digit manipulation or codec selection, assuming that a translation profile or codec class can control inbound dial-peer selection.

Full explanation →
66
MCQeasy

Which component is required to synchronize users from an on-premises Active Directory to the Webex Cloud?

A.Cisco Directory Connector
B.Webex Calling Admin Console
C.Expressway-C
D.Cisco Webex HDS
AnswerA

Cisco Directory Connector runs on-premises, reads the Active Directory forest, and pushes user and group objects to Webex Control Hub via secure synchronisation. No other component bridges on-premises AD to the Webex Cloud, satisfying the stem's synchronisation requirement.

Why this answer

Cisco Directory Connector is the on-premises application that synchronizes users from Active Directory to Webex Cloud. It is specifically designed for directory synchronization in Webex Hybrid Services, ensuring user accounts are consistent between on-premises AD and the cloud.

Exam trap

CLCOR often tests the specific component names for hybrid services, and candidates may confuse Directory Connector with Expressway or other management consoles.

How to eliminate wrong answers

Option B is wrong because the Webex Calling Admin Console is for managing Webex Calling configurations, not for directory synchronization. Option C is wrong because Expressway-C is used for hybrid services connectivity (e.g., Calendar, Call, Message), not for AD user synchronization. Option D is wrong because Cisco Webex HDS (Hybrid Data Security) is for message encryption and compliance, not directory sync.

Full explanation →
67
MCQhard

You are designing a Cisco Unified Communications Manager (CUCM) dial plan for a multinational company. The company has offices in North America, Europe, and Asia. Each region has its own PSTN gateway and dial plan requirements. The customer wants to minimize dial plan complexity and ensure that users can dial local numbers as they are accustomed to. Which approach should you recommend?

A.Use a single dial plan with no partitions and rely on translation patterns to handle regional differences.
B.Implement separate CUCM clusters for each region to isolate dial plans.
C.Deploy a centralized dial plan with location-specific partitions and calling search spaces (CSS) to provide regional dialing habits.
D.Implement a single global dial plan with a uniform dialing pattern for all regions.
AnswerC

Using partitions and CSS, you can create a centralized dial plan that allows each region to have its own dialing patterns while sharing a common CUCM cluster. This minimizes complexity by centralizing administration, but still provides the flexibility for local dialing habits. Each region's phones are assigned a CSS that grants access to the appropriate partitions.

Why this answer

A centralized dial plan with partitions and calling search spaces allows each region to have its own dialing patterns while maintaining a single CUCM cluster. This reduces complexity by centralizing administration, but provides the flexibility for local dialing habits. The other options either do not meet the requirement or introduce unnecessary complexity.

Exam trap

The trap here is assuming that a single global dial plan or separate clusters are the only ways to handle multinational dialing, when partitions and CSS provide a more efficient solution.

Full explanation →
68
Multi-Selecteasy

Which TWO of the following are valid components of a Cisco IOS voice gateway configuration?

Select 2 answers
A.Phone button template
B.Dial-peers
C.CUCM route pattern
D.Voice translation rules
E.Device pool
AnswersB, D

Dial-peers are the core matching construct in Cisco IOS voice gateways, pairing inbound call legs with outbound ones via destination patterns. Without them, the gateway cannot route POTS-to-VoIP or VoIP-to-POTS calls, so they are a mandatory component of any voice gateway configuration.

Why this answer

Dial-peers (B) are a core component of Cisco IOS voice gateway configuration, defined globally with the dial-peer voice command to match inbound/outbound call legs and specify destination patterns, sessions targets, and codecs. Voice translation rules (D) are also valid IOS voice gateway components, configured with the voice translation-rule and voice translation-profile commands to manipulate calling/called numbers via regex. In contrast, a phone button template (A) is a Cisco Unified Communications Manager (CUCM) configuration element applied to IP phones, not an IOS gateway feature.

A CUCM route pattern (C) is configured on CUCM to route calls toward gateways or trunks, not on the IOS gateway itself. A device pool (E) is likewise a CUCM construct that groups devices for common settings such as region, date/time, and softkey templates, and is not part of IOS gateway configuration.

Exam trap

CLCOR often tests the boundary between IOS gateway and CUCM configuration — candidates confuse CUCM constructs (device pools, phone button templates, route patterns) with IOS voice gateway components (dial-peers, translation rules, voice classes).

Full explanation →
69
MCQhard

A collaboration engineer is designing a CUCM cluster with two nodes: a publisher and a subscriber. The engineer must ensure that the cluster can support 2500 IP phones and that call processing continues if the publisher fails. Which statement accurately describes the design requirement?

A.The subscriber can register up to 2500 phones, and if the publisher fails, the subscriber continues call processing but database changes are not possible until the publisher is restored.
B.The cluster can support up to 2500 phones per node, but only if the publisher and subscriber are in the same LAN.
C.The subscriber must be configured to handle all call processing, and the publisher should only be used for database and TFTP services.
D.The publisher must be dedicated solely to database replication and not perform call processing.
AnswerA

In a two-node cluster, the subscriber can handle call processing for up to 2500 phones (depending on the OVA). If the publisher fails, the subscriber continues to process calls, but database replication stops, and no configuration changes can be made until the publisher is restored. This is a key consideration for high availability.

Why this answer

In a two-node CUCM cluster, the subscriber can register up to 2500 phones and handle call processing. If the publisher fails, the subscriber continues to process calls, but database replication and configuration changes are halted until the publisher is back online. This design provides call processing redundancy but not database redundancy without additional nodes.

Exam trap

The trap here is assuming that the publisher cannot perform call processing or that a two-node cluster cannot support 2500 phones; the actual limitation is that database changes are unavailable during publisher outage.

Full explanation →
70
MCQmedium

You are troubleshooting a scenario where G.711 is being used instead of the expected Opus codec between two endpoints in the same region. What is the most likely cause in the CUCM configuration?

A.Audio Codec Preference List
B.Phone NTP settings
C.SIP Trunk Security Profile mismatch
D.Location-based bandwidth limit
AnswerA

The Audio Codec Preference List defines the ordered codecs an endpoint offers during negotiation. If G.711 ranks above Opus on that list, the endpoints select G.711 despite Opus being available, directly explaining the unexpected codec in the same region.

Why this answer

The Audio Codec Preference List in CUCM defines the ordered list of codecs an endpoint will offer and accept during SIP/SDP negotiation. If Opus is not placed above G.711 (or is missing entirely) in the assigned codec preference list, the endpoints will fall back to G.711 even when both support Opus. Since both endpoints are in the same region, no location-based bandwidth constraint would force G.711, so the codec preference list is the most likely culprit.

Exam trap

CLCOR often tests the misconception that codec selection is driven by bandwidth or security settings, when in fact the Audio Codec Preference List and region configuration are the authoritative controls for codec negotiation between endpoints.

How to eliminate wrong answers

Option B is wrong because Phone NTP settings only control time synchronization for IP phones and have no bearing on which audio codec is negotiated in the SDP offer/answer. Option C is wrong because a SIP Trunk Security Profile mismatch affects TLS/SIP signaling authentication and encryption, not the media codec chosen between two registered endpoints. Option D is wrong because Location-based bandwidth limits only restrict the maximum bitrate allowed between locations; within the same region, Opus and G.711 both fit comfortably, and bandwidth limits do not reorder codec preference.

Full explanation →
71
MCQhard

A user reports that their calls are failing when they roam to a different physical office site. You observe that the device is registered to the primary CUCM cluster but the Location-based CAC is blocking the call. What is the most likely cause?

A.The device is not assigned to a Device Pool with the correct Location configured for the new site.
B.The Region setting in the Device Pool has a lower bitrate than the required codec.
C.The device must be re-registered to a local subscriber instead of the primary cluster.
D.The Calling Search Space is not configured to allow inter-site dialing.
AnswerA

Location-based CAC derives its bandwidth budget from the Location configured on the device pool. If the roaming device keeps its original site's device pool, CUCM applies the wrong Location's limits, blocking the call. Assigning a device pool with the new site's Location corrects the calculation.

Why this answer

Location-based Call Admission Control (CAC) in Cisco Unified Communications Manager (CUCM) uses the Location configured in the Device Pool to determine available bandwidth between sites. If a device roams to a different physical site but remains in a Device Pool with the original Location, the CAC will incorrectly calculate bandwidth as if the device is still at the original site, potentially blocking calls due to insufficient bandwidth. The most likely cause is that the device is not assigned to a Device Pool with the correct Location for the new site.

Exam trap

CLCOR often tests the confusion between Location-based CAC and Region settings, or the assumption that re-registration affects CAC. The trap is overlooking that Location is tied to the Device Pool, not the registration server.

How to eliminate wrong answers

Option B is wrong because the Region setting defines the codec policy between locations, not the bandwidth available; a lower bitrate in the Region would affect codec selection but not directly cause CAC to block the call. Option C is wrong because re-registering to a local subscriber does not affect Location-based CAC; CAC is based on the Location configuration, not the registration server. Option D is wrong because the Calling Search Space (CSS) controls dialing privileges, not bandwidth allocation; a CSS issue would result in call routing failures, not CAC blocking.

Full explanation →
72
MCQmedium

Which component in the CUCM dial plan allows you to match a pattern and simultaneously transform it into a new number before the final route lookup?

A.Route Pattern
B.Partition
C.Translation Pattern
D.Route Group
AnswerC

A translation pattern matches the dialled digit string and applies digit manipulation, such as discarding or prefixing digits, before the resulting number undergoes further route lookup. This satisfies the simultaneous match-and-transform requirement, unlike route patterns, which match and route without altering the called number.

Why this answer

A Translation Pattern in CUCM matches a dialed pattern and transforms it into a new number before the final route lookup. It can add, remove, or replace digits and then apply a CSS to route the translated number. This is used for dial plan normalization, such as converting a 4-digit extension to an E.164 number or implementing global dialing.

Exam trap

CLCOR often tests the difference between Translation Patterns (transform before routing) and Route Patterns (route without transformation), so candidates may pick Route Pattern when the question mentions transformation.

How to eliminate wrong answers

Option A is wrong because a Route Pattern matches a pattern and routes the call to a gateway or trunk, but it does not transform the number before routing (though it can discard digits, it doesn't rewrite the called number for further lookup). Option B is wrong because a Partition is a logical grouping of dialable numbers used with CSS for access control, not for number transformation. Option D is wrong because a Route Group is a list of gateways for outbound routing, not a transformation mechanism.

Full explanation →
73
MCQhard

A network designer is planning a centralized call processing deployment with 20 remote sites. Each site has 50 IP phones and uses G.711 for internal calls. The designer must ensure that calls between sites use G.729 to conserve WAN bandwidth, but calls within a site use G.711. Which CUCM configuration element should be used to achieve this?

A.Location
B.Region
C.Device Pool
D.Media Resource Group List (MRGL)
AnswerB

Regions define the maximum codec for calls between devices assigned to different regions. By placing each site's devices in its own region and configuring inter-region links to use G.729, while intra-region calls default to G.711, the designer can enforce the required codec policy. This directly controls call bandwidth between sites.

Why this answer

Regions are the correct configuration element to enforce codec selection between sites. By defining regions and inter-region relationships with G.729 as the maximum codec, while allowing G.711 within a region, the designer ensures that inter-site calls use the lower-bandwidth codec. This directly meets the requirement without relying on call admission control.

Exam trap

The trap here is confusing Locations with Regions; Locations manage bandwidth and call admission, while Regions control which codecs are permitted between groups of devices.

Full explanation →
74
MCQhard

A company is deploying Webex Hybrid Video Mesh to optimize media traversal for internal meetings. The administrator has deployed two Video Mesh nodes in a cluster. During testing, users on the internal network report poor video quality when joining meetings hosted by the Video Mesh nodes, while external users have good quality. What is the most likely cause?

A.The internal clients cannot reach the Video Mesh nodes due to firewall or routing issues.
B.The Video Mesh nodes are not configured with public IP addresses.
C.The Video Mesh cluster is not configured for high availability.
D.The Video Mesh nodes are not registered with Cisco Unified Communications Manager.
AnswerA

Internal clients need to be able to reach the Video Mesh nodes directly on the internal network. If firewall rules or routing configurations prevent this, clients will fall back to the Webex cloud for media, which can cause poor quality due to suboptimal paths or additional latency. This is a common misconfiguration when deploying Video Mesh, as internal clients must have direct IP connectivity to the Video Mesh nodes.

Why this answer

For Video Mesh to improve media quality, internal clients must have direct IP connectivity to the Video Mesh nodes. If firewall rules or routing prevent this, clients will use the Webex cloud for media, which may result in poor quality. Ensuring that internal clients can reach the Video Mesh nodes on the required ports is essential for optimal performance.

Exam trap

The trap here is assuming that Video Mesh nodes need public IP addresses or CUCM registration, when in fact they are internal components that require direct client connectivity.

Full explanation →
75
MCQmedium

Which feature allows you to group multiple gateways and route lists to provide load balancing and failover for outbound calls?

A.Route Pattern
B.Translation Pattern
C.Route List
D.Device Pool
AnswerC

A Route List groups route groups and gateways, and distributes outbound calls across them using a routing algorithm, providing both load balancing and failover. This satisfies the stem's requirement to group gateways and route lists for resilient outbound call routing.

Why this answer

A Route List is the CUCM object that groups one or more Route Groups (each containing gateways/trunks) in a prioritized order, providing both load balancing across members and failover to the next member if the first is unavailable. Route Lists are then referenced by Route Patterns to determine which gateways handle outbound calls. This makes them the correct answer for grouping gateways for outbound redundancy.

Exam trap

CLCOR often tests the confusion between Route Pattern (digit matching) and Route List (gateway grouping/failover) — candidates pick Route Pattern because it 'routes' calls, but it does not provide load balancing or failover.

How to eliminate wrong answers

Option A is wrong because a Route Pattern only matches dialed digits and points to a Route List, Route Group, gateway, or trunk — it does not itself group gateways for load balancing. Option B is wrong because a Translation Pattern rewrites dialed digits (number manipulation) and has no role in gateway selection or failover. Option D is wrong because a Device Pool groups devices for common settings like region, date/time group, and SRST reference, not for outbound gateway selection.

Full explanation →

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