Cisco Implementing Cisco Collaboration Core Technologies (CLCOR, 350-801, v2.0 blueprint, CCNP Collaboration) (CLCOR) CLCOR Questions 76–150 | Page 2/4

MCQhard

You are designing a high-availability solution for Cisco Expressway. Which element is required to load-balance traffic between multiple Expressway nodes?

A.Cisco CUBE redundancy

B.Cisco UCM Group

C.External Load Balancer

D.DNS Round Robin only

AnswerC

Cisco Expressway clusters do not natively distribute inbound signalling across nodes; an external load balancer is required to spread traffic between them. This satisfies the high-availability load-balancing requirement, since DNS round-robin alone cannot health-check or evenly balance the Expressway nodes.

Why this answer

To load-balance traffic across multiple Expressway nodes (e.g., Expressway-C or Expressway-E clusters), an external load balancer is required because Expressway does not provide built-in load balancing. The load balancer distributes incoming connections across the nodes for scalability and high availability.

Exam trap

CLCOR often tests the misconception that DNS Round Robin or built-in clustering provides load balancing for Expressway, when an external load balancer is mandatory for true HA and scalability.

How to eliminate wrong answers

Option A is wrong because Cisco CUBE redundancy is for voice gateways and SIP trunk high availability, not for Expressway clustering. Option B is wrong because a Cisco UCM Group is a CUCM concept for phone registration redundancy, not for Expressway load balancing. Option D is wrong because DNS Round Robin alone is not sufficient for Expressway high availability; it does not perform health checks or session persistence, and Expressway requires a proper load balancer for clustering.

Full explanation →

MCQmedium

You are troubleshooting a registration issue where the phone sends a REGISTER request, but the server does not respond. What is the most likely issue?

A.Incorrect Model

B.Device Pool mismatch

C.Wrong Password

D.Network Connectivity/Firewall

AnswerD

A REGISTER request that reaches the server but draws no response indicates the reply is blocked or the path is broken. Firewall rules or routing commonly drop SIP signalling, satisfying the stem's no-response symptom, whereas authentication failures would return a 4xx rejection.

Why this answer

If the phone sends a REGISTER request but the server does not respond at all, the most likely cause is a network connectivity or firewall issue blocking the traffic. If the server received the request, it would typically respond with a 401 Unauthorized challenge or a 200 OK. A complete lack of response indicates the packet is not reaching the server or the response is blocked.

Exam trap

CLCOR often tests the difference between no response and an error response; candidates pick wrong password because they assume authentication failure, but a wrong password yields a 401/403, not silence.

How to eliminate wrong answers

Option A is wrong because an incorrect phone model would not prevent the server from responding; the phone would still send a REGISTER and the server would process it, possibly rejecting with an error. Option B is wrong because a device pool mismatch is a configuration issue that would typically result in the server responding with an error or the phone failing to register after a response, not a total lack of response. Option C is wrong because a wrong password would cause the server to respond with a 401 Unauthorized or 403 Forbidden, not silence.

Full explanation →

MCQeasy

A collaboration administrator needs to provide users with the ability to make and receive PSTN calls through Webex Calling without deploying any on-premises gateways or maintaining a separate carrier contract. Which Webex Calling PSTN option should the administrator choose?

A.Webex Edge for Calling

B.Cisco PSTN

C.Local Gateway

D.Cloud Connected PSTN (CCP)

AnswerB

Cisco PSTN is a cloud-delivered PSTN service from Cisco that provides local, long-distance, and emergency calling directly through the Webex Calling cloud. It requires no on-premises gateway and no separate carrier contract, so it perfectly matches the administrator's need for a completely cloud-based PSTN option.

Why this answer

Cisco PSTN is the only Webex Calling PSTN option that is entirely cloud-delivered by Cisco, with no on-premises gateway and no separate carrier contract needed. It provides dial tone, emergency calling, and number management directly through the Webex Calling cloud, making it the correct choice for an organization that wants to avoid on-premises PSTN infrastructure and additional carrier agreements.

Exam trap

The trap here is confusing Cloud Connected PSTN with Cisco PSTN, assuming that any cloud-based PSTN option eliminates the need for a separate carrier contract, when only Cisco PSTN is fully delivered by Cisco without a third-party carrier.

Full explanation →

MCQmedium

You are troubleshooting a scenario where internal calls between two offices are using the G.711 codec, but you want them to use G.729 to save bandwidth. Where should you change the codec setting?

A.Device > Device Settings > Common Device Configuration

B.System > Region Information > Region

C.System > Device Pool

D.System > Location

AnswerB

System > Region Information > Region sets the codec for calls within a region, which matches the inter-office internal call constraint. Regions define bandwidth management and codec policy per site, so assigning both offices to one region and selecting G.729 there forces internal calls to negotiate G.729 instead of G.711.

Why this answer

In Cisco Unified Communications Manager, codec selection for calls between locations is controlled by regions. Regions define the codec used for calls between devices in different regions. To change the codec between two offices, you modify the region settings (System > Region Information > Region) and set the appropriate codec (e.g., G.729) for the region pair.

Exam trap

CLCOR often tests the confusion between Region (codec selection) and Location (call admission control) — candidates incorrectly pick Location to change codec.

How to eliminate wrong answers

Option A is wrong because Common Device Configuration is used for device-level settings like softkey templates, not codec selection. Option C is wrong because Device Pool is used to group devices with common settings like date/time, region, and location, but the codec itself is defined in the region. Option D is wrong because Location is used for call admission control and bandwidth management, not for selecting the codec.

Full explanation →

Multi-Selectmedium

Which TWO of the following describe the purpose of the Webex Edge for Devices service?

Select 2 answers

A.Automatically provisions local gateways

B.Provides a SIP trunking gateway

C.Replaces the need for a web proxy

D.Enables hybrid registration (CUCM + Cloud)

E.Allows cloud-based management of on-premises registered devices

AnswersD, E

Webex Edge for Devices enables hybrid registration, letting on-premises CUCM-registered endpoints connect to Webex cloud services for calling, meetings and messaging without full cloud migration. This satisfies the stem's requirement by describing a genuine purpose: bridging existing CUCM infrastructure with cloud-delivered capabilities while preserving the on-premises call control deployment.

Why this answer

Option D is correct because Webex Edge for Devices enables hybrid registration, allowing endpoints to register to on-premises Cisco Unified Communications Manager (CUCM) while still consuming Webex cloud services such as Webex Meetings and Webex Calling. Option E is correct because the service provides cloud-based management and control of devices that remain registered on-premises, letting administrators manage them through Control Hub. Options A and B are incorrect because Webex Edge for Devices is not a local gateway provisioning tool nor a SIP trunking gateway; those functions belong to services like Cisco Unified Border Element (CUBE) or Webex Edge Connect.

Option C is incorrect because Webex Edge for Devices does not replace a web proxy; proxy requirements are handled separately by the organization's network and Webex Edge Connect design.

Exam trap

CLCOR often tests the two core purposes of Webex Edge for Devices (hybrid registration + cloud management), and candidates confuse it with gateway, proxy, or SIP trunking functions that belong to other services.

Full explanation →

MCQhard

What is the primary difference between a 'Route Pattern' and a 'Translation Pattern'?

A.Translation patterns route calls; route patterns modify numbers.

B.There is no functional difference.

C.Route patterns route calls; translation patterns modify numbers.

D.Route patterns are for internal calls; translation patterns are for external.

AnswerC

Route patterns select the path and destination for an outbound call, whereas translation patterns rewrite the dialled digits before further processing. That axis of difference — routing versus digit manipulation — is precisely what the question asks you to distinguish.

Why this answer

Route patterns are the dial plan elements that actually route calls to a destination (gateway, trunk, or another DN) by matching the dialed digits and applying a route list/route group. Translation patterns, by contrast, do not route calls on their own — they transform the dialed digits (via discard/prefix digits, calling/called party transformations) and then hand the call back to the dial plan for further matching. This separation lets administrators normalize numbers before routing decisions are made.

Exam trap

CLCOR often tests the misconception that translation patterns can route calls on their own, when in fact they only transform digits and must hand off to a route pattern or DN for actual routing.

How to eliminate wrong answers

Option A is wrong because it reverses the roles — translation patterns modify digits, not route them, and route patterns route rather than modify. Option B is wrong because the two pattern types serve distinct functions in CUCM dial plan processing; they are not interchangeable. Option D is wrong because both pattern types can apply to internal and external calls; the distinction is functional (routing vs. digit manipulation), not internal vs. external scope.

Full explanation →

MCQeasy

You are deploying CUBE. Which command enables the CUBE functionality on the Cisco IOS XE router?

A.enable cube

B.mode border-element

C.voice-service enable

D.service cube

AnswerB

Entering voice service configuration and issuing mode border-element switches the router into Cisco Unified Border Element mode, activating CUBE's session border controller functions such as SIP trunking, dial-peer handling and media interworking on the IOS XE platform.

Why this answer

CUBE (Cisco Unified Border Element) is enabled on IOS XE by entering voice service configuration mode and issuing the 'mode border-element' command. This switches the router's voice service into border-element mode, activating the SIP-based session border controller functionality that provides demarcation, protocol interworking, and security between the enterprise and the service provider.

Exam trap

CLCOR often tests the exact IOS XE syntax hierarchy for enabling CUBE, luring candidates toward plausible-sounding but non-existent commands like 'enable cube' or 'service cube' instead of the real 'voice service voip / mode border-element' path.

How to eliminate wrong answers

Option A is wrong because 'enable cube' is not a valid IOS XE command — there is no top-level 'enable cube' syntax; CUBE is activated through voice service configuration mode. Option C is wrong because 'voice-service enable' is not a real command; the correct entry point is 'voice service voip' followed by 'mode border-element'. Option D is wrong because 'service cube' does not exist in IOS XE; the keyword is 'border-element' under voice service voip, not a standalone 'service' command.

Full explanation →

MCQmedium

When configuring CUBE, why would you use a 'voice class uri' command?

A.To define codec

B.To match SIP URI host

C.To route by time

D.To match caller ID

AnswerB

A voice class uri groups SIP URIs by host or pattern so dial peers can match incoming or outgoing requests against them. This lets CUBE route or apply policy based on the URI host portion rather than only IP addresses or dialled digits.

Why this answer

The 'voice class uri' command in CUBE is used to define a class of SIP URIs for matching purposes, typically to match the host portion of a SIP URI. This allows CUBE to categorize incoming or outgoing SIP requests based on the URI host, which can then be referenced in dial peers for routing or policy decisions. It is a key tool for URI-based routing and manipulation.

Exam trap

CLCOR often tests the misconception that 'voice class uri' is used for caller ID or codec matching, when it is specifically designed to match SIP URI components like the host, and candidates may confuse it with other voice class types.

How to eliminate wrong answers

Option A is wrong because codec selection is configured using 'voice class codec' or directly under dial peers, not 'voice class uri'. Option C is wrong because time-based routing is handled by time-of-day routing configurations, such as 'time-range' and dial peer preferences, not by URI classes. Option D is wrong because matching caller ID is done with 'voice class calling-number' or similar, not 'voice class uri', which focuses on the URI itself.

Full explanation →

Multi-Selecthard

Which THREE items are required to successfully register a room device to Webex?

Select 3 answers

A.A valid Webex Device license

B.A local SIP trunk

C.An existing CUCM registration

D.A valid activation code from Control Hub

E.An active internet connection

AnswersA, D, E

A valid Webex Device license is mandatory because Webex cloud enforces licensing at registration: the device's MAC address must be associated with a paid device entitlement before the RoomOS device can activate against Webex. Without it, registration fails authorisation, satisfying the stem's licensing constraint.

Why this answer

Option A is correct because registering a room device to Webex requires a valid Webex Device license assigned to the device in Control Hub, which authorizes the device to use Webex cloud services. Option D is correct because the device must be onboarded using a valid activation code generated from Control Hub, which the device uses to authenticate and complete registration. Option E is correct because the device needs an active internet connection to reach Webex cloud endpoints and complete the cloud registration process.

Options B and C are not required: a local SIP trunk and an existing CUCM registration are on-premises telephony concepts and are not prerequisites for Webex cloud device registration.

Exam trap

CLCOR often tests the misconception that a SIP trunk or CUCM registration is needed for Webex cloud registration — candidates confuse on-premises calling prerequisites with cloud onboarding requirements.

Full explanation →

MCQmedium

You are observing that video traffic is marked as DSCP 0. How can you fix this at the network level without changing the endpoint configuration?

A.Disable traffic shaping

B.Use an ACL to match the traffic and a policy-map to re-mark it

C.Change the interface speed

D.Enable auto-qos on the interface

AnswerB

Re-marking at the network layer via an ACL classifies the DSCP 0 video flow, and the policy-map rewrites its DSCP value on ingress, so endpoints stay untouched. This satisfies the constraint of fixing marking without changing endpoint configuration.

Why this answer

When video endpoints send DSCP 0 (best-effort), the marking must be corrected in the network path. A class-map matches the traffic (e.g., by ACL identifying RTP/Video over the correct port range), and a policy-map with a set dscp command re-marks it to the desired value such as AF41 for interactive video. Applying this via MQC (Modular QoS CLI) on the ingress interface fixes the marking without touching the endpoint configuration.

Exam trap

CLCOR often tests the misconception that auto-qos will fix bad markings — it actually trusts or applies a fixed template, so if the endpoint sends DSCP 0, you need explicit MQC re-marking.

How to eliminate wrong answers

Option A is wrong because disabling traffic shaping removes rate limiting but does nothing to change the DSCP value carried in the packet header. Option C is wrong because changing interface speed affects bandwidth and congestion behavior, not packet marking. Option D is wrong because auto-qos applies a predefined trust-and-mark template based on what the endpoint already sends — if the endpoint sends DSCP 0, auto-qos will not magically upgrade it to a video class; it typically trusts existing markings or applies a fixed template that may not match the desired video DSCP.

Full explanation →

Multi-Selecteasy

Which TWO features are provided by the Cisco Webex App when registered as a Softphone?

Select 2 answers

A.TFTP service hosting

B.SIP signaling

C.DNS root server

D.RTP media transmission

E.DHCP server functionality

AnswersB, D

Required for call control.

Why this answer

The Cisco Webex App, when registered as a softphone, functions as a SIP endpoint, so it performs SIP signaling (Option B) to set up, manage, and tear down calls with the call control platform. It also handles RTP media transmission (Option D), sending and receiving the actual audio/video packets for the call. These two functions are core to any softphone client, which must both signal and carry media.

Options A, C, and E are incorrect because a softphone does not host TFTP configuration files, act as a DNS root server, or provide DHCP address assignment; those are infrastructure services typically provided by dedicated servers or network devices.

Exam trap

CLCOR often tests the distinction between endpoint functions (SIP signaling, RTP media) and network infrastructure services (TFTP, DNS, DHCP), so candidates may mistakenly select infrastructure roles that a softphone does not provide.

Full explanation →

MCQeasy

In CUCM, what is the purpose of the 'Calling Search Space' (CSS)?

A.To define the codec used for calls.

B.To define which partitions a device can access.

C.To define the physical location of the device.

D.To define the bandwidth limit of a call.

AnswerB

A Calling Search Space is an ordered list of partitions that a device or line may search when placing a call, determining which dialled numbers it can reach. This satisfies the constraint by controlling partition access, which combined with partition membership enforces call-routing permissions in CUCM.

Why this answer

A Calling Search Space is an ordered list of partitions that a device or line is allowed to search when matching dialed digits. It effectively defines the class of service for that device by determining which route patterns, translation patterns, and directory numbers it can reach. Without a CSS, a device has no partitions to search and cannot complete any calls beyond its own registered DN.

Exam trap

CLCOR often tests the confusion between CSS (which partitions a device can search) and Partitions (which dialable objects exist), causing candidates to pick answers about codecs, locations, or bandwidth instead of dial-plan access.

How to eliminate wrong answers

Option A is wrong because codec selection is controlled by regions, device pools, and the negotiated codec list on the SIP profile or phone configuration, not by the CSS. Option C is wrong because physical location is represented by the Location and Device Pool configuration, which drive CAC bandwidth calculations, not by the CSS. Option D is wrong because bandwidth limits per call are enforced by regions and locations (and their associated links), not by the CSS.

Full explanation →

MCQhard

You are implementing a centralized dial plan. You have multiple sites, each with local PSTN gateways. How can you ensure that an emergency call from a specific site is always routed to the local PSTN gateway of that site?

A.Use Transformation Masks on the device.

B.Configure the location in the device pool to match the site.

C.Create a separate CUCM cluster for each site.

D.Use site-specific partitions and assign the local gateway to a CSS that includes that partition.

AnswerD

Site-specific partitions isolate each site's emergency number, and assigning only that site's local gateway to the CSS forces calls in that partition out through the local PSTN gateway. This guarantees the emergency call reaches the correct local emergency services rather than a remote gateway.

Why this answer

Emergency calls (911/999/112) must always egress through the PSTN gateway physically closest to the caller so that the correct Public Safety Answering Point (PSAP) is reached. The reliable way to enforce this in a centralized dial plan is to place each site's local gateway in a site-specific partition, then assign each site's devices a Calling Search Space (CSS) that includes only that site's partition. This guarantees that when a device dials the emergency number, the matching route pattern in the local partition is the one that gets used.

Exam trap

CLCOR often tests whether candidates confuse Location (used for CAC and E911 ELIN) with partitions/CSS (used for actual gateway selection), causing them to pick the Device Pool location answer.

How to eliminate wrong answers

Option A is wrong because Transformation Masks only rewrite the calling or called number digits — they do not influence which gateway or route pattern is selected for call routing. Option B is wrong because the Location in a Device Pool is used for call admission control (bandwidth management) and for E911/ELIN location information, not for selecting which PSTN gateway a call egresses through. Option C is wrong because deploying a separate CUCM cluster per site is an unnecessarily complex and costly design; a centralized cluster with partitions and CSSs achieves the same routing outcome far more simply.

Full explanation →

MCQeasy

A user reports that when they dial an internal extension, they hear a fast busy tone. The administrator checks the dialed number and confirms it exists in the CUCM database. Which configuration element should the administrator verify first to resolve the fast busy tone?

A.The device pool assigned to the called device

B.The media resource group list assigned to the calling device

C.The partition and calling search space of the calling device

D.The location configuration for both devices

AnswerC

A fast busy tone typically indicates that CUCM cannot find a matching route pattern or directory number in the calling device's calling search space. Even if the number exists, if the calling device's CSS does not include the partition containing that number, the call will be blocked. Verifying the CSS and partition assignment is the first troubleshooting step for internal call failures.

Why this answer

A fast busy tone on an internal call usually means CUCM cannot route the call because the dialed number is not in a partition that the calling device's calling search space can access. Even if the directory number exists, it must be in a partition listed in the calling device's CSS. The other options relate to device settings, locations, or media resources, which do not prevent dialing an internal number.

Therefore, checking the CSS and partition is the correct first step.

Exam trap

The trap here is assuming that if the number exists, the call will always route; CUCM enforces partition/CSS permissions even for internal numbers.

Full explanation →

MCQhard

A company uses Webex Hybrid Calendar Service. Users report that the 'Join' button is missing from their Webex Room devices for scheduled meetings. What is the most likely cause?

A.The Webex site is not integrated with the Control Hub

B.The Room device has not been registered to Webex

C.The Expressway-C does not have a public certificate

D.The user's Exchange mailbox is not assigned to the Hybrid Calendar service

AnswerD

Hybrid Calendar injects the Join button by pairing the user's Exchange mailbox with the service; if that mailbox is not assigned, meeting data never reaches Webex and the button is absent. This is the specific provisioning prerequisite the symptom points to.

Why this answer

The 'Join' button on Webex Room devices for scheduled meetings is provided by the Hybrid Calendar Service, which requires the user's Exchange mailbox to be assigned to the service. If the mailbox is not assigned, the device cannot retrieve calendar information, and the Join button will not appear.

Exam trap

The trap is assuming that the issue is with device registration or certificate problems, when the most common cause is a per-user configuration in Control Hub. Candidates must remember that Hybrid Calendar requires explicit assignment of mailboxes.

How to eliminate wrong answers

Option A is wrong because if the Webex site were not integrated with Control Hub, the device would not be registered or managed at all, but the issue is specific to the Join button. Option B is wrong because if the Room device were not registered to Webex, it would not function as a Webex device, but the question implies it is otherwise working. Option C is wrong because an Expressway-C public certificate issue would affect call signaling or registration, not specifically the calendar Join button.

Full explanation →

MCQmedium

You need to migrate users from an on-premises Cisco Unified Communications Manager (CUCM) to Webex Calling. Which tool allows for the bulk onboarding of users while maintaining existing directory synchronization?

A.Webex Edge for Devices

B.CUCM AXL API bulk exporter

C.CSV Template upload in Control Hub

D.Webex Migration Tool

AnswerC

Control Hub's CSV template bulk-adds users and maps them to existing directory-synchronised identities, satisfying the requirement to preserve directory synchronisation. It avoids manual per-user creation, letting large CUCM populations be onboarded efficiently into Webex Calling.

Why this answer

The CSV template upload in Control Hub is the supported bulk onboarding method for migrating users to Webex Calling while preserving directory synchronization. You export user data (often from Active Directory or CUCM), populate the Webex-provided CSV template with required fields, and upload it to Control Hub, which provisions users and links them to the existing directory sync (e.g., via Azure AD or Cisco Directory Connector). This maintains identity consistency across the migration.

Exam trap

CLCOR often tests the confusion between extraction tools (AXL/BAT) and onboarding tools (Control Hub CSV) — candidates pick the exporter thinking it also provisions, but extraction and onboarding are separate steps.

How to eliminate wrong answers

Option A is wrong because Webex Edge for Devices is a hybrid calling feature that lets on-premises registered devices connect to Webex services — it is not a user migration/onboarding tool. Option B is wrong because the CUCM AXL API bulk exporter extracts configuration data from CUCM but does not onboard users into Webex Calling or maintain directory sync. Option D is wrong because there is no product called 'Webex Migration Tool' in the Webex Calling migration path — the actual tools are Control Hub CSV templates, Directory Connector, and Webex Calling migration utilities, not a single branded 'Migration Tool'.

Full explanation →

MCQmedium

What is the primary function of a dial-peer destination-pattern?

A.To define the codec

B.To specify the gateway IP

C.To match the dialed number

D.To enable SIP

AnswerC

The destination-pattern matches the dialled number string so the router selects the correct outbound dial peer for that call. It functions as the matching criterion, while session target, codec and digit manipulation are configured separately on the same peer.

Why this answer

The primary function of a dial-peer destination-pattern is to match the dialed number (or digits) to determine which dial-peer to use for routing the call. It acts as a matching criterion for inbound and outbound dial-peers. The other options describe different dial-peer attributes.

Exam trap

CLCOR often tests the distinction between dial-peer matching commands; candidates confuse destination-pattern with session target or codec commands because they all appear in dial-peer configuration.

How to eliminate wrong answers

Option A is wrong because the codec is defined by the 'codec' command under voice class or dial-peer, not by destination-pattern. Option B is wrong because the gateway IP is specified by the 'session target' command, not destination-pattern. Option D is wrong because enabling SIP is done with 'session protocol sipv2', not destination-pattern.

Full explanation →

MCQmedium

A company is deploying Webex Hybrid Directory Service to synchronize on-premises Active Directory users to Webex Control Hub. The administrator has installed the connector on a Windows Server 2019 machine and configured the required DNS SRV record. During the connector setup, the administrator is prompted to provide credentials for a service account. Which of the following must be true about this service account?

A.It must have read access to the Active Directory partitions containing the users to be synchronized.

B.It must be a member of the Enterprise Admins group to synchronize users across all domains in the forest.

C.It must be a member of the Domain Admins group in Active Directory.

D.It must be a Microsoft Exchange mailbox-enabled account with a valid SMTP address.

AnswerA

The Hybrid Directory Service connector uses the service account to read user objects from Active Directory and replicate them to the Webex cloud. Only read permissions on the relevant directory partitions are needed. Write permissions, Domain Admin rights, or Exchange-specific roles are unnecessary and would introduce unnecessary security risk.

Why this answer

The Hybrid Directory Service connector authenticates to Active Directory using a service account that must be able to read the directory partitions containing the users to be synchronized. No administrative write privileges or special group memberships are required. This least-privilege model ensures the connector can perform its synchronization task without exposing the environment to unnecessary risk.

Exam trap

The trap here is assuming that a directory synchronization service account requires elevated domain administrative rights, when in fact only read access to the relevant partitions is necessary.

Full explanation →

MCQhard

A collaboration engineer is configuring Webex Hybrid Call Service Connect to integrate an on-premises Cisco Unified Communications Manager (CUCM) cluster with Webex. The engineer has deployed Expressway-C and Expressway-E in the enterprise and has enabled the Hybrid Call Service Connector on the Expressway-C. After configuration, users in Webex Teams cannot place calls to on-premises CUCM-registered endpoints. Which of the following is the most likely cause?

A.The Expressway-C does not have a valid certificate trusted by CUCM for the SIP TLS connection.

B.The Webex Teams users have not been assigned the Webex Calling license.

C.The CUCM cluster is not configured with a SIP trunk to the Expressway-C.

D.The Expressway-E is not configured with a public IP address and cannot reach the Webex cloud.

AnswerA

Hybrid Call Service Connect uses SIP TLS between the Expressway-C and CUCM. If the Expressway-C certificate is not trusted by CUCM (or vice versa), the TLS handshake fails, and calls cannot be established. This is a common misconfiguration when certificates are self-signed or from an untrusted CA. Ensuring mutual certificate trust is essential for the SIP trunk to come up.

Why this answer

Hybrid Call Service Connect relies on a SIP TLS connection between Expressway-C and CUCM. For this connection to work, both sides must trust each other's certificates. If the Expressway-C certificate is not trusted by CUCM, the TLS handshake fails, and calls between Webex Teams and on-premises endpoints cannot be established.

Verifying and importing the correct certificates is a critical troubleshooting step.

Exam trap

The trap here is assuming that Webex Calling licenses are needed for Hybrid Call Service Connect, when in fact the service uses a different entitlement model and the issue is more likely a certificate trust problem.

Full explanation →

MCQmedium

A Cisco IOS XE gateway is configured with two POTS dial peers that both match an inbound PSTN call on voice port 0/0/0. The first dial peer has preference 1 and destination pattern 9T; the second has preference 2 and destination pattern 9.... An engineer notices that calls are always routed using the second dial peer. What is the most likely explanation?

A.The preference command only applies to outbound VoIP dial peers, so the second dial peer is chosen solely because it has a higher preference number.

B.The second dial peer has a lower preference value because the gateway interprets preference 2 as higher priority than preference 1.

C.Inbound POTS dial-peer matching ignores the preference command; the dial peer with the longest match on the called number is selected.

D.The first dial peer is missing the incoming called-number command, so it cannot match inbound calls and the gateway falls back to the second dial peer.

AnswerC

For inbound POTS calls, the gateway matches the called number against destination patterns and selects the dial peer with the longest explicit match, not the lowest preference value. The pattern 9.... matches more digits than 9T, so it wins even though its preference is higher. Preference only breaks ties when multiple dial peers have identical match lengths.

Why this answer

Inbound POTS dial-peer selection uses the longest match on the called number from the destination-pattern, not the preference value. The pattern 9.... matches more digits than 9T, so it is chosen regardless of preference. Preference only acts as a tiebreaker when two dial peers have equal match lengths, which is not the case here.

Exam trap

The trap here is assuming that the preference command is the primary selection criterion for inbound POTS dial peers, when the longest destination-pattern match takes precedence.

Full explanation →

MCQhard

An endpoint sends a SIP REGISTER request. CUCM returns a 401 Unauthorized response. What is the next logical step in the SIP signaling flow for the endpoint?

A.Switch to SCCP

B.Resend REGISTER with Authorization header

C.Terminate the TCP connection

D.Send a SIP INVITE

AnswerB

A 401 Unauthorized response is a challenge demanding credentials, so the endpoint must resend REGISTER including an Authorization header carrying its digest credentials. This directly satisfies the challenge-response authentication constraint in the stem, completing the SIP registration handshake rather than retrying blindly or abandoning the attempt.

Why this answer

When CUCM returns 401 Unauthorized to a SIP REGISTER, it is issuing a challenge under SIP digest authentication (RFC 3261). The endpoint must compute the digest response using its credentials and the challenge nonce, then resend the REGISTER with an Authorization header containing that response. This is the standard challenge-response handshake for SIP registration.

Exam trap

CLCOR often tests whether candidates understand that a 401 is a challenge, not a failure — the trap is treating it as a terminal error and choosing to terminate the connection or switch protocols instead of resending with credentials.

How to eliminate wrong answers

Option A is wrong because switching to SCCP is not a SIP signaling step — SCCP is a different protocol entirely, and endpoints do not switch protocols mid-registration in response to a 401. Option C is wrong because terminating the TCP connection would abort the registration attempt; the 401 is a challenge, not a fatal error, so the connection should remain open for the authenticated retry. Option D is wrong because a SIP INVITE is used to initiate a session (call), not to complete registration; sending an INVITE before successful registration would fail because the endpoint is not yet registered.

Full explanation →

MCQmedium

You are designing a network for video collaboration. What is the recommended maximum one-way latency for high-quality video conferencing?

A.300ms

B.500ms

C.50ms

D.150ms

AnswerD

150ms is the industry-standard threshold for real-time media.

Why this answer

For high-quality, interactive video conferencing, the ITU-T G.114 recommendation specifies that one-way latency should not exceed 150 ms to maintain conversational quality. Beyond 150 ms, participants begin to notice talker overlap and difficulty with turn-taking, degrading the collaboration experience. 150 ms is the accepted threshold for real-time voice and video applications.

Exam trap

CLCOR often tests the confusion between one-way and round-trip latency; candidates who recall '300 ms' from RTT guidelines may incorrectly select 300 ms instead of the 150 ms one-way limit.

How to eliminate wrong answers

Option A (300 ms) is wrong because 300 ms one-way latency is well above the G.114 limit and causes noticeable degradation in interactivity, though it may still be tolerable for one-way streaming. Option B (500 ms) is wrong because 500 ms one-way latency is far too high for real-time video conferencing and would make conversation feel like a satellite call with awkward pauses. Option C (50 ms) is wrong because while 50 ms is excellent and desirable, it is not the recommended maximum — it is an ideal target, not the threshold; the question asks for the maximum acceptable one-way latency.

Full explanation →

MCQeasy

When deploying Cisco IP phones, what is the best practice for VLAN configuration?

A.Use the default VLAN 1 for all voice and data.

B.Use a dedicated Voice VLAN.

C.Use a separate physical switch for phones.

D.Do not use VLANs; use static routing.

AnswerB

A dedicated voice VLAN separates signalling and RTP traffic from data traffic, letting you apply QoS trust boundaries and voice-specific DHCP scopes. This satisfies the stem's best-practise requirement by isolating latency-sensitive voice from broadcast-heavy data, easing troubleshooting and security policy.

Why this answer

Best practice is to place voice traffic on a dedicated Voice VLAN, separate from the data VLAN, so that QoS policies, security, and broadcast domain boundaries can be applied specifically to voice. This also enables features like voice VLAN access (CDP/LLDP-MED) that let the phone tag its own traffic while the PC behind it uses the data VLAN untagged.

Exam trap

CLCOR often tests the misconception that VLAN 1 or physical separation is acceptable for voice — candidates underestimate why a dedicated Voice VLAN is the standard, especially for QoS and LLDP-MED-based phone VLAN assignment.

How to eliminate wrong answers

Option A is wrong because using the default VLAN 1 for both voice and data mixes traffic, prevents clean QoS classification, and is a security and broadcast-domain best-practice violation. Option C is wrong because using a separate physical switch for phones is not a best practice — it's costly, doesn't scale, and VLANs exist precisely to logically separate traffic on shared infrastructure. Option D is wrong because avoiding VLANs and using static routing is nonsensical for voice deployment — VLANs are the standard Layer 2 mechanism for traffic separation, and static routing doesn't address voice segmentation at all.

Full explanation →

MCQmedium

What is the primary function of the CUBE in a SIP trunking scenario?

A.Call routing for internal phones

B.Demarcation and SIP normalization

C.Voice messaging storage

D.DHCP Server functionality

AnswerB

CUBE sits at the network demarcation point, terminating the SIP trunk from the provider and normalising SIP signalling and media between disparate implementations. This satisfies the stem's requirement for the primary function: interworking vendor-specific SIP dialects, headers and codecs so the enterprise call control platform interoperates cleanly with the carrier.

Why this answer

The CUBE (Cisco Unified Border Element) is Cisco's session border controller function that sits at the edge between the enterprise VoIP network and the ITSP's SIP trunk. Its primary role is to provide a demarcation point and perform SIP normalization — translating between different SIP dialects, header formats, and SDP variations so that the enterprise and provider can interoperate. It also handles topology hiding, security, and call admission control at that boundary.

Exam trap

CLCOR often tests the confusion between CUBE's edge demarcation/normalization role and CUCM's internal call routing role — candidates mistakenly pick 'call routing' because CUBE does route calls, but only at the SIP trunk boundary.

How to eliminate wrong answers

Option A is wrong because call routing for internal phones is the function of Cisco Unified Communications Manager (CUCM) or the dial plan, not the CUBE — CUBE only routes at the SIP trunk edge. Option C is wrong because voice messaging storage is provided by Cisco Unity Connection, not CUBE. Option D is wrong because DHCP server functionality is provided by a router, switch, or dedicated DHCP server, not by CUBE.

Full explanation →

100

MCQmedium

When a call is made to a destination that matches multiple partitions in a user's CSS, how does CUCM determine which pattern to use?

A.The pattern configured first.

B.Random selection.

C.Longest match rule.

D.The pattern with the lowest partition index.

AnswerC

When several partitions in the calling user's CSS match the dialled digits, CUCM selects the pattern with the most specific (longest) match, so the longest-match rule resolves the ambiguity and determines which partition's pattern is used.

Why this answer

Cisco Unified Communications Manager (CUCM) uses the longest match rule when a dialed number matches multiple patterns across the partitions in a calling search space (CSS). The system evaluates all patterns in the CSS partitions and selects the one with the most specific (longest) match to the dialed digits. This ensures that more specific dial plans take precedence over generic ones, which is critical for correct call routing.

Exam trap

CLCOR often tests the misconception that CUCM uses the order of partitions in the CSS or the order of pattern configuration to determine which pattern to use, when in fact the longest match rule takes precedence.

How to eliminate wrong answers

Option A is wrong because CUCM does not prioritize based on configuration order; it uses digit length and specificity. Option B is wrong because CUCM's route selection is deterministic, not random; it follows a defined algorithm. Option D is wrong because partition index (order in CSS) is not used to break ties; the longest match rule is applied first, and if multiple patterns have the same length, then the order of partitions in the CSS may be considered, but the primary criterion is longest match.

Full explanation →

101

MCQhard

A user reports that they cannot hold an active call when the Media Resource Group List (MRGL) is configured with a specific Transcoder. The CUCM trace shows 'Allocation Failure'. What is the most likely cause?

A.The CUCM version does not support hardware transcoders.

B.The Transcoder device is registered but the MRGL priority is set to 0.

C.The MRG associated with the MRGL lacks the required Transcoder resource for the codec mismatch.

D.The device is using G.711 which requires a transcoder by default.

AnswerC

The MRGL points to an MRG whose members lack a transcoder capable of bridging the codec mismatch, so CUCM cannot allocate the media resource and logs 'Allocation Failure'. Adding a transcoder to that MRG, or referencing an MRG that contains one, resolves the failed allocation.

Why this answer

The 'Allocation Failure' in CUCM traces during a hold operation with a transcoder in the MRGL indicates that CUCM could not find a usable transcoder resource to satisfy the media requirement. The most likely cause is that the Media Resource Group (MRG) referenced by the MRGL does not actually contain the transcoder needed for the codec mismatch, so the resource lookup fails and the hold (which may require renegotiation or MTP/transcoding) cannot complete.

Exam trap

CLCOR often tests the MRGL/MRG containment hierarchy, tricking candidates into blaming CUCM version support or MRGL priority values when the real issue is that the required resource type is simply absent from the MRG.

How to eliminate wrong answers

Option A is wrong because CUCM has supported hardware transcoders (e.g., PVDM-based DSP farms and IOS-based transcoders) for many versions; version support is not the likely cause of an allocation failure. Option B is wrong because MRGL priority is not set to '0' as a disabling mechanism — resources are ordered by MRG priority, and a priority value does not zero out a device; the failure is about resource availability, not priority numbering. Option D is wrong because G.711 is the baseline codec that does not require transcoding by default; transcoding is needed when endpoints negotiate different codecs (e.g., G.711 to G.729), so this premise is factually incorrect.

Full explanation →

102

Multi-Selecthard

Which THREE of the following are necessary to ensure successful NTP deployment in a CUCM cluster?

Select 3 answers

A.Use broadcast mode

B.Configure external Stratum 1 or 2 servers

C.Open UDP 123 on firewalls

D.Enable NTP on the local voice gateway

E.Consistent configuration across all nodes

AnswersB, C, E

CUCM has no internal authoritative clock, so it must sync to an external reliable time source. Stratum 1 or 2 servers provide accurate reference time, satisfying the stem's requirement. Without reachable external servers, nodes drift and cluster services fail.

Why this answer

Option B is correct because CUCM requires an external NTP source of good quality, and Stratum 1 or 2 servers provide the accurate, reliable time reference needed to keep all cluster nodes synchronized. Option C is correct because NTP communicates over UDP port 123, so any firewall between CUCM and the NTP servers must permit UDP 123 traffic or synchronization will fail. Option E is correct because every node in the CUCM cluster must point to the same consistent NTP configuration so that all servers share a common time source and avoid clock drift that could break database replication and certificate validation.

Option A is not required because CUCM uses unicast NTP to reach specified servers, not broadcast mode. Option D is not required because enabling NTP on a local voice gateway is unrelated to CUCM cluster time synchronization.

Exam trap

CLCOR often tests the misconception that broadcast mode or gateway NTP is required, when the focus should be on external servers, firewall ports, and consistent configuration.

Full explanation →

103

MCQmedium

A Cisco IP Phone 8845 in the New York office registers to CUCM and downloads its configuration. The administrator notices that the phone uses a DHCP scope option to discover the TFTP server address. Which DHCP option is used by the phone for this purpose?

A.Option 42

B.Option 150

C.Option 43

D.Option 66

AnswerB

DHCP Option 150 provides a list of TFTP server IP addresses to Cisco IP phones during network boot. In this scenario, the phone uses Option 150 to locate the TFTP server from which it downloads its configuration and firmware. This is the standard method for Cisco endpoints to discover the TFTP service in a CUCM environment, allowing the phone to proceed with registration.

Why this answer

Cisco IP phones use DHCP Option 150 to obtain a list of TFTP server addresses. This allows the phone to contact the correct TFTP server to download its configuration file and firmware. In a CUCM environment, Option 150 is the standard method for phone boot and registration, ensuring the device can locate the necessary files to become operational.

Exam trap

The trap here is confusing DHCP Option 150 with Option 66, as both can provide TFTP server information, but Cisco phones specifically use Option 150.

Full explanation →

104

Multi-Selecthard

Which THREE of the following are key features provided by a QoS policy-map?

Select 3 answers

A.Bandwidth reservation

B.Traffic marking

C.Traffic policing

D.Interface IP address assignment

E.Dynamic routing updates

AnswersA, B, C

Bandwidth reservation guarantees a minimum share of interface capacity to a class using the priority or bandwidth command within a policy-map. It satisfies the stem by being a core policy-map action, ensuring latency-sensitive or business-critical traffic is not starved during congestion.

Why this answer

A QoS policy-map is the component that applies actions to traffic classes, and bandwidth reservation (A) is a core action it can enforce, typically via the bandwidth or priority commands to guarantee a minimum or strict-priority share of link capacity. Traffic marking (B) is also a key policy-map action, using commands like set dscp or set cos to rewrite QoS bits so downstream devices can classify and treat packets consistently. Traffic policing (C) is another standard policy-map action, implemented with the police command to meter flows and drop or re-mark packets that exceed a committed rate.

The remaining options do not belong: interface IP address assignment (D) is a Layer 3 interface configuration task, not a QoS policy-map function, and dynamic routing updates (E) are handled by routing protocols such as OSPF or EIGRP, not by QoS policy-maps.

Exam trap

CLCOR often tests the distinction between QoS policy-map actions and unrelated interface or routing configurations, so candidates may mistakenly include IP address assignment or routing updates as QoS features.

Full explanation →

105

Multi-Selectmedium

Which TWO of the following are essential components of a Cisco network readiness assessment for collaboration?

Select 2 answers

A.Bandwidth capacity analysis

B.Analyzing NTP Stratum levels

C.QoS configuration audit

D.Verifying phone firmware versions

E.Checking IP address assignment

AnswersA, C

Bandwidth capacity analysis determines whether links and WAN circuits can carry projected voice, video and sharing traffic at peak load. Collaboration flows are bandwidth-intensive and delay-sensitive, so confirming sufficient capacity is essential to a readiness assessment.

Why this answer

Bandwidth capacity analysis (A) is essential because Cisco collaboration services such as voice, video, and conferencing have strict per-call and per-stream bandwidth requirements, and the readiness assessment must confirm the WAN/LAN can carry the projected concurrent media load with adequate headroom. A QoS configuration audit (C) is equally essential because real-time collaboration traffic (RTP media, signaling) must be properly classified, marked, and queued (e.g., EF for voice, AF41 for video) to guarantee low latency, jitter, and packet loss across the network. Analyzing NTP Stratum levels (B) is not a core readiness component, since NTP accuracy matters for call detail records and certificates but stratum level itself is not a standard readiness metric.

Verifying phone firmware versions (D) is a deployment/upgrade task rather than a network readiness assessment element. Checking IP address assignment (E) is a general design consideration, not one of the two essential readiness assessment components for collaboration.

Exam trap

CLCOR often tests whether candidates can separate core readiness pillars (bandwidth, QoS) from peripheral operational tasks (firmware, NTP, IP assignment) — the trap is picking device-level checks as 'essential components.'

Full explanation →

106

MCQeasy

Which license type is required to manage Webex Calling services in Control Hub?

A.Webex Meetings License

B.Webex Calling License

C.Webex Room Device License

D.Webex Messaging License

AnswerB

A Webex Calling licence is mandatory because Control Hub only exposes calling configuration—users, numbers, and PSTN settings—when that entitlement is assigned. Without it, the organisation appears as a Webex Meetings or Messaging customer, so the Calling tab and its management controls remain unavailable.

Why this answer

Managing Webex Calling services in Control Hub — configuring users, numbers, dial plans, and PSTN — requires a Webex Calling license assigned to the administrator and to the users being managed. The Webex Calling license is what entitles the organization to the calling service and unlocks the Calling configuration sections in Control Hub. Without it, the Calling tabs are unavailable or read-only.

Exam trap

CLCOR often tests the confusion between Webex service licenses (Meetings, Messaging, Calling, Room Devices), so candidates who pick Meetings or Messaging miss that only a Webex Calling license unlocks Calling management in Control Hub.

How to eliminate wrong answers

Option A is wrong because a Webex Meetings license grants meeting capabilities (scheduling, hosting) but does not entitle the user or admin to Webex Calling configuration in Control Hub. Option C is wrong because a Webex Room Device license is for registering room devices (Room Kit, Board) and does not grant calling service management rights. Option D is wrong because a Webex Messaging license covers messaging and spaces, not PSTN calling configuration.

Full explanation →

107

MCQeasy

What is the function of a 'Device Pool' in CUCM?

A.To define user permissions.

B.To group shared settings for devices.

C.To manage media resources.

D.To control call routing.

AnswerB

A Device Pool groups common settings — region, date/time group, softkey template, SRST reference and CallManager group — so these are assigned once rather than per device. This satisfies the stem's requirement by defining the function: bundling shared configuration for phones, gateways and trunks within CUCM.

Why this answer

A Device Pool in CUCM groups devices that share common settings such as region, date/time group, softkey template, SRST reference, and media resource group list. It simplifies bulk configuration by allowing administrators to apply these settings to many phones at once rather than per-device.

Exam trap

The trap here is confusing Device Pool with other grouping constructs like Line Groups, Pickup Groups, or Route Groups, which serve entirely different purposes in CUCM.

How to eliminate wrong answers

Option A is wrong because user permissions are controlled by roles and access control groups (e.g., Standard CCM Admin Users), not Device Pools. Option C is wrong because media resources (conference bridges, MTPs, transcoders) are managed via Media Resource Groups and Media Resource Group Lists, which are referenced by Device Pools but not defined by them. Option D is wrong because call routing is governed by route patterns, route lists, route groups, translation patterns, and partitions/CSS, not Device Pools.

Full explanation →

108

Multi-Selecteasy

Which THREE types of media resources can be managed by an MRGL?

Select 3 answers

A.Cisco Unity Connection.

B.Transcoder

C.SIP Trunk.

D.Conference Bridge

E.Music on Hold (MoH) Server

AnswersB, D, E

A transcoder converts between codecs, such as G.711 and G.729, so endpoints lacking a common codec can still communicate. Including transcoders in an MRGL satisfies the stem by letting Cisco Unified Communications Manager allocate hardware transcoding resources to devices in the listed region.

Why this answer

A Media Resource Group List (MRGL) in Cisco Unified Communications Manager is used to prioritize and group media resources, and the three media resource types it can manage are transcoders (B), conference bridges (D), and Music on Hold (MoH) servers (E). Transcoder (B) is correct because MRGLs assign transcoders to devices for codec conversion between different codecs (e.g., G.711 and G.729). Conference Bridge (D) is correct because MRGLs control which conference bridge resources are used for ad hoc and Meet-Me conferences.

Music on Hold (MoH) Server (E) is correct because MRGLs determine which MoH server(s) provide audio streams to callers on hold. Cisco Unity Connection (A) is a voicemail system, not a media resource managed by MRGLs, and SIP Trunk (C) is a signaling/transport interface, not a media resource type in an MRGL.

Exam trap

CLCOR often tests the specific resource types that can be included in an MRGL, and candidates may mistakenly include voicemail or SIP trunks, which are not media resources managed by MRGL.

Full explanation →

109

Multi-Selecthard

Which THREE of the following tasks are required to set up a basic SIP trunk in CUCM?

Select 3 answers

A.Create a Route Filter

B.Create a SIP Trunk Security Profile

C.Create a SIP Profile

D.Create a Translation Pattern

E.Create a SIP Trunk

AnswersB, C, E

A SIP Trunk Security Profile defines the transport and authentication parameters the trunk uses, such as TLS or TCP and digest credentials. CUCM requires one to be assigned when configuring the SIP trunk, satisfying the stem's requirement for a security profile as part of basic SIP trunk setup.

Why this answer

To set up a basic SIP trunk in CUCM, you must create a SIP Trunk Security Profile (option B), which defines the security settings such as device authentication mode, incoming/outgoing transport (TCP/UDP/TLS), and digest authentication for the trunk. You must also create a SIP Profile (option C), which defines SIP timers, normalization scripts, and header handling behavior applied to the trunk. Finally, you must create the SIP Trunk itself (option E), which binds the trunk to a device pool, calling search space, and associates the SIP Trunk Security Profile and SIP Profile.

A Route Filter (option A) is used with route lists and route groups for digit manipulation and is not mandatory for a basic SIP trunk, and a Translation Pattern (option D) is used for number translation and is likewise not required for basic SIP trunk setup.

Exam trap

CLCOR often tests whether candidates over-select optional dial-plan objects (Route Filters, Translation Patterns) as required for SIP trunk setup, when only the security profile, SIP profile, and trunk itself are mandatory.

Full explanation →

110

Multi-Selectmedium

A collaboration engineer is designing a Cisco Unified Communications Manager (CUCM) cluster for a company with 5000 users. The cluster will have one publisher and three subscribers. The engineer must ensure that the design supports high availability and that IP phones can register and make calls even if one subscriber fails. Which two statements are true regarding CUCM redundancy and device registration? (Choose two.)

Select 2 answers

A.Each IP phone can be configured with a prioritized list of up to three CUCM servers for registration using a CUCM Group.

B.The publisher must be included in the CUCM Group to ensure that phones can register if all subscribers fail.

C.The maximum number of CUCM servers in a CUCM Group is five, allowing for greater redundancy.

D.If a subscriber fails, phones registered to it will automatically re-register to another subscriber in the same CUCM Group based on priority order.

E.CUCM Groups are assigned to phones directly on each phone configuration page, not via Device Pool.

AnswersA, D

A CUCM Group can contain up to three CUCM servers in a prioritized order. When a phone boots, it attempts to register with the first server in its assigned CUCM Group. If that server is unavailable, it tries the next server in the list. This provides redundancy and ensures that phones can register even if one or more servers fail.

Why this answer

CUCM Groups provide a prioritized list of up to three CUCM servers for device registration. When a subscriber fails, phones automatically re-register to the next available server in their CUCM Group. The publisher should not be included in CUCM Groups for call processing to avoid overloading it.

CUCM Groups are assigned via Device Pools, not directly to phones.

Exam trap

The trap here is thinking that the publisher should be used for call-processing redundancy or that CUCM Groups can contain more than three servers, when in fact the limit is three and the publisher should be reserved for database and service management.

Full explanation →

111

MCQmedium

You are configuring a Webex Calling environment for a customer that requires integration with an existing on-premises PBX using a Cisco Unified Border Element (CUBE). Which gateway type should you register to Webex Control Hub to enable this connectivity?

A.Cloud Connected PSTN Gateway

B.Local Gateway

C.Webex Edge Connect Gateway

D.Virtual Gateway

AnswerB

Registering a Local Gateway in Webex Control Hub provisions a premises-based gateway that connects Webex Calling to your on-premises PBX through the CUBE, satisfying the requirement for local PSTN or PBX integration rather than cloud-delivered connectivity.

Why this answer

Local Gateway is the correct answer because in Webex Calling, a Local Gateway is an on-premises CUBE (Cisco Unified Border Element) registered to Webex Control Hub that provides connectivity between Webex Calling and an existing on-premises PBX or PSTN. It is the specific gateway type designed for premises-based PSTN or PBX integration via CUBE. Registering the CUBE as a Local Gateway in Control Hub enables call routing between Webex Calling users and the on-premises PBX.

Exam trap

CLCOR often tests the confusion between Local Gateway (on-premises CUBE for PBX/PSTN integration) and Cloud Connected PSTN (service provider-hosted PSTN), causing candidates to pick the cloud option when the scenario explicitly mentions an on-premises CUBE.

How to eliminate wrong answers

Option A is wrong because Cloud Connected PSTN Gateway refers to a service provider-hosted PSTN connection delivered through the Cisco Cloud Connected PSTN program, not an on-premises CUBE integration with a PBX. Option C is wrong because Webex Edge Connect Gateway is a dedicated network connection (via Equinix Fabric or similar) for optimizing Webex traffic, not a gateway type for PBX integration. Option D is wrong because 'Virtual Gateway' is not a valid Webex Calling gateway type in Control Hub — the valid types are Local Gateway, Cloud Connected PSTN, and Webex Calling PSTN.

Full explanation →

112

MCQeasy

A user reports that when they dial an internal extension, they hear a fast busy tone. The administrator checks the CUCM configuration and finds that the user's phone is registered and the dialed extension exists. Which CUCM component is responsible for determining that the dialed digits are a valid internal extension and routing the call?

A.Cisco CTIManager service

B.Cisco TFTP service

C.Cisco IP Voice Media Streaming App service

D.CallManager service

AnswerD

The Cisco CallManager service performs digit analysis and call routing. It matches dialed digits against configured route patterns, translation patterns, and directory numbers. A fast busy tone often indicates that digit analysis failed to find a matching pattern or that the call was rejected. The CallManager service is the core call control engine responsible for this decision.

Why this answer

The Cisco CallManager service is the call control application that performs digit analysis and determines how to route calls. When a user dials an extension, the CallManager service checks for matching patterns and directory numbers. If no match is found or the call is rejected, the user hears a fast busy tone.

Thus, the CallManager service is the correct component.

Exam trap

The trap here is confusing call control services with media or configuration services, leading to checking TFTP or CTIManager instead of the CallManager service for routing failures.

Full explanation →

113

MCQmedium

You are deploying Webex Calling with Local Gateway. Which component must be configured to allow PSTN connectivity for users who have migrated from on-premises CUCM to the cloud?

A.Webex Edge Connect

B.SIP Trunk via Local Gateway

C.Webex Calling Cloud Connected PSTN

D.Unified Border Element (CUBE) API

AnswerB

The Local Gateway registers to Webex Calling and connects to the PSTN through a SIP trunk, providing the required voice path for migrated users. Without that trunk configuration, the gateway has no carrier connectivity, so cloud-registered endpoints cannot place or receive external calls.

Why this answer

A SIP Trunk via Local Gateway must be configured because the Local Gateway (LGW) is the on-premises component that connects Webex Calling to the PSTN through a local SIP trunk, allowing migrated users to make and receive external calls. The LGW registers with Webex Calling and bridges cloud-registered endpoints to the on-premises PSTN provider. This is the required component for PSTN connectivity when using Local Gateway.

Exam trap

CLCOR often tests the distinction between cloud-connected PSTN (provider handles PSTN in the cloud) and Local Gateway (on-premises SIP trunk to local PSTN), so candidates pick the cloud option when the scenario specifies Local Gateway.

How to eliminate wrong answers

Option A is wrong because Webex Edge Connect is a dedicated private network connection between Webex and the customer's data center for quality and security, not the PSTN connectivity mechanism itself. Option C is wrong because Webex Calling Cloud Connected PSTN is a cloud-based PSTN option where Cisco or a partner provides the PSTN connection directly in the cloud — it does not use the on-premises Local Gateway. Option D is wrong because Unified Border Element (CUBE) API is not a Webex Calling PSTN connectivity component; CUBE is a Cisco IOS XE session border controller used in on-premises SIP trunking, and the API reference is a distractor.

Full explanation →

114

MCQmedium

If a Translation Pattern is configured in CUCM, at what stage of the call processing does the digit manipulation occur?

A.After the call is sent to the gateway.

B.At the device level during registration.

C.Only when the call fails to route.

D.Before the Route Pattern is matched.

AnswerD

CUCM applies Translation Pattern digit manipulation during digit analysis, before the transformed digits are matched against a Route Pattern. This ordering lets the translation rewrite the called number so the subsequent route lookup uses the altered digits.

Why this answer

Translation patterns in CUCM perform digit manipulation before the Route Pattern is matched. This allows the translation pattern to modify the dialed digits so that they match the appropriate route pattern, enabling correct routing. The manipulation occurs during the digit analysis phase, prior to route selection.

Exam trap

CLCOR often tests the stage at which translation patterns apply, with candidates mistakenly thinking they occur after route pattern matching or only on failure, when they actually occur before route pattern matching.

How to eliminate wrong answers

Option A is wrong because digit manipulation occurs before the call is sent to the gateway, not after; the gateway receives the already-translated digits. Option B is wrong because translation patterns are not applied at device registration; they are applied during call routing. Option C is wrong because translation patterns are applied regardless of whether the call fails to route; they are part of the normal call processing to ensure proper routing.

Full explanation →

115

MCQmedium

A Cisco CUBE is deployed to connect a Cisco Unified Communications Manager (CUCM) cluster to a service provider SIP trunk. The service provider requires that all outgoing SIP INVITE messages contain a specific custom header, X-Provider-ID, with a value of 'CUBE123'. Which CUBE configuration is required to insert this header into outgoing INVITEs?

A.voice class sip-profiles

B.sip-header insert

C.header-manipulation

D.sip-header profile

AnswerA

The voice class sip-profiles command is used to create a SIP profile that can manipulate SIP headers. Within the profile, you can use rules to add, modify, or remove headers. To insert a custom header like X-Provider-ID: CUBE123 in outgoing INVITEs, you define a rule with the request INVITE and header insertion. This is the correct method for header manipulation on CUBE.

Why this answer

To insert a custom SIP header into outgoing INVITEs, you must configure a SIP profile using voice class sip-profiles. Inside the profile, a rule with the request INVITE and header insertion syntax adds the X-Provider-ID header with the value CUBE123. This profile is then applied to the outbound dial peer toward the service provider.

Exam trap

The trap here is assuming that any command with 'sip-header' or 'header' will work, but the correct construct is a voice class sip-profiles with header rules.

Full explanation →

116

MCQhard

An administrator needs to restrict a group of phones from making long-distance calls. Which component of the CUCM dial plan should be modified to achieve this while maintaining internal dialing?

A.SIP Trunk

B.Calling Search Space

C.Region

D.Device Pool

AnswerB

Calling search spaces control which partitions a phone may reach, so assigning a restricted CSS blocks long-distance patterns while internal dialling remains available through the partitions it still permits. This satisfies the stem's constraint of preserving internal calls.

Why this answer

The Calling Search Space (CSS) is the CUCM dial plan component that defines which partitions a device or line can access. By assigning a CSS that excludes the long-distance partition (while including internal partitions), the administrator restricts long-distance calls while preserving internal dialing. This is the standard method for class of service restrictions.

Exam trap

CLCOR often tests the confusion between CSS (call permissions/partitions) and Device Pool (device settings) — candidates pick Device Pool because it groups phones, but it does not control dialing permissions.

How to eliminate wrong answers

Option A is wrong because a SIP Trunk is a network connection to an external system, not a dial plan restriction mechanism. Option C is wrong because a Region controls codec selection between locations, not call permissions. Option D is wrong because a Device Pool groups devices for settings like region, date/time, and SRST, but does not itself control dialing permissions.

Full explanation →

117

MCQmedium

A collaboration engineer is configuring a Webex Calling PSTN option where the enterprise wants to keep its existing Cisco Unified Border Element (CUBE) and SIP trunk contract with the incumbent carrier, while still using Webex Calling for call control and user features. Which Webex Calling PSTN connection type should the engineer select in Webex Control Hub?

A.Webex Edge for Calling

B.Cloud Connected PSTN (CCP)

C.Local Gateway

D.Cisco PSTN

AnswerC

Local Gateway is the Webex Calling PSTN option that registers an on-premises Cisco Unified Border Element (CUBE) or IOS-XE gateway to Webex Calling, allowing the enterprise to keep its existing SIP trunk contract and carrier while Webex Calling handles call control and features. This matches the scenario exactly, because the CUBE remains the PSTN demarcation point and only signaling is extended to the Webex cloud.

Why this answer

Local Gateway is designed for organizations that want to retain their own on-premises PSTN connectivity, including the existing CUBE and carrier SIP trunk, while adopting Webex Calling for users and features. The CUBE registers to Webex Calling and acts as the local PSTN gateway, so the enterprise keeps its carrier contract and gateway investment without deploying a cloud-connected provider or Cisco PSTN service.

Exam trap

The trap here is assuming that any Webex Calling PSTN option that involves a carrier automatically preserves the existing on-premises CUBE and SIP trunk contract, when only Local Gateway keeps that on-premises gateway in the call path.

Full explanation →

118

MCQhard

A collaboration engineer is designing a SIP trunk between Cisco Unified Communications Manager (CUCM) and Cisco Unified Border Element (CUBE). The engineer must ensure that calls are routed correctly and that CUCM can send calls to CUBE. Which element should be configured on CUCM to point to CUBE as the next hop for calls destined to the ITSP?

A.SIP trunk configured on CUCM with destination address of CUBE

B.Translation pattern that rewrites the called number to CUBE's IP address

C.H.323 gateway configured on CUCM with destination address of CUBE

D.Route group with CUBE as the only member

AnswerA

A SIP trunk on CUCM is used to route calls to an external device such as CUBE. The trunk is configured with the destination address (IP address or DNS name) of CUBE's signaling interface. When a call matches a route pattern that points to this SIP trunk, CUCM sends the SIP INVITE to CUBE, which then forwards it to the ITSP. This is the correct way to integrate CUCM with CUBE.

Why this answer

To route calls from CUCM to CUBE, you configure a SIP trunk on CUCM with the destination address of CUBE. This trunk is then referenced by a route group, route list, and route pattern. When a call matches the route pattern, CUCM sends the call to CUBE via the SIP trunk.

The trunk itself is the element that points to CUBE as the next hop.

Exam trap

The trap here is confusing the role of a route group or translation pattern with the actual trunk that specifies the destination address.

Full explanation →

119

MCQhard

You are configuring a Cisco IOS router to perform traffic shaping for voice traffic. Why is traffic shaping often used in conjunction with queuing?

A.To prioritize voice over data traffic

B.To reduce the CPU utilization of the router

C.To prevent interface congestion by smoothing out traffic bursts

D.To increase the total throughput of the interface

AnswerC

Shaping smooths bursts by buffering excess packets and releasing them at the configured rate, so the interface queue is not overwhelmed by sudden spikes. This directly satisfies the stem's voice-traffic constraint: steady pacing keeps delay and jitter low, letting queuing mechanisms schedule voice reliably.

Why this answer

Traffic shaping is used with queuing to prevent interface congestion by smoothing out traffic bursts — it buffers excess traffic and releases it at a configured rate (e.g., the CIR), ensuring the interface does not exceed the contracted bandwidth. Queuing then schedules the shaped traffic according to class priorities, so shaping and queuing work together to manage congestion and protect voice quality.

Exam trap

CLCOR often tests the confusion between shaping (smooths bursts, buffers excess) and policing (drops excess) and between shaping and queuing (rate control vs. priority scheduling) — candidates must remember that shaping prevents congestion by rate-limiting, while queuing prioritizes within that rate.

How to eliminate wrong answers

Option A is wrong because prioritizing voice over data is the role of queuing (e.g., LLQ), not shaping — shaping controls the rate, not the priority. Option B is wrong because traffic shaping actually adds processing overhead (buffering, scheduling) and does not reduce CPU utilization. Option D is wrong because shaping deliberately limits throughput to the shaped rate; it does not increase total interface throughput.

Full explanation →

120

MCQmedium

Which service allows Webex users to see the presence status of an on-premises Jabber user?

A.Hybrid Presence Service.

B.Hybrid Calendar Service.

C.Webex Calling.

D.Directory Connector.

AnswerA

Hybrid Presence Service federates presence between Webex and on-premises Jabber deployments, allowing Webex users to view the presence status of Jabber users registered on-premises. It is the specific service providing this cross-deployment presence visibility, unlike messaging or directory connectors.

Why this answer

Hybrid Presence Service (also known as Webex Hybrid Presence) allows Webex users to see the presence status of on-premises Jabber users by connecting the on-premises IM and Presence Service to Webex. The service synchronizes presence between the on-premises Cisco IM and Presence cluster and Webex, so a Webex user can see whether a Jabber user is available, busy, or away. This is the specific service designed for cross-presence visibility.

Exam trap

CLCOR often tests the distinction between Hybrid Services — candidates confuse Hybrid Calendar (meetings), Directory Connector (user sync), and Hybrid Presence (status) — and pick the wrong service for presence visibility.

How to eliminate wrong answers

Option B is wrong because Hybrid Calendar Service connects calendars (Microsoft 365 or Google) to Webex for meeting scheduling and Join buttons — it does not handle presence. Option C is wrong because Webex Calling is a cloud calling service for PSTN and internal calls, not a presence federation service. Option D is wrong because Directory Connector synchronizes user directories between on-premises and Webex, but it does not provide presence status — it only ensures users exist in both systems.

Full explanation →

121

MCQmedium

Which tool can be used to perform a bulk import of users and devices into CUCM?

A.Cisco Unified Reporting

B.Bulk Administration Tool (BAT)

C.Dialed Number Analyzer

D.Cisco Prime Collaboration

AnswerB

Bulk Administration Tool inserts users and devices into CUCM via CSV-based batched jobs, handling large volumes that manual entry cannot. It directly satisfies the stem's bulk import constraint, unlike the GUI's one-at-a-time provisioning or AXL scripting.

Why this answer

The Bulk Administration Tool (BAT) is specifically designed for bulk operations in Cisco Unified Communications Manager (CUCM), including importing users and devices. It allows administrators to add, update, or delete large numbers of records using CSV files, and it validates the data before insertion. BAT is the only tool among the options that provides this bulk import capability natively within CUCM.

Exam trap

CLCOR often tests the distinction between tools for bulk provisioning versus reporting or troubleshooting, so candidates might confuse Cisco Prime Collaboration's broader management capabilities with CUCM's native BAT for bulk import.

How to eliminate wrong answers

Option A is wrong because Cisco Unified Reporting is used to generate and view reports on CUCM system data, not to import users or devices. Option C is wrong because Dialed Number Analyzer is a troubleshooting tool that simulates call routing to test dial plans, not for bulk provisioning. Option D is wrong because Cisco Prime Collaboration is a separate management application for unified communications that can provide provisioning and monitoring across multiple systems, but it is not the native CUCM tool for bulk import; the question specifically asks for a tool to perform bulk import into CUCM, which is BAT.

Full explanation →

122

MCQeasy

When designing an NTP deployment for a Cisco Collaboration environment, what is the best practice for time synchronization across CUCM nodes?

A.Point all CUCM nodes to the same external NTP stratum 1 or 2 servers.

B.Configure NTP only on the publisher node.

C.Point each CUCM node to a different NTP server for redundancy.

D.Use the publisher as the NTP server for all subscriber nodes.

AnswerA

Pointing every CUCM node at the same external stratum 1 or 2 sources keeps cluster time consistent and avoids CUCM acting as an NTP server for itself. Uniform external reference prevents drift between nodes that would disrupt CDR timestamps and database replication.

Why this answer

Cisco Collaboration best practice requires every CUCM node (publisher and subscribers) to synchronize directly with the same external, reliable NTP source — typically stratum 1 or 2 servers. This ensures consistent, accurate time across all nodes, which is critical for database replication, CDR timestamps, certificate validation, and log correlation. Pointing all nodes to the same external source avoids a single point of failure and prevents time drift that could break cluster operations.

Exam trap

CLCOR often tests the misconception that the CUCM publisher should act as the NTP server for subscribers, confusing the publisher's role in database replication with time synchronization.

How to eliminate wrong answers

Option B is wrong because configuring NTP only on the publisher leaves subscribers unsynchronized, causing time drift that breaks database replication and certificate checks. Option C is wrong because pointing each node to a different NTP server introduces inconsistent time sources, leading to clock skew across the cluster. Option D is wrong because using the publisher as an NTP server for subscribers creates a single point of failure and is not a Cisco-supported best practice; if the publisher goes down, subscribers lose time synchronization.

Full explanation →

123

MCQeasy

Which menu path in CUCM Administration is used to configure the 'Cisco Unified Communications Manager Group'?

A.Serviceability -> CM Group

B.System -> Cisco Unified Communications Manager Group

C.Device -> Device Settings -> CM Group

D.Advanced Features -> CM Group

AnswerB

CUCM Administration groups all cluster-level service and node settings under the System menu, so the Cisco Unified Communications Manager Group list, which defines ordered call-processing redundancy, is reached via System -> Cisco Unified Communications Manager Group.

Why this answer

The Cisco Unified Communications Manager Group is configured under System -> Cisco Unified Communications Manager Group in CUCM Administration. This menu path is where you define ordered lists of CUCM servers (groups) that devices use for call processing redundancy. The group is then assigned to device pools, which are applied to endpoints.

The other menu paths do not contain this configuration option.

Exam trap

CLCOR often tests the exact menu path for configuring CM Groups, and candidates may confuse it with device pool configuration or serviceability menus.

How to eliminate wrong answers

Option A is wrong because Serviceability is used for monitoring and troubleshooting tools like traces, logs, and service activation, not for configuring CM Groups. Option C is wrong because Device -> Device Settings contains items like device pools, phone button templates, and softkey templates, but not CM Groups. Option D is wrong because Advanced Features includes items like presence, QoS, and SIP normalization, not CM Groups.

Full explanation →

124

MCQeasy

A network engineer is configuring a Cisco IOS router to classify traffic for a QoS policy. The engineer needs to identify packets based on the DSCP value EF that has already been marked by an upstream IP phone. Which configuration element should be used to match this traffic?

A.A route-map with the match ip dscp ef command applied to the interface.

B.A policy-map with the set dscp ef command.

C.A class-map with the match dscp ef command.

D.An access list applied with the ip access-group command on the interface.

AnswerC

A class-map defines the traffic classification criteria, and match dscp ef identifies packets already marked with the Expedited Forwarding DSCP value. This is exactly the tool used to select voice bearer traffic that an upstream phone or switch has marked, so the policy-map can then apply the desired treatment to that class.

Why this answer

Classification for QoS is performed with class-maps, and matching an existing DSCP marking uses match dscp ef. Since the IP phone has already marked voice bearer as EF, the router only needs to identify those packets so the policy-map can apply priority or bandwidth treatment. Policy-maps define actions, not matching criteria.

Exam trap

The trap here is mixing up the role of class-maps, which classify, with policy-maps, which act on classified traffic.

Full explanation →

125

MCQmedium

A collaboration engineer is designing a Webex Hybrid Services deployment for an organization that requires all media to be processed on-premises while still using Webex Control Hub for user management and analytics. The organization has an existing Expressway-C and Expressway-E pair. Which Webex Hybrid service must be deployed to meet the requirement that all call signaling and media remain on-premises?

A.Webex Hybrid Data Security

B.Webex Hybrid Message Service

C.Webex Hybrid Call Service

D.Webex Hybrid Calendar Service

AnswerC

Hybrid Call Service keeps call signaling and media on-premises by using Expressway-C and Expressway-E to connect Webex App to the on-premises CUCM. It provides a cloud-based user experience while preserving on-premises call control and media handling, which directly satisfies the requirement that all media be processed locally.

Why this answer

Webex Hybrid Call Service is designed to connect Webex App users to on-premises CUCM for call control while keeping media on-premises through Expressway-C and Expressway-E. It enables a hybrid experience where users get cloud-based convenience but calls are processed locally. The other hybrid services address calendaring, messaging, or data security, not call signaling and media processing.

Exam trap

The trap here is confusing Hybrid Call Service with Hybrid Message Service or Hybrid Calendar Service, assuming that any hybrid service keeps all communication on-premises.

Full explanation →

126

MCQeasy

Which component in a Cisco Collaboration design is responsible for providing the dial tone and call routing logic for IP phones?

A.Cisco Expressway

B.Cisco Unity Connection

C.Cisco CUBE

D.Cisco Unified Communications Manager

AnswerD

Cisco Unified Communications Manager provides call processing, generating dial tone and executing call routing logic for registered IP phones. It is the call-control core, distinct from gateways and endpoints that merely carry or originate media.

Why this answer

Cisco Unified Communications Manager (CUCM) is the call-processing component that provides dial tone and call routing logic for IP phones in a Cisco Collaboration design. It handles call setup, routing, and features for endpoints.

Exam trap

CLCOR often tests the functions of different collaboration components, and candidates may confuse CUCM with CUBE or Expressway, which serve different purposes in the architecture.

How to eliminate wrong answers

Option A is wrong because Cisco Expressway provides firewall traversal and mobile remote access, not dial tone or call routing for internal IP phones. Option B is wrong because Cisco Unity Connection is a voicemail and unified messaging system, not a call-processing server. Option C is wrong because Cisco CUBE (Cisco Unified Border Element) is a session border controller for SIP trunking and interconnecting networks, not for providing dial tone to IP phones.

Full explanation →

127

MCQeasy

When configuring an IP phone for automatic provisioning, which DHCP option is most commonly used to provide the TFTP server address to the phone?

A.Option 150

B.Option 43

C.Option 66

D.Option 12

AnswerA

Option 150 returns the TFTP server address as a single IP in the vendor-specific field, which Cisco IP phones read directly for provisioning. Option 66 carries it as a hostname string, which phones do not reliably parse.

Why this answer

DHCP Option 150 is the Cisco-proprietary option most commonly used to deliver a list of TFTP server IP addresses to Cisco IP phones during auto-provisioning. The phone uses this TFTP address to download its configuration file (SEP<MAC>.cnf.xml) and firmware.

Exam trap

CLCOR often tests the confusion between Option 150 (Cisco TFTP IP list) and Option 66 (generic single TFTP server name), causing candidates to pick the more 'standard' option.

How to eliminate wrong answers

Option B is wrong because Option 43 is used for vendor-specific information (commonly for wireless access point controllers and some vendor encapsulation), not the standard Cisco phone TFTP delivery. Option C is wrong because Option 66 provides a single TFTP server name (often a hostname/FQDN) and is more common in non-Cisco VoIP environments; Cisco phones prefer Option 150's IP list. Option D is wrong because Option 12 supplies the client hostname, not a TFTP server address.

Full explanation →

128

MCQmedium

A collaboration engineer is designing a Webex Calling deployment for a company that requires all PSTN calls to egress through their existing on-premises Cisco Unified Border Element (CUBE) with a PRI circuit. The company wants to keep the PSTN gateway on-premises and use Webex Calling for internal and external calling. Which deployment option should the engineer choose?

A.Webex Calling with Cisco PSTN

B.Cloud Connected PSTN (CCP)

C.Local Gateway

D.Webex Edge Audio

AnswerC

Local Gateway is the correct choice because it allows Webex Calling to route PSTN calls through an on-premises CUBE or gateway, leveraging the existing PRI circuit. This option is designed for customers who want to retain their on-premises PSTN connectivity while using Webex Calling for call control. It supports the required egress through the local CUBE.

Why this answer

Local Gateway is the correct deployment option because it enables Webex Calling to route PSTN calls through an on-premises Cisco Unified Border Element (CUBE) or gateway, preserving the existing PRI circuit. This option is specifically designed for organizations that want to retain their on-premises PSTN connectivity while adopting Webex Calling for cloud-based call control and features.

Exam trap

The trap here is confusing Local Gateway with Cloud Connected PSTN, assuming that any PSTN option allows on-premises egress when only Local Gateway does.

Full explanation →

129

Multi-Selecthard

Which THREE of the following are valid reasons to use a Media Termination Point (MTP)?

Select 3 answers

A.Providing transcoding for G.711 to G.729

B.Bridging different signaling protocols

C.Enabling media anchoring for specific features

D.Increasing voice bandwidth

E.DTMF interworking between SIP and H.323

AnswersB, C, E

MTPs can act as a bridge for protocol-specific signaling needs.

Why this answer

Option B is correct because an MTP can bridge different signaling protocols by terminating the signaling and media on each side, allowing calls to interconnect between otherwise incompatible protocol variants. Option C is correct because MTPs enable media anchoring, which is required for features such as call recording, lawful intercept, and certain supplementary services that need the media stream to pass through a controlled point. Option E is correct because MTPs provide DTMF interworking, converting out-of-band DTMF (for example, SIP INFO or H.323) into in-band or RFC 2833 events so that endpoints using different DTMF transport methods can communicate.

Option A is not correct because transcoding between G.711 and G.729 is the function of a transcoder (hardware or software DSP resource), not an MTP. Option D is not correct because an MTP does not increase voice bandwidth; it terminates and relays media, and any bandwidth impact would typically be neutral or negative, not an increase.

Exam trap

CLCOR often tests the confusion between MTP and transcoder functions — candidates incorrectly select transcoding as an MTP role, but MTPs only relay media, they do not change codecs.

Full explanation →

130

MCQhard

You are troubleshooting a one-way audio issue. The phone can register, but no audio is heard. What should you check first?

A.The phone's directory number.

B.The CSS of the device.

C.The phone's firmware version.

D.Network ACLs and port accessibility for RTP.

AnswerD

Signalling succeeded, so registration and SIP are working; one-way audio points to RTP being blocked in one direction. ACLs and firewall rules filtering the RTP port range, or asymmetric routing, prevent media delivery even though call setup completes.

Why this answer

One-way audio in VoIP is almost always a media path issue, not a signaling issue. Since the phone registers successfully, SIP signaling is working, but RTP (Real-time Transport Protocol) packets are not traversing the network. The first thing to check is whether network ACLs, firewalls, or NAT devices are blocking the UDP port range used for RTP (typically 16384–32767 on Cisco Unified Communications Manager).

Exam trap

CLCOR often tests the misconception that one-way audio is a signaling or configuration issue (CSS, DN, firmware) when it is almost always a network ACL/firewall/NAT problem affecting RTP media.

How to eliminate wrong answers

Option A is wrong because an incorrect directory number would affect call routing and dialing, not audio once a call is established. Option B is wrong because the CSS (Calling Search Space) controls call permissions and routing, not media flow; a CSS issue would prevent calls from being placed, not cause one-way audio. Option C is wrong because firmware version affects phone features and stability, but a registered phone with one-way audio points to network media path problems, not firmware.

Full explanation →

131

MCQmedium

Which feature in CUCM is used to block calls when the available bandwidth in a location is exceeded?

A.Gatekeeper

B.Bandwidth Manager

C.Locations CAC

D.RSVP

AnswerC

Locations call admission control totals the bandwidth allocated to phones at a location and rejects new calls once that limit is exceeded, blocking them before they consume WAN capacity. This directly satisfies the stem's requirement to block calls when location bandwidth is exhausted.

Why this answer

Locations CAC (Call Admission Control) is the correct answer because CUCM uses the Locations configuration to define the maximum bandwidth available between locations and to deny new calls when that bandwidth is exhausted. When a call would exceed the configured audio or video bandwidth for a location pair, CUCM rejects the call with a reorder tone or redirects it to a secondary path if configured. This is the native, built-in CAC mechanism in CUCM.

Exam trap

CLCOR often tests the confusion between CUCM Locations CAC (for SCCP/SIP endpoints registered to CUCM) and gatekeeper CAC (for H.323 zones) or RSVP (network-layer reservation), causing candidates to pick the wrong CAC mechanism.

How to eliminate wrong answers

Option A is wrong because a gatekeeper is an H.323 device that provides CAC for H.323-controlled zones, not the CUCM Locations-based mechanism used for SCCP/SIP endpoints registered directly to CUCM. Option B is wrong because 'Bandwidth Manager' is not a CUCM feature — it is a generic term and not the specific CUCM configuration element that enforces location-based CAC. Option D is wrong because RSVP is a resource reservation protocol that provides CAC at the network layer by reserving bandwidth along the path; while CUCM can be configured to use RSVP, the question asks about the feature that blocks calls when location bandwidth is exceeded, which is Locations CAC.

Full explanation →

132

MCQmedium

When deploying Webex Hybrid Services, what is the purpose of the Expressway-C cluster?

A.To act as a firewall for all enterprise traffic.

B.To store all user messages on-premises.

C.To host the connectors that communicate with the Webex cloud.

D.To manage public DNS records for the organization.

AnswerC

The Expressway-C cluster sits inside the enterprise network and hosts the connectors (such as Calendar, Management, and Call Connector) that establish outbound connections to the Webex cloud, satisfying the hybrid services requirement for secure connector hosting.

Why this answer

The Expressway-C (Concentrator) cluster sits inside the enterprise network and hosts the various connectors (e.g., Calendar Connector, Call Connector, Management Connector, and the edge traversal pair) that establish secure outbound connections to the Webex cloud. It acts as the trusted on-premises endpoint that brokers hybrid service traffic between internal resources like CUCM, Exchange, and the Webex cloud.

Exam trap

CLCOR often tests the confusion between Expressway-C and Expressway-E roles, where candidates incorrectly assume Expressway-C is the internet-facing edge or a firewall component rather than the internal connector host.

How to eliminate wrong answers

Option A is wrong because Expressway-C is not a firewall; it is a traversal/connector server that works with Expressway-E to traverse the DMZ, and firewalls remain separate security appliances. Option B is wrong because user messages are stored in the Webex cloud (or on the messaging platform), not on the Expressway-C cluster, which handles signaling/connector functions rather than message storage. Option D is wrong because public DNS records are managed by the organization's DNS provider or Cisco's cloud infrastructure, not by Expressway-C, which handles internal-to-cloud connector communication.

Full explanation →

133

MCQhard

You have implemented 'Local Route Group' in your dial plan. What is the main advantage of using this instead of static Route Groups?

A.It forces the use of G.729.

B.It requires less CPU on the CUCM.

C.It simplifies the dial plan by removing the need for location-specific route patterns.

D.It allows faster call setup.

AnswerC

Local Route Group assigns a route group per device pool, so a single route pattern references a logical group that resolves to the correct gateway at each location. This removes the need for location-specific route patterns, satisfying the simplification stated in the stem.

Why this answer

Local Route Groups allow a single set of route patterns to be used across multiple locations by dynamically selecting the appropriate gateway based on the device's location. This eliminates the need to create duplicate route patterns for each location, simplifying the dial plan and reducing administrative overhead.

Exam trap

The trap is assuming that Local Route Groups affect codec or performance; they are purely a dial plan simplification tool, and the exam may distract with technical-sounding but irrelevant options.

How to eliminate wrong answers

Option A is wrong because Local Route Groups do not force the use of G.729; codec selection is independent of route group configuration. Option B is wrong because Local Route Groups do not inherently reduce CPU usage on CUCM; they are a logical grouping mechanism, and CPU impact is negligible either way. Option D is wrong because Local Route Groups do not directly affect call setup speed; they simplify configuration but do not change the signaling performance.

Full explanation →

134

Multi-Selecteasy

Which TWO of the following are common audio codecs used in Cisco collaboration?

Select 2 answers

A.JPEG

B.G.711

C.G.729

D.H.264

E.MP3

AnswersB, C

G.711 is a narrowband waveform codec using pulse-code modulation at 64 kbps, delivering toll quality with minimal processing delay. It is the baseline codec in Cisco collaboration, satisfying the requirement for a common audio codec used across Cisco endpoints and gateways.

Why this answer

G.711 (option B) is a correct answer because it is a standard ITU-T narrowband audio codec (64 kbps PCM, using either A-law or mu-law companding) that is universally supported in Cisco collaboration endpoints and gateways for toll-quality voice. G.729 (option C) is also correct because it is a widely used ITU-T compressed audio codec (8 kbps CS-ACELP) commonly deployed in Cisco voice and collaboration environments to conserve bandwidth on WAN links. Option A (JPEG) is incorrect because it is a still-image compression standard, not an audio codec.

Option D (H.264) is incorrect because it is a video compression standard used for video streams, not audio. Option E (MP3) is incorrect because, although it is an audio codec, it is a consumer music format and is not a standard codec used in Cisco collaboration call signaling/media negotiation.

Exam trap

CLCOR often tests the confusion between audio and video codecs, causing candidates to select H.264 or JPEG (video/image codecs) when asked about audio codecs like G.711 and G.729.

Full explanation →

135

MCQmedium

You are implementing a 'Call Hunt' sequence. You want to ensure that if no agent answers within 20 seconds, the call moves to the next member in the list. Which configuration object manages this timer?

A.Line Group

B.Hunt List

C.Route Group

D.Hunt Pilot

AnswerA

The Line Group object holds the hunt-scheme timer settings, including the no-answer timeout that advances the call to the next member. Configuring the 20-second interval there satisfies the stem's requirement to move the call onward.

Why this answer

In CUCM Call Hunt (a line group hunt), the Line Group configuration object contains the 'Hunt Options' settings, including the 'No Answer' timer (default 12 seconds, configurable up to 20+ seconds) that determines how long a member rings before the call moves to the next member. The Line Group also defines the distribution algorithm (top-down, longest idle, circular, broadcast) and the member order.

Exam trap

CLCOR often tests the hierarchy confusion between Hunt Pilot, Hunt List, and Line Group, so candidates who assume the timer lives on the Hunt Pilot or Hunt List (the more visible objects) pick the wrong answer.

How to eliminate wrong answers

Option B is wrong because the Hunt List is a container that references one or more Line Groups and defines the order in which line groups are tried, but it does not hold the per-member no-answer timer. Option C is wrong because a Route Group is a dial-plan object for outbound PSTN routing via gateways/trunks, unrelated to hunt timers. Option D is wrong because the Hunt Pilot is the dialable number that maps to a Hunt List; it defines the pilot number, voicemail settings, and final forwarding, but not the per-line no-answer timer.

Full explanation →

136

MCQhard

An administrator needs to enforce a bandwidth limit for video calls at a specific site in CUCM. Which configuration object should be used?

A.Region

B.Location

C.Partition

D.Device Pool

AnswerB

Locations define the maximum audio and video bandwidth allocated between sites or to a device pool. Applying the limit at the Location object enforces call admission control for video, which per-device or region settings cannot achieve.

Why this answer

In Cisco Unified CM, the Location object is the configuration element that enforces call admission control (CAC) by defining the maximum audio and video bandwidth available between locations. When a video call is placed, CUCM subtracts the call's bandwidth from the Location's video bandwidth pool, and if the limit is exceeded, the call is denied or rerouted. This makes Location the correct object for enforcing a per-site bandwidth limit for video calls.

Exam trap

CLCOR often tests the confusion between Region (codec selection) and Location (bandwidth enforcement) — candidates frequently pick Region thinking it controls bandwidth because both are used in call admission.

How to eliminate wrong answers

Option A is wrong because a Region defines which codec is used between groups of devices (codec preference lists) and does not itself enforce a bandwidth cap. Option C is wrong because a Partition is a logical grouping of directory numbers/route patterns used for calling search space dial-plan control, not bandwidth management. Option D is wrong because a Device Pool groups devices for common settings like date/time, region, and SRST reference, but it does not enforce bandwidth limits.

Full explanation →

137

Multi-Selectmedium

Which TWO of the following are true regarding the use of voice translation-rules?

Select 2 answers

A.They perform load balancing

B.They use regular expressions

C.They replace the SIP protocol

D.They can modify calling numbers

E.They define the codec

AnswersB, D

Voice translation-rules match dialled digits using regular expressions, the same pattern syntax applied by Cisco IOS voice translation profiles. This satisfies the stem's requirement that the statement be true: regex patterns define the match and replace clauses that rewrite calling or called numbers before routing, so the claim is accurate.

Why this answer

Voice translation-rules on Cisco voice gateways are built on regular-expression matching, so option B is correct: each rule uses a regex pattern to match the called or calling number and a replacement string to rewrite it. Option D is also correct because translation-rules can manipulate the calling number (the ANI) as well as the called number, for example to strip digits, prepend a prefix, or change the numbering plan before the call is routed. They do not perform load balancing, which is handled by dial-peer hunting or gateway redundancy mechanisms, so A is wrong.

They do not replace SIP, which is a signaling protocol independent of number translation, so C is wrong. They do not define codecs, which are configured on dial-peers via the codec command, so E is wrong.

Exam trap

CLCOR often tests the misconception that translation rules perform routing or load balancing; candidates confuse them with route patterns/route groups, which actually select the destination.

Full explanation →

138

MCQhard

Which statement best describes the difference between policing and shaping?

A.Policing drops traffic, whereas shaping buffers it

B.Policing buffers traffic, whereas shaping drops it

C.Both use buffers to manage congestion

D.Both are applied only on ingress interfaces

AnswerA

Policing enforces a rate limit by discarding excess packets immediately, causing retransmissions and TCP throughput collapse. Shaping instead queues and delays outbound traffic to smooth bursts, avoiding drops. The stem's contrast hinges on this drop-versus-buffer mechanism, making the statement accurate.

Why this answer

Policing and shaping both measure traffic against a configured rate, but they react differently when the rate is exceeded. Policing simply drops (or remarks) packets that exceed the committed rate, making it a hard enforcement mechanism typically applied on ingress. Shaping, on the other hand, buffers excess packets and delays their transmission so they are sent later within the configured rate, smoothing bursts rather than discarding them.

Exam trap

CLCOR often tests the buffer-versus-drop distinction between policing and shaping, and the most common error is reversing the two behaviors or assuming both buffer traffic.

How to eliminate wrong answers

Option B is wrong because it reverses the definitions — policing does not buffer traffic and shaping does not drop it; that inversion is the classic distractor. Option C is wrong because only shaping uses buffers (a shaping queue); policing has no buffering mechanism and instead drops or remarks excess packets. Option D is wrong because policing is commonly applied on ingress (inbound) interfaces, while shaping is typically applied on egress (outbound) interfaces — they are not both ingress-only.

Full explanation →

139

MCQeasy

Which command is used to configure the IP address or hostname of the destination device on a dial-peer?

A.session target

B.target ip

C.call target

D.destination-ip

AnswerA

`session target` defines the destination endpoint for a dial-peer, accepting either an IP address (for VoIP) or a hostname, which directly satisfies the stem's requirement to configure the destination device's address. It is the specific command that binds a dial-peer to its remote peer.

Why this answer

The 'session target' command is used on a dial-peer to specify the destination IP address or hostname of the next-hop device for that dial-peer. It is a fundamental command in Cisco IOS voice gateway configuration for directing calls. The other options are not valid Cisco IOS commands for this purpose.

Exam trap

CLCOR often tests the exact syntax of dial-peer commands; candidates confuse 'session target' with similar-sounding but invalid commands like 'target ip' or 'destination-ip'.

How to eliminate wrong answers

Option B is wrong because 'target ip' is not a valid Cisco IOS command; the correct syntax is 'session target ipv4:<address>'. Option C is wrong because 'call target' is not a recognized command in dial-peer configuration. Option D is wrong because 'destination-ip' is not a valid command; the destination is specified via 'session target'.

Full explanation →

140

Multi-Selectmedium

Which TWO of the following fields are required when defining a Route Pattern?

Select 2 answers

A.Gateway/Route List

B.Codec

C.Bandwidth Limit

D.Softkey Template

E.Route Pattern String

AnswersA, E

The gateway or route list defines where calls matching the pattern are sent, so it is mandatory for every route pattern. Without it, Cisco Unified Communications Manager cannot route the dialled digits to a destination, making the field required.

Why this answer

In Cisco Unified Communications Manager, a Route Pattern must specify the destination to match and where to send the matched calls, so option A (Gateway/Route List) is required to define the egress path for the pattern, and option E (Route Pattern String) is required to define the dialed digits or wildcard pattern that the route pattern matches. Options B (Codec), C (Bandwidth Limit), and D (Softkey Template) are not mandatory fields when creating a route pattern; codec and bandwidth are typically associated with regions/locations or device pools, and softkey templates apply to devices such as phones, not to route patterns themselves.

Exam trap

CLCOR often tests the misconception that codec or bandwidth settings are part of a route pattern, when those belong to regions, locations, and device pools.

Full explanation →

141

MCQeasy

When adding a new phone, what is the 'Device Name' parameter based on by default?

A.The device pool name.

B.The extension number.

C.The user ID.

D.The MAC address of the phone.

AnswerD

By default, the Device Name field is auto-populated with the phone's MAC address, giving each device a unique identifier during registration. This satisfies the stem's question about what the parameter is based on without manual entry.

Why this answer

When adding a new phone in Cisco Unified Communications Manager, the Device Name field is automatically populated with the phone's MAC address by default. This ensures each device has a unique identifier in the CUCM database, which is essential for configuration file delivery and registration.

Exam trap

CLCOR often tests the assumption that the device name is based on the extension or user ID, when in fact it defaults to the MAC address; candidates must remember that the MAC is the unique hardware identifier used for TFTP config retrieval.

How to eliminate wrong answers

Option A is wrong because the device pool name is a logical grouping for settings like region, date/time, and CM group, not the device identifier. Option B is wrong because the extension number (directory number) is associated with lines on the phone, not the device itself. Option C is wrong because the user ID is used for user association and authentication, not as the device name.

Full explanation →

142

MCQmedium

A Cisco IP Phone 8845 registered to CUCM is not receiving its configuration file from the TFTP server after a factory reset. The phone has received its IP address via DHCP and can ping the TFTP server. Which option in the DHCP scope must be configured to allow the phone to locate the TFTP server?

A.Option 42

B.Option 66

C.Option 3

D.Option 150

AnswerD

Option 150 provides the TFTP server address to Cisco IP Phones. Without it, the phone cannot locate the TFTP server to download its configuration file. This is the correct answer because the phone has network connectivity but lacks the TFTP server information, which is typically delivered via DHCP option 150 in Cisco environments.

Why this answer

Cisco IP Phones rely on DHCP option 150 to learn the IP address of the TFTP server that hosts their configuration files. Without this option, the phone cannot proceed with provisioning. The other options serve different purposes: option 66 is a non-standard alternative, option 42 is for NTP, and option 3 is for the default gateway.

Only option 150 directly addresses the phone's need to locate the TFTP server.

Exam trap

The trap here is confusing DHCP option 150 with option 66, as both can provide TFTP server information, but Cisco IP Phones specifically require option 150.

Full explanation →

143

MCQmedium

You are configuring Webex Hybrid Message Service. What is the primary purpose of this service?

A.To provide external federation for IM users.

B.To enable messaging between on-premises IM&P users and Webex App users.

C.To archive all Webex messages to an on-premises server.

D.To synchronize user passwords with Active Directory.

AnswerB

Webex Hybrid Message Service connects the on-premises IM and Presence deployment with the Webex App cloud, letting users exchange messages across both environments. It satisfies the requirement to federate messaging rather than replace the existing IM&P infrastructure.

Why this answer

Webex Hybrid Message Service (Hybrid Message) connects an on-premises Cisco IM and Presence (IM&P) deployment with Webex App so users on both sides can exchange instant messages. It is a hybrid connector-based service that federates the on-prem XMPP infrastructure with the Webex cloud messaging fabric, allowing a phased migration where some users remain on-prem while others move to the cloud.

Exam trap

CLCOR often tests the confusion between Hybrid Message (on-prem to cloud IM bridging) and federation or archiving features, so candidates who pick 'external federation' miss the on-prem-to-cloud scope.

How to eliminate wrong answers

Option A is wrong because external federation for IM users is handled by Expressway or Webex federation features, not by Hybrid Message Service. Option C is wrong because message archiving to an on-premises server is a compliance/eDiscovery function typically handled by Webex Compliance or a third-party archiver, not Hybrid Message. Option D is wrong because password synchronization with Active Directory is handled by Cisco Directory Connector or SSO integration, not Hybrid Message Service.

Full explanation →

144

MCQmedium

What is the result of applying 'voice translation-profile' to an incoming dial-peer?

A.It enables SIP

B.It changes the codec

C.It modifies the digits

D.It drops the call

AnswerC

Applying a voice translation-profile to an incoming dial-peer rewrites the called or calling number before outbound dial-peer matching occurs, satisfying the requirement to manipulate digits on inbound calls. The profile's rules execute at that point in the call leg, so translation happens prior to any further routing decisions.

Why this answer

A voice translation-profile applied to an incoming dial-peer rewrites the calling, called, or redirecting numbers before the call is matched and routed. It uses translation rules to manipulate digits (add, strip, replace prefixes), which is why the correct result is digit modification.

Exam trap

CLCOR often tests whether candidates confuse translation-profile (digit manipulation) with codec or protocol settings on a dial-peer, so the trap is assuming any dial-peer attribute changes the codec or signaling.

How to eliminate wrong answers

Option A is wrong because SIP enablement is controlled by session protocol configuration on the dial-peer (session protocol sipv2), not by a translation profile. Option B is wrong because codec selection is determined by the codec preference list on the dial-peer (codec preference), not translation rules. Option D is wrong because translation profiles rewrite digits; they do not drop calls — call rejection is handled by other mechanisms like call blocking or translation rules that produce no match.

Full explanation →

145

MCQhard

A user is unable to use their 'Speed Dial' button. What is the first thing you should verify?

A.The device pool.

B.The user's password.

C.The DN's partition.

D.The Phone Button Template.

AnswerD

The Phone Button Template defines which features, such as Speed Dial, appear on each line's buttons. If the template assigned to the device omits the Speed Dial button, the user cannot access it regardless of other configuration, so verifying the template first isolates the actual constraint.

Why this answer

A Speed Dial button is a programmable line key whose function is defined by the Phone Button Template assigned to the phone. If the template does not include a Speed Dial (or the button is not configured as Speed Dial), the button will not work regardless of DN or partition settings. Verifying the Phone Button Template is therefore the first step.

Exam trap

CLCOR often tests the misconception that a speed dial failure is a dial plan/partition issue, when the first check should be whether the Phone Button Template even defines a Speed Dial button.

How to eliminate wrong answers

Option A is wrong because the device pool controls region, location, date/time, and softkey template — not button functions. Option B is wrong because the user's password affects authentication/extension mobility login, not the physical button behavior. Option C is wrong because the DN's partition affects call routing/CSS reachability, but a Speed Dial button that is not defined in the template will not function even if the DN is in a reachable partition.

Full explanation →

146

MCQmedium

A collaboration engineer is troubleshooting a one-way audio issue between two Cisco IP Phones registered to the same Cisco Unified Communications Manager cluster. The phones are in different locations separated by a WAN link. The engineer captures traffic and notices that the SIP signaling is correct, but RTP packets are only flowing in one direction. The engineer suspects a NAT or firewall issue. Which protocol or feature should the engineer verify to ensure that RTP can traverse the NAT and firewalls correctly?

A.SIP Early Offer

B.Media Termination Point (MTP)

C.Skinny Client Control Protocol (SCCP)

D.Interactive Connectivity Establishment (ICE)

AnswerD

ICE is a protocol that enables NAT traversal for media streams by discovering the best path between endpoints. It uses STUN and TURN servers to gather candidate addresses and perform connectivity checks. In a CUCM environment, ICE can be enabled on SIP profiles to allow endpoints to negotiate media through NAT. For one-way audio due to NAT, verifying ICE configuration is essential. This is the correct feature to ensure RTP traverses NAT.

Why this answer

One-way audio often results from NAT or firewall blocking RTP in one direction. ICE is a protocol that helps endpoints discover and negotiate media paths through NAT by using STUN and TURN. In CUCM, ICE can be enabled on SIP profiles to facilitate NAT traversal.

Verifying ICE configuration is the correct step to ensure RTP flows bidirectionally. Other options like Early Offer or MTP do not directly address NAT traversal for media.

Exam trap

The trap here is assuming that any media-related feature like MTP or Early Offer will solve NAT traversal, when ICE is specifically designed for that purpose.

Full explanation →

147

Multi-Selecthard

When troubleshooting Webex Calling, which THREE factors are critical for maintaining call quality over a hybrid network?

Select 3 answers

A.Quality of Service (QoS) tagging

B.Local database replication speed

C.Low round-trip time (latency)

D.Sufficient upload/download bandwidth

E.Hardware firewall bypass

AnswersA, C, D

Quality of Service tagging marks voice packets with DSCP values so routers and switches can prioritise them over bulk data across the WAN. This directly satisfies the hybrid network constraint, where traffic traverses both on-premises and cloud paths, keeping jitter and packet loss within acceptable bounds for Webex Calling media.

Why this answer

Option A (QoS tagging) is correct because marking voice RTP packets with DSCP values such as EF (46) lets routers and switches prioritize real-time media over bulk data, preventing jitter and packet loss during congestion. Option C (low round-trip time/latency) is correct because Webex Calling uses RTP over UDP, and excessive latency (generally above 150 ms one-way) causes talk-over and degraded interactivity that no amount of bandwidth can fix. Option D (sufficient upload/download bandwidth) is correct because each concurrent call consumes roughly 100 kbps per direction plus overhead, and insufficient throughput leads to dropped packets, choppy audio, and failed call setup.

Option B (local database replication speed) is not relevant because Webex Calling is a cloud service and does not depend on on-premises directory replication for media quality. Option E (hardware firewall bypass) is not a valid practice; firewalls should be properly configured with the correct ports and media traversal rather than bypassed, which would create security risks without guaranteeing call quality.

Exam trap

CLCOR often tests the factors affecting call quality, and candidates may select irrelevant options like database replication or firewall bypass due to misunderstanding of voice transport requirements.

Full explanation →

148

MCQhard

You are integrating an external voicemail system with CUCM. Which object is used to define the dialable number that users call to reach their voicemail box?

A.Voicemail Pilot

B.Voicemail Profile

C.Route Pattern

D.Translation Pattern

AnswerA

The Voicemail Pilot defines the dialable pilot number that routes callers to the external voicemail system, and CUCM uses it to allocate a contiguous range of voicemail ports. Without this object, users have no number to dial to reach their mailbox.

Why this answer

The Voicemail Pilot is the CUCM object that defines the actual dialable number (directory number) users dial to reach the voicemail system — for example, 5000 for Unity Connection. It acts as the routing anchor that points CUCM to the voicemail server, and it is referenced by the Voicemail Profile. Without a Voicemail Pilot, CUCM has no number to send calls to when a phone's voicemail button is pressed or a busy/no-answer condition triggers a forward to voicemail.

Exam trap

CLCOR often tests the confusion between the Voicemail Pilot (the dialable number) and the Voicemail Profile (the assignment/mask object), causing candidates to pick the profile when asked for the dialable number.

How to eliminate wrong answers

Option B is wrong because the Voicemail Profile does not define a dialable number; it associates a voicemail pilot number with a set of voicemail box masks and is assigned to line/device configurations so CUCM knows which pilot to use. Option C is wrong because a Route Pattern is a general call-routing construct that matches dialed digits and points to a route list/gateway — it is not the voicemail-specific object that defines the voicemail access number. Option D is wrong because a Translation Pattern manipulates dialed digits (rewriting, prefixing, stripping) before routing; it does not define the voicemail pilot number itself.

Full explanation →

149

MCQeasy

Which dashboard in Control Hub provides insights into the quality and usage of hybrid services?

A.Device Dashboard

B.Troubleshooting Dashboard

C.Analytics Dashboard

D.Management Dashboard

AnswerC

The Analytics Dashboard in Control Hub aggregates hybrid service telemetry, showing call quality metrics and adoption trends across registered nodes. It satisfies the requirement for both quality and usage insight, which the troubleshooting and configuration views do not provide.

Why this answer

The Analytics Dashboard in Control Hub provides insights into quality and usage of hybrid services, including call quality, media statistics, and service adoption. It aggregates data from hybrid connectors and services to show trends and performance. This is the designated dashboard for hybrid service analytics.

Exam trap

CLCOR often tests whether candidates confuse the Analytics Dashboard with the Troubleshooting Dashboard, picking the latter for hybrid service insights when the question asks about quality and usage trends rather than live diagnostics.

How to eliminate wrong answers

Option A is wrong because the Device Dashboard focuses on device inventory, status, and management, not hybrid service quality and usage analytics. Option B is wrong because the Troubleshooting Dashboard is for diagnosing specific call issues, not for broad hybrid service insights. Option D is wrong because there is no 'Management Dashboard' in Control Hub that serves this purpose; management functions are spread across other sections.

Full explanation →

150

MCQmedium

A Cisco IOS XE gateway is failing to route calls because the dial-peer is not matching the incoming call. Which command helps you identify the dial-peer that matched the incoming call?

A.show voice call status

B.show dial-peer voice summary

C.debug ccsip messages

D.debug voip dialpeer all

AnswerD

debug voip dialpeer all traces dial-peer matching, showing which peer is selected for the incoming call and why others were rejected. This directly reveals the mismatch causing the routing failure, unlike show commands that only display configuration.

Why this answer

The command 'debug voip dialpeer all' enables debugging for all dial-peer matching events, showing which dial-peer is matched for incoming calls. This helps identify why a dial-peer is not matching, as it displays the matching process and the resulting dial-peer number.

Exam trap

CLCOR often tests whether candidates confuse show commands that summarize dial-peer configuration with debug commands that reveal real-time matching decisions, leading them to pick 'show dial-peer voice summary' instead of the debug command.

How to eliminate wrong answers

Option A is wrong because 'show voice call status' displays active call status but does not show dial-peer matching details. Option B is wrong because 'show dial-peer voice summary' provides a summary of configured dial-peers but does not indicate which one matched a specific incoming call. Option C is wrong because 'debug ccsip messages' shows SIP signaling messages, which can help with call setup issues but does not directly reveal dial-peer matching decisions.

Full explanation →

Cisco Implementing Cisco Collaboration Core Technologies (CLCOR, 350-801, v2.0 blueprint, CCNP Collaboration) (CLCOR) (CLCOR) — Questions 76–150