CLCOR Practice Test 14 — 15 Questions

Question 1

You are verifying the network readiness for a Cisco TelePresence endpoint. Which tool would you use to measure the packet loss, latency, and jitter on the path to the CUCM cluster?

Accepted Answer

IP SLA. IP SLA is the correct tool to measure packet loss, latency, and jitter on the path to the CUCM cluster. It uses synthetic traffic to monitor network performance metrics continuously, providing accurate measurements for voice and video readiness. This is essential for verifying network readiness for TelePresence endpoints.

Answer

NetFlow

Answer

SNMP

Answer

SPAN

Question 2

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

Accepted Answer

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

Answer

Device Pool configuration.

Answer

Route Group configuration.

Answer

SIP Profile.

Question 3

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

Accepted Answer

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

Answer

Route Pattern

Answer

Translation Pattern

Answer

Device Pool

Question 4

What is the default behavior of CUCM when a phone is moved to a new site and registers with the local CUCM node, but the Device Pool is not updated?

Accepted Answer

The phone registers but keeps old Region/Location settings.. When a phone moves to a new site and registers with a local CUCM node but its Device Pool is not updated, it retains the old Region and Location settings from its assigned Device Pool. The Device Pool defines these settings, and they are not automatically changed based on the registration node.

Answer

The phone refuses to register.

Answer

The phone forces a factory reset.

Answer

The phone defaults to the 'Default' device pool.

Question 5

Which TWO features are managed by the 'Cisco Unified Serviceability' tool?

Accepted Answer

Alarm and Trace configuration. Cisco Unified Serviceability is the Cisco Unified Communications Manager (CUCM) administration interface dedicated to operational and diagnostic functions, so Alarm and Trace configuration (option C) is correct because this tool is where you enable and configure alarms, traces, and the Real-Time Monitoring Tool (RTMT) alert/log settings for CUCM services. Service Activation (option E) is also correct because Cisco Unified Serviceability provides the Service Activation page used to start, stop, and restart network services and feature services (e.g., Cisco CallManager, Cisco TFTP, Cisco CTIManager) on the cluster nodes. IP Phone registration (option A) is not managed here; it is handled by the phone's configuration in Cisco Unified CM Administration and the registration process via the TFTP/CallManager services, not by Serviceability. Bulk phone import (option B) is performed with the Bulk Administration Tool (BAT) in Cisco Unified CM Administration, and user password reset (option D) is done through Cisco Unified CM Administration's End User configuration or via the Cisco Unified CM User Options/self-care pages, not through Serviceability.

Answer

IP Phone registration.

Answer

Bulk phone import.

Answer

User password reset.

Question 6

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

Accepted Answer

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

Answer

Translation patterns route calls; route patterns modify numbers.

Answer

There is no functional difference.

Answer

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

Question 7

You want to implement 'Presence' so users can see the status of other users. Which service must be running on the IM and Presence server?

Accepted Answer

Cisco Presence Engine. The Cisco Presence Engine is the core service on the IM and Presence server that processes and distributes presence information (status, availability, capabilities) between users. Without it running, presence updates cannot be published or subscribed to, so users cannot see each other's status.

Answer

Cisco SIP Proxy

Answer

Cisco XCP Router

Answer

Cisco TFTP

Question 8

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?

Accepted Answer

Line Group. 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.

Answer

Hunt List

Answer

Route Group

Answer

Hunt Pilot

Question 9

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

Accepted Answer

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

Answer

The account remains active indefinitely.

Answer

The account is moved to a guest group.

Answer

The account becomes a free account.

Question 10

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

Accepted Answer

System -> Cisco Unified Communications Manager Group. 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.

Answer

Serviceability -> CM Group

Answer

Device -> Device Settings -> CM Group

Answer

Advanced Features -> CM Group

Question 11

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

Accepted Answer

Media Termination Point (MTP). In Cisco Unified Communications Manager (CUCM), media resource types are configured under Media Resources and include annunciators, conference bridges, Media Termination Points (MTPs), music on hold (MOH) servers, and transcoders. Option C, Media Termination Point (MTP), is correct because an MTP is a software or hardware media resource registered to CUCM that provides supplementary services such as DTMF conversion, RSVP agent functionality, and H.323-to-SIP interworking by terminating and reoriginating media streams. Option E, Transcoder, is correct because a transcoder is a media resource that converts one codec (for example, G.711) to another (for example, G.729) so endpoints with mismatched codecs can communicate. Option A, IP Phone, is incorrect because a phone is an endpoint device, not a media resource type configured under Media Resources. Option B, Dial-Peer, is incorrect because dial peers are configuration elements on IOS voice gateways (used for call routing), not a CUCM media resource type. Option D, Voice Gateway, is incorrect because a voice gateway is a network device that connects CUCM to external networks such as PSTN or other PBXs, not a media resource type within CUCM.

Answer

IP Phone

Answer

Dial-Peer

Answer

Voice Gateway

Question 12

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

Accepted Answer

Network ACLs and port accessibility for RTP.. 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).

Answer

The phone's directory number.

Answer

The CSS of the device.

Answer

The phone's firmware version.

Question 13

Which THREE of the following items must be verified during a site-to-site VPN check for collaboration traffic?

Accepted Answer

MTU size. Option B (MTU size) is correct because site-to-site VPN tunnels add encapsulation overhead (e.g., IPsec ESP/AH headers), and if the MTU is not adjusted or fragmentation/PMTUD is not handled properly, large collaboration packets such as voice and video streams will be dropped or fragmented, degrading real-time traffic. Option D (UDP port range availability) is correct because collaboration traffic relies heavily on UDP (for example, RTP media streams and SIP signaling), so the required UDP port ranges must be permitted through the VPN and any intervening firewalls/ACLs for the tunnel to carry that traffic. Option E (DSCP tag preservation) is correct because collaboration traffic depends on QoS markings; if DSCP values are not preserved across the VPN (e.g., due to tunnel encapsulation or a misconfigured QoS policy), real-time voice and video packets lose their prioritization and suffer latency, jitter, and packet loss. Option A (DHCP relay configuration) is not required for a site-to-site VPN check because DHCP relay concerns local address assignment, not the transport of collaboration traffic across the tunnel. Option C (NTP server reachability) is not a mandatory item for this check, since NTP is important for general network health and certificate/logging accuracy but is not a direct requirement for forwarding collaboration traffic over a site-to-site VPN.

Answer

DHCP relay configuration

Answer

NTP server reachability

Question 14

Which TWO of the following are required components of a Cisco Local Gateway deployment? (Choose two)

Accepted Answer

A Cisco Unified Border Element (CUBE).. In a Cisco Local Gateway (LGW) deployment for Webex Calling, the Local Gateway itself is implemented on a Cisco Unified Border Element (CUBE), which provides the SIP normalization, media handling, and session border control between the enterprise voice network and Webex Calling — so option B is required. The Local Gateway must also be reachable over a SIP trunk that connects it to Webex Calling, since the LGW registers with and routes calls to the Webex Calling cloud via that SIP trunk, making option C required. Option A is not required because the Local Gateway does not depend on an on-premises Active Directory server; identity and user management reside in Webex Control Hub. Option D is not required because any supported internet connection with adequate bandwidth and QoS can carry the SIP trunk, not specifically dedicated fiber. Option E is not required because the Local Gateway connects to Webex Calling for PSTN, not to a separate cloud-based PSTN provider.

Answer

An on-premises Active Directory server.

Answer

A dedicated fiber-optic internet connection.

Answer

A cloud-based PSTN provider.

Question 15

You are migrating a legacy PBX to CUCM. You need to normalize incoming caller IDs from the PSTN to E.164 format. Which tool is best suited for this at the gateway level?

Accepted Answer

Calling Party Transformation Pattern. Calling Party Transformation Patterns are applied at the gateway or trunk level to normalize the calling party number (ANI) into E.164 format before the call is routed within CUCM. They can be assigned to a gateway, trunk, or route list, and they manipulate the calling number using digit manipulations (prefix, strip, etc.). For normalizing incoming PSTN caller IDs, this is the correct tool because it specifically targets the calling party field.

Answer

Translation Pattern

Answer

Route Filter

Answer

Route Pattern