CLCOR Practice Test 3 — 30 Questions

Question 1

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

Accepted Answer

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

Answer

The Hybrid Calendar connector.

Answer

A local PSTN gateway.

Answer

Cisco Jabber for Windows.

Question 2

Which service must be running for a Cisco IP Phone to download its configuration file via TFTP?

Accepted Answer

Cisco TFTP. Cisco IP Phones download their configuration files (XML) from the CUCM TFTP service, which runs as the 'Cisco TFTP' service on the CUCM server. Without this service running, phones cannot retrieve their config and will fail to register. The Cisco TFTP service is separate from CallManager and must be active on at least one node in the cluster.

Answer

Cisco CallManager

Answer

Cisco CTIManager

Answer

Cisco Database Layer Monitor

Question 3

You need to implement redundancy for a remote site using Survivable Remote Site Telephony (SRST). Which parameter in the Device Pool ensures that the phones fail over to the local router during a CUCM WAN outage?

Accepted Answer

SRST Reference. The SRST Reference is a Device Pool parameter that points to an SRST Reference object, which contains the IP address of the local router running SRST. When CUCM is unreachable, phones configured with that Device Pool fall back to the SRST router for call processing. This is the specific parameter that enables SRST failover for a remote site.

Answer

AAR Calling Search Space

Answer

Cisco Unified CM Group

Answer

Network Locale

Question 4

You are configuring a SIP trunk between CUCM and a CUBE. Which setting is critical to ensure that the SIP signaling is encrypted?

Accepted Answer

SIP Trunk Security Profile. The SIP Trunk Security Profile in CUCM defines the security settings for a SIP trunk, including whether TLS is used for signaling encryption and whether digest authentication is enabled. To ensure SIP signaling is encrypted, the profile must be configured with a 'Secure' transport (TLS) and applied to the trunk. This is the critical setting that controls encryption of SIP messages between CUCM and CUBE.

Answer

Enterprise Parameters

Answer

Device Pool

Answer

SIP Profile

Question 5

Which TWO of the following are functions performed by the CUBE as a Session Border Controller?

Accepted Answer

Media interworking. Option B (Media interworking) is correct because a CUBE acting as a Session Border Controller terminates the signaling and media on both legs, allowing it to transcode between different codecs (e.g., G.711 to G.729) and perform media flow-around or flow-through, which is a core SBC function. Option E (Topology hiding) is correct because CUBE uses back-to-back user agent (B2BUA) behavior to strip and rewrite SIP headers such as Via, Record-Route, and Contact, concealing the internal CUCM/ITSP topology from the far-end network. Option A (Call routing in CUCM) is not an SBC function; dial plan and call routing decisions are performed by Cisco Unified Communications Manager, not CUBE. Option C (IP phone registration) is handled by CUCM's TFTP and SCCP/SIP registration services, not by CUBE. Option D (Voice mail storage) is provided by Cisco Unity Connection, not by CUBE.

Answer

Call routing in CUCM

Answer

IP phone registration

Answer

Voice mail storage

Question 6

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

Accepted Answer

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

Answer

SIP Trunk Security Profile

Answer

Device Pool

Answer

Route List

Question 7

An administrator is designing a hub-and-spoke collaboration deployment. The hub has a 100 Mbps WAN link shared by many branches, and the engineer needs to ensure that a single branch cannot consume more than its fair share of bandwidth during peak periods. The engineer also wants voice calls to remain prioritized. Which QoS mechanism should be configured on the hub router to achieve both goals?

Accepted Answer

Configure CBWFQ with a bandwidth guarantee per branch class and add LLQ for voice in the same policy-map.. CBWFQ assigns each branch class a minimum bandwidth guarantee and distributes leftover bandwidth proportionally, preventing any single branch from dominating the shared hub link. Pairing it with LLQ for voice in the same policy-map gives voice strict-priority service, so both fairness among branches and voice quality are maintained during peak utilization.

Answer

Configure WRED on the WAN interface to drop packets from aggressive branches before they consume too much bandwidth.

Answer

Configure traffic shaping with shape average on each branch class to cap the bandwidth each branch can use.

Answer

Configure priority queueing only, with voice in the high queue and all branch traffic in the default queue.

Question 8

What is the purpose of the 'Common Device Configuration' in CUCM?

Accepted Answer

To group shared settings like softkey templates.. The Common Device Configuration in CUCM is a template that groups shared device-level settings — such as softkey templates, phone button templates, and other common parameters — so they can be applied to multiple devices without configuring each one individually. This reduces administrative overhead and ensures consistency across phones.

Answer

To manage trunk security.

Answer

To manage bandwidth.

Answer

To manage user passwords.

Question 9

Which service in CUCM is responsible for handling the registration of SIP devices?

Accepted Answer

Cisco CallManager. The Cisco CallManager service (Cisco CallManager, often abbreviated CCM) is the core call-processing service in CUCM that handles SIP device registration, call routing, and signaling. SIP phones and gateways register with the CallManager service running on a CUCM node, which maintains the registration state and provides call control.

Answer

Cisco Database Layer Monitor

Answer

Cisco CTIManager

Answer

Cisco TFTP

Question 10

When designing a redundant CUCM deployment, which mechanism ensures that if a primary subscriber fails, phones automatically register to a secondary subscriber?

Accepted Answer

CM Group. A Cisco CallManager Group (CM Group) is an ordered list of CUCM nodes that is assigned to a Device Pool, which in turn is assigned to phones. The CM Group defines the primary subscriber and up to two backup subscribers; if the primary fails, phones fall back to the next node in the list in the order specified. This is the native CUCM device redundancy mechanism for phone registration.

Answer

SIP Trunk Security Profile

Answer

Route List

Answer

Device Pool

Question 11

You are designing a solution for secure voice signaling. Which protocol provides encryption for SIP signaling messages in a CUCM environment?

Accepted Answer

TLS. TLS (Transport Layer Security) is the protocol used to encrypt SIP signaling messages in a CUCM environment. SIP over TLS (SIPS) provides encryption and authentication for signaling, ensuring that messages like INVITE, BYE, and REGISTER are secure. CUCM supports TLS for SIP trunking and line-side signaling.

Answer

IPsec

Answer

SRTP

Answer

SSH

Question 12

Which protocol is used by IP phones to register with the Cisco Unified Communications Manager?

Accepted Answer

SCCP or SIP. Cisco IP phones register with Cisco Unified Communications Manager using either SCCP (Skinny Client Control Protocol) or SIP (Session Initiation Protocol), depending on the phone model and firmware configuration. Both are supported signaling protocols for phone registration and call control in CUCM.

Answer

RTCP

Answer

MGCP

Answer

H.323

Question 13

When configuring Hybrid Call Service for Cisco Unified Communications Manager, which specific requirement must be met for the Expressway-C and Expressway-E pair?

Accepted Answer

They must be configured with a Unified Communications traversal zone.. For Hybrid Call Service with Cisco Unified Communications Manager, the Expressway-C and Expressway-E pair must be configured with a Unified Communications traversal zone. This zone enables Mobile and Remote Access (MRA) by allowing signaling and media to traverse the firewall securely between the internal CUCM and external endpoints.

Answer

They must use the same certificate authority as the CUCM nodes.

Answer

They must use a public IP address on the Expressway-C.

Answer

They must be in the same subnet.

Question 14

Which option allows a customer to keep their existing PSTN numbers while moving to Webex Calling?

Accepted Answer

Number porting.. Number porting is the regulated process (governed by FCC/local authorities) that transfers an existing PSTN number from one carrier to another while preserving the number itself. When migrating to Webex Calling, the customer submits a port request with a Letter of Authorization (LOA) and recent bill so Cisco's carrier partner can pull the DIDs from the incumbent provider. This is the only option that actually moves ownership of the number rather than just redirecting or labeling it.

Answer

Number aliasing.

Answer

Number forwarding.

Answer

Number provisioning.

Question 15

A collaboration engineer is troubleshooting poor video quality on a Cisco Collaboration deployment. The engineer suspects that video traffic is not receiving appropriate QoS treatment and wants to verify the classification and marking on a Cisco IOS router. Which two actions should the engineer take to confirm that video traffic is being handled correctly? (Choose two.)

Accepted Answer

Use the show class-map command to confirm the match criteria and verify the video class is referencing the correct DSCP or ACL.. Verifying QoS for video requires checking both the classification logic and the applied policy. The show class-map command confirms the match criteria used to identify video, while show policy-map interface reveals per-class match counts and drops on the interface. Together they confirm that video is classified correctly and treated as intended.

Answer

Use the show call active voice brief command to inspect the codec negotiated for video calls.

Answer

Run the show interfaces command to check the interface error counters for video drops.

Answer

Run the debug ip rsvp command to observe reservation messages for video flows.

Question 16

A user's Cisco IP Phone 8841 is registered to Cisco Unified Communications Manager but cannot place calls to any internal or external numbers. The phone displays 'Registration Complete' but every call attempt results in a fast busy tone. Which configuration element is most likely missing or misconfigured for this phone?

Accepted Answer

The phone's Calling Search Space does not include the partitions containing the dialed numbers.. A registered phone that cannot place any calls typically has a Calling Search Space that lacks access to the partitions containing the dialed numbers. The CSS must include the partitions where directory numbers and route patterns reside. Without this, digit analysis fails and the phone receives a fast busy tone. Other settings like Region or security profile do not block call placement in this manner.

Answer

The phone's Device Security Profile is set to Non-Secure.

Answer

The phone's Network Locale is set to United States.

Answer

The phone's Device Pool is not assigned to a Region.

Question 17

When configuring CUBE for media flow-around, which command ensures the media goes directly between endpoints?

Accepted Answer

media flow-around. The 'media flow-around' command in CUBE configuration tells the Cisco Unified Border Element to leave the RTP media streams out of the signaling path, allowing endpoints to exchange media directly. This is the correct syntax for enabling media flow-around on CUBE. It is configured under the voice service voip or dial-peer context to control whether media is anchored or bypassed.

Answer

media pass-through

Answer

media-bypass

Answer

media flow-through

Question 18

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

Accepted Answer

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

Answer

To monitor network bandwidth.

Answer

To manage real-time call logs.

Answer

To perform bulk configuration.

Question 19

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

Accepted Answer

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

Answer

Phone NTP settings

Answer

SIP Trunk Security Profile mismatch

Answer

Location-based bandwidth limit

Question 20

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

Accepted Answer

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

Answer

The call is placed but with a lower quality codec

Answer

The call is put on hold until bandwidth is available

Answer

The call proceeds, but marks the packets with Best Effort