CLCOR Exam Simulation 1 — 100 Questions

Question 1

A user is unable to receive incoming PSTN calls via a Webex Calling Local Gateway. You observe that the INVITE from the provider reaches the CUBE, but the CUBE sends a 403 Forbidden. What is the most likely cause?

Choose an answer to begin — your selection is scored in the full session.

Accepted Answer

The incoming dial peer is missing or misconfigured. A 403 Forbidden response from the CUBE indicates that the incoming INVITE is being rejected due to a dial peer mismatch. The incoming dial peer is responsible for matching incoming calls and applying the correct configuration; if it is missing or misconfigured, the CUBE cannot route the call and returns 403.

Answer

The PSTN provider is using an unsupported codec

Answer

The Webex cloud is unreachable

Answer

The user's directory number is not in the PSTN white list

Question 2

When using 'Call Pickup', how does a user pick up a call that is ringing on another phone in the same pickup group?

Accepted Answer

By pressing the 'Pickup' softkey.. Call Pickup allows a user to answer a call ringing on another phone in the same pickup group by pressing the 'Pickup' softkey (or dialing the pickup group code). The softkey triggers the Call Pickup feature, which redirects the ringing call to the user's phone. This is a standard CUCM feature for group call answering.

Answer

By dialing the target phone extension.

Answer

By selecting 'Hold' on their own line.

Answer

By accessing the CUCM web interface.

Question 3

A network administrator is configuring a Cisco IP Phone 7811 to register with Cisco Unified Communications Manager. The phone is powered on and connected to the network. The administrator notices that the phone is not obtaining an IP address. Which protocol is responsible for providing the phone with an IP address and other network parameters?

Accepted Answer

DHCP. Cisco IP Phones use DHCP to obtain an IP address and other network settings such as the TFTP server address and default gateway. Without DHCP, the phone cannot proceed with the boot process. TFTP is used later to download configuration files, CDP is for neighbor discovery, and SIP is for call signaling. Therefore, DHCP is the correct protocol.

Answer

TFTP

Answer

SIP

Answer

CDP

Question 4

A user complains that they cannot see the 'Transfer' softkey on their phone during an active call. Where is the most appropriate place to check the softkey configuration?

Accepted Answer

Device > Device Settings > Softkey Template. Softkey templates in Cisco Unified Communications Manager define which softkeys appear on a phone during different call states, and they are configured under Device > Device Settings > Softkey Template. The template is then associated with a device pool or directly with a phone, but the actual softkey definitions (including Transfer) live in the template. Checking the template is the correct first step to verify whether the Transfer softkey is enabled for the connected/active call state.

Answer

System > Enterprise Phone Configuration

Answer

Device > Device Pool

Answer

Device > Phone

Question 5

A Cisco IOS XE gateway is configured with a POTS dial-peer that forwards calls to a PRI interface. Users report that inbound PSTN calls are being rejected with a cause code indicating that all circuits are busy, even though the PRI shows no active calls. Which command should you use to verify that the POTS dial-peer is matching the inbound calls correctly?

Accepted Answer

debug dialpeer. The debug dialpeer command is essential for troubleshooting dial-peer matching because it logs the inbound and outbound dial-peer selection process. When inbound PSTN calls are rejected with a busy cause despite idle circuits, the gateway may be failing to match a POTS dial-peer. This debug reveals whether the call is matching the intended peer or falling through to a default that rejects the call.

Answer

debug isdn q931

Answer

show dial-peer voice summary

Answer

show voice port summary

Question 6

What is the impact of removing a node from a Hybrid Video Mesh cluster that is currently hosting active meetings?

Accepted Answer

Active calls are dropped or experience disruption. Removing a node from a Hybrid Video Mesh cluster that is actively hosting meetings will cause active calls to be dropped or experience disruption. The node is immediately removed from the cluster, and any meetings hosted on it are not gracefully migrated, leading to service interruption. This is a known behavior in Cisco Webex Hybrid Video Mesh.

Answer

The node remains active until the last call ends

Answer

The cloud takes over the media immediately

Answer

The meetings are automatically migrated to another node

Question 7

A company has deployed Webex Hybrid Call Service Connect to allow Webex App users to make calls using their on-premises Cisco Unified Communications Manager (CUCM) infrastructure. Users report that they can see presence but cannot make calls. Which component should be checked first to ensure that call signaling is working?

Accepted Answer

Expressway-C connector. The Expressway-C connector is the correct component to check first because it is responsible for call signaling between Webex and CUCM in Hybrid Call Service Connect. If it is not functioning, users may experience presence but cannot make calls. Verifying its status and logs will help identify signaling problems.

Answer

Cisco Expressway-E

Answer

Cisco Unified Communications Manager IM and Presence Service

Answer

Webex Hybrid Data Security

Question 8

When managing Webex Calling using the 'Webex Calling' app versus the 'Webex' app, what determines the calling capabilities?

Accepted Answer

The user's assigned license in Control Hub.. In Webex Control Hub, the calling capabilities a user receives are determined by the license assigned to that user. A Webex Calling license (e.g., Professional or Standard) enables PSTN calling, voicemail, and enterprise features, while a Webex Meetings-only license does not. The app used (Webex Calling app vs. Webex app) is just a client; the backend entitlement comes from the license. Therefore, the assigned license in Control Hub is the determining factor.

Answer

The firmware version of the desk phone.

Answer

The location of the user's home network.

Answer

The operating system of the device.

Question 9

A collaboration engineer is deploying a new branch office with Cisco Unified Communications Manager and wants to ensure that voice traffic is marked with the appropriate DSCP value. The service provider requires that voice bearer traffic use EF ( Expedited Forwarding). Which DSCP value should be configured on the IP phones for voice media?

Accepted Answer

EF (46). Voice bearer traffic should be marked with EF (Expedited Forwarding, DSCP 46) to ensure low latency, jitter, and loss. This is a standard recommendation and is often required by service providers. The IP phone configuration must set the voice media DSCP to EF, while call signaling typically uses CS3.

Answer

AF41 (34)

Answer

CS3 (24)

Answer

CS5 (40)

Question 10

A company's main site runs a Cisco Unified Communications Manager (CUCM) cluster. They are adding a new remote branch that will have 50 IP phones. The branch has limited bandwidth and must survive a WAN outage with basic call functionality between local phones. Which design approach should be used at the branch?

Accepted Answer

Deploy a Cisco Unified Communications Manager Express (CME) router at the branch to provide local call processing.. For small remote branches with limited bandwidth and a need for local survivability, a Cisco Unified Communications Manager Express (CME) router is the appropriate choice. It provides local call processing and can be integrated with the central CUCM for centralized management and dial plan. This ensures that intra-branch calls continue even if the WAN link fails.

Answer

Deploy a CUCM subscriber at the branch to handle local call processing.

Answer

Deploy a Cisco Expressway-C and Expressway-E pair at the branch for call processing.

Answer

Use the existing CUCM cluster at the main site and rely on the WAN for all call processing.

Question 11

You are deploying Webex Hybrid Data Security (HDS). After deploying the virtual machines for the HDS nodes, you find they are not appearing in Control Hub. What is the first step you should take to troubleshoot the registration?

Accepted Answer

Check the local log files in the HDS node using the CLI.. When HDS nodes do not appear in Control Hub, the first troubleshooting step is to check the local log files on the HDS node via the CLI. The logs will show whether the node successfully reached the Webex cloud, whether the registration token was accepted, and whether there are certificate or connectivity errors. This is the fastest way to identify the root cause before making configuration changes.

Answer

Disable the firewall on the HDS node.

Answer

Restart the Cisco Webex Hybrid Services host.

Answer

Re-download the management certificate.

Question 12

Which THREE of the following factors impact the quality of video calls in a network?

Accepted Answer

Packet loss. Packet loss (A) is correct because dropped RTP/UDP packets in a video stream cause visible artifacts, freezing, and audio gaps, directly degrading call quality. Jitter (C) is correct because variation in packet inter-arrival times disrupts the playout buffer, causing choppy audio and stuttering video even when average delay is acceptable. Latency (E) is correct because high end-to-end delay (typically above 150 ms one-way) impairs interactivity and, combined with jitter and loss, degrades the real-time experience. Interface description (B) is only a text label on an interface and has no effect on traffic handling, and router uptime (D) is an availability metric that does not by itself determine video quality as long as the device is forwarding packets.

Answer

Interface description

Answer

Router uptime

Question 13

You are configuring a Cisco IOS XE voice gateway to register with an ITSP over SIP. The provider requires registration to be refreshed every 60 seconds. You have already configured the registrar and credentials under sip-ua. Which command correctly sets the registration expiry timer to 60 seconds?

Accepted Answer

timers expires 60. The registration expiry timer on a Cisco IOS XE voice gateway is configured under sip-ua with the timers expires command. This timer determines how often the gateway sends a REGISTER refresh to the SIP registrar. Setting it to 60 seconds aligns with the provider's requirement for a 60-second refresh interval, preventing the registration from expiring unexpectedly.

Answer

authentication username user password pass

Answer

registrar ipv4:10.1.1.1 expires 60

Answer

retry invite 3

Question 14

You are configuring a local gateway for Webex Calling. Which mechanism ensures that the Webex Cloud can communicate with your on-premises edge device without requiring inbound firewall ports?

Accepted Answer

Outbound TLS connection from CUBE to Webex Cloud. Webex Calling uses a persistent outbound TLS connection from the on-premises CUBE (Customer Edge) to the Webex Cloud. This connection is initiated from the enterprise side, typically over TCP port 8934 (or 443 for some services), and is kept alive with keepalives. Because the connection is outbound, no inbound firewall ports need to be opened, and the cloud can send messages back over the established TLS session. This design is fundamental to Webex Calling's cloud-to-edge architecture, ensuring secure and NAT-friendly connectivity.

Answer

Webex Calling Connector via port forwarding

Answer

SIP Trunk over VPN

Answer

H.323 Gatekeeper registration

Question 15

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

Accepted Answer

Analytics Dashboard. 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.

Answer

Device Dashboard

Answer

Troubleshooting Dashboard

Answer

Management Dashboard

Question 16

Which protocol provides the signaling for Webex Room devices when they are registered to CUCM (on-premises)?

Accepted Answer

SIP. Webex Room devices, when registered on-premises to CUCM, use SIP (Session Initiation Protocol) for signaling. SIP is the standard protocol for establishing, modifying, and terminating multimedia sessions, and it is supported by Webex devices for call control with CUCM.

Answer

MGCP

Answer

SCCP

Answer

H.323

Question 17

A network engineer is configuring a Cisco IOS router to mark voice bearer traffic with the DSCP value EF (46) and call signaling traffic with CS3 (24). The engineer wants to ensure that both marking operations are applied in a single policy-map on the ingress interface. Which configuration element is required to differentiate the two traffic classes before marking?

Accepted Answer

A class-map that matches voice bearer and signaling using ACLs or protocol-specific match criteria.. The correct approach is to create a class-map that identifies voice bearer and signaling traffic based on protocol and port numbers, then reference that class-map in a policy-map where DSCP values are set. This ensures that both traffic types are properly marked regardless of their initial markings. Using ACLs or protocol-specific match criteria allows precise differentiation, which is essential for downstream QoS treatment.

Answer

A class-map that matches on IP precedence 5 for voice and 3 for signaling.

Answer

A policy-map that uses the 'set ip dscp' command with a table-map to translate incoming markings.

Answer

A route-map that matches on RTP and SIP ports and sets the DSCP values.

Question 18

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

Accepted Answer

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

Answer

Set the 'Call Processing Load' value in the System settings.

Answer

Enable Auto-registration for all phones.

Answer

Configure the cluster-wide failover parameter in the Service Parameter page.

Question 19

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

Accepted Answer

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

Answer

Latency

Answer

Codec Mismatch

Answer

Packet Loss

Question 20

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

Accepted Answer

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

Answer

show voice class sip-profiles

Answer

debug voip ccapi inout

Answer

debug sip-ua

Answer

show sip-profile