CLCOR Timed Practice Test 1 — 10 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?

10 minute time limit — choose an answer to begin.

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.