CLCOR Practice Test 20 — 15 Questions

Question 1

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?

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

Accepted Answer

voice class sip-profiles. 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.

Answer

sip-header insert

Answer

header-manipulation

Answer

sip-header profile

Question 2

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?

Accepted Answer

Local Gateway. 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.

Answer

Webex Edge for Calling

Answer

Cloud Connected PSTN (CCP)

Answer

Cisco PSTN

Question 3

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?

Accepted Answer

Calling Search Space. 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.

Answer

SIP Trunk

Answer

Region

Answer

Device Pool

Question 4

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?

Accepted Answer

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

Answer

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

Answer

A policy-map with the set dscp ef command.

Answer

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

Question 5

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

Accepted Answer

Option 150. 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.

Answer

Option 43

Answer

Option 66

Answer

Option 12

Question 6

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

Accepted Answer

Cisco Unified Communications Manager. 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.

Answer

Cisco Expressway

Answer

Cisco Unity Connection

Answer

Cisco CUBE

Question 7

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

Accepted Answer

Locations CAC. 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.

Answer

Gatekeeper

Answer

Bandwidth Manager

Answer

RSVP

Question 8

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

Accepted Answer

They use regular expressions. 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.

Answer

They perform load balancing

Answer

They replace the SIP protocol

Answer

They define the codec

Question 9

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

Accepted Answer

session target. 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.

Answer

target ip

Answer

call target

Answer

destination-ip

Question 10

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

Accepted Answer

Quality of Service (QoS) tagging. 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.

Answer

Local database replication speed

Answer

Hardware firewall bypass

Question 11

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?

Accepted Answer

Voicemail Pilot. 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.

Answer

Voicemail Profile

Answer

Route Pattern

Answer

Translation Pattern

Question 12

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?

Accepted Answer

debug voip dialpeer all. 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.

Answer

show voice call status

Answer

show dial-peer voice summary

Answer

debug ccsip messages

Question 13

Which object acts as the primary container for a specific set of users in CUCM?

Accepted Answer

User Group. In Cisco Unified Communications Manager (CUCM), a User Group is the primary container for a set of users and serves as the mechanism for assigning roles and permissions. It aggregates users for administrative purposes, such as granting them access to CUCM administration or specific features via role assignments.

Answer

Device Pool

Answer

Location

Answer

Partition

Question 14

A collaboration administrator needs to verify that the Webex Hybrid Services connectors are healthy and operational. Where in Cisco Webex Control Hub can the administrator check the status of the connectors?

Accepted Answer

Monitoring > Hybrid Services. The health status of Webex Hybrid Services connectors is found under Monitoring > Hybrid Services in Control Hub. This dashboard provides a comprehensive view of each connector's status, including connectivity and any reported errors. Administrators use this page to quickly assess whether hybrid services are functioning correctly and to identify any components that require attention.

Answer

Troubleshooting > Hybrid Services

Answer

Services > Hybrid Services

Answer

Users > Hybrid Services

Question 15

What is the default behavior for DSCP markings on traffic originating from a Cisco IP Phone that has not been configured with specific QoS policies?

Accepted Answer

Marked as DSCP 46 for voice and CS3 for signaling. Cisco IP Phones ship with a built-in QoS trust behavior: voice media (RTP) is marked DSCP 46 (EF) and call signaling (SCCP/SIP) is marked CS3 (DSCP 24) by default, even before any switch or CUCM QoS policy is applied. This allows the phone to self-mark its traffic so the upstream switch can trust it via 'mls qos trust device cisco-phone' or 'trust dscp'. The phone only re-marks if a CUCM softphone/phone security profile or a switch policy overrides it.

Answer

Marked as DSCP 34 (AF41)

Answer

Traffic is not marked until a policy-map is applied

Answer

Marked as DSCP 0 (Best Effort)