Cisco Implementing Cisco Collaboration Core Technologies (CLCOR, 350-801, v2.0 blueprint, CCNP Collaboration) (CLCOR) CLCOR Questions 226–259 | Page 4/4

226

MCQhard

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?

A.Translation Pattern

B.Route Filter

C.Route Pattern

D.Calling Party Transformation Pattern

AnswerD

Calling Party Transformation Patterns applied at the gateway rewrite inbound calling numbers into E.164 before CUCM processes them. This satisfies the stem's normalisation requirement at the gateway level, unlike route patterns or translation patterns that operate on dialled digits rather than the calling party.

Why this answer

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.

Exam trap

CLCOR often tests the difference between Translation Patterns (called party/dialed digits) and Calling Party Transformation Patterns (calling party/ANI), causing candidates to pick Translation Pattern for caller ID normalization.

How to eliminate wrong answers

Option A is wrong because a Translation Pattern is used to translate dialed digits (called party) and route calls; it can also transform calling party, but it is not the dedicated gateway-level tool for normalizing incoming ANI and is typically used for dial plan translation, not PSTN caller ID normalization. Option B is wrong because a Route Filter is used to filter route groups based on dialed digits or calling party number for route selection, not to transform the number itself. Option C is wrong because a Route Pattern matches dialed digits and points to a route list/gateway; it does not normalize the calling party number.

Full explanation →

227

MCQeasy

Which component in the Cisco Collaboration architecture provides the initial configuration files (XML) to an IP phone upon registration?

A.CallManager Service

B.TFTP Service

C.CTI Manager

D.Webex Control Hub

AnswerB

The TFTP service delivers the XML configuration files an IP phone requests during boot, satisfying the stem's requirement for initial provisioning on registration. Phones retrieve their SEP<MAC>.cnf.xml and dial-plan definitions from this service before registering with Cisco Unified Communications Manager, making it the correct component.

Why this answer

The Cisco TFTP service, running on Cisco Unified Communications Manager, is responsible for serving XML configuration files, phone firmware, and other provisioning files to IP phones during registration. When a phone boots, it requests its SEP<MAC>.cnf.xml file from the TFTP server, which contains its directory number, device pool, and cluster settings. Without TFTP, phones cannot complete registration.

Exam trap

CLCOR often tests whether candidates confuse the CallManager service (call control) with the TFTP service (provisioning) — registration signaling and configuration file delivery are handled by different services.

How to eliminate wrong answers

Option A is wrong because the CallManager service (Cisco CallManager) handles call signaling, registration, and call control — it does not serve configuration files. Option C is wrong because CTI Manager provides JTAPI/TAPI interfaces for applications like contact centers and does not handle phone provisioning. Option D is wrong because Webex Control Hub is the cloud management portal for Webex, not the on-premises TFTP provisioning service for CUCM-registered phones.

Full explanation →

228

MCQhard

A Cisco Unified Border Element (CUBE) is deployed between Cisco Unified Communications Manager and a SIP provider. The provider sends an INVITE with a Diversion header containing the original called number. You need to strip the Diversion header before sending the INVITE to Unified CM. Which configuration element should you use?

A.voice class sip-profiles with a sip-header Diversion remove rule applied on the outbound dial peer

B.dial-peer voice with the diversion-header disable command

C.voice class uri with a diversion pattern to match and block the header

D.voice service voip with the sip-header Diversion remove command

AnswerA

SIP profiles allow you to manipulate SIP headers. A sip-header Diversion remove rule within a voice class sip-profiles will remove the Diversion header from the outgoing SIP message. Applying the profile to the outbound dial peer ensures the header is stripped before the INVITE is sent to Unified CM. This is the correct and supported method for header manipulation on CUBE.

Why this answer

To remove a SIP header such as Diversion, you must use a voice class sip-profiles with a sip-header command that specifies the header name and the action. The profile is then applied to the outbound dial peer so that the header is stripped from the INVITE before it is sent to the next hop. This is the standard method on CUBE.

Exam trap

The trap here is assuming that voice class uri or global voice service voip can manipulate SIP headers; only voice class sip-profiles provides that capability.

Full explanation →

229

MCQmedium

You have configured a Cisco Unified CM Route List with two Route Groups: one containing a SIP trunk to Provider A and another containing a SIP trunk to Provider B. You want calls to be attempted to Provider A first, and only if that fails, to Provider B. Which Route List configuration ensures this behavior?

A.Set the Route List's distribution algorithm to Top Down and ensure the Route Group for Provider A is listed first.

B.Set the Route List's distribution algorithm to Broadcast and ensure the Route Group for Provider A is listed first.

C.Set the Route List's distribution algorithm to Circular and ensure the Route Group for Provider A is listed first.

D.Set the Route List's distribution algorithm to Longest Idle and ensure the Route Group for Provider A is listed first.

AnswerA

The Top Down distribution algorithm in a Route List tries Route Groups in the order they are listed. By placing the Route Group for Provider A first, calls will be attempted to Provider A first. If that fails, CUCM will then try the next Route Group, Provider B. This is the standard way to achieve primary/secondary routing behavior.

Why this answer

To route calls to Provider A first and then Provider B only on failure, the Route List must use the Top Down distribution algorithm and list the Route Group for Provider A before Provider B. Top Down tries each Route Group in order, providing sequential failover. Other algorithms like Circular or Longest Idle do not provide this strict primary/secondary behavior.

Exam trap

The trap here is assuming that any distribution algorithm will respect the order of Route Groups for failover, when in fact only Top Down guarantees sequential attempts.

Full explanation →

230

MCQeasy

What is the primary benefit of the 'Webex Edge' strategy?

A.It allows you to use your own cloud hosting.

B.It removes the need for user authentication.

C.It optimizes the quality and reliability of Webex traffic.

D.It eliminates the need for internet access.

AnswerC

Webex Edge connects your organisation's Webex services through dedicated peering into the Webex backbone, bypassing the public internet for media. This optimises quality and reliability of Webex traffic, which is the primary benefit the question asks for.

Why this answer

Webex Edge is a set of hybrid options that let enterprises leverage their existing on-premises infrastructure and network to improve the quality, reliability, and manageability of Webex traffic. By peering Webex traffic through the customer's own network or dedicated connections rather than the public internet, latency and jitter are reduced, and the customer gains visibility and control. It does not replace the Webex cloud — it optimizes how traffic reaches it.

Exam trap

CLCOR often tests the misconception that Webex Edge means hosting Webex yourself or removing authentication, when it actually means optimizing connectivity and hybrid integration with the Cisco cloud.

How to eliminate wrong answers

Option A is wrong because Webex Edge does not let customers host the Webex cloud themselves; the service still runs in Cisco's cloud. Option B is wrong because user authentication is still required and is typically handled via SSO or Control Hub identity management. Option D is wrong because Webex Edge still requires internet or private connectivity to reach the Webex cloud — it does not eliminate the need for connectivity.

Full explanation →

231

MCQeasy

A collaboration administrator is troubleshooting a Webex Hybrid Services deployment. The connector for Hybrid Calendar Service shows as 'Down' in Control Hub. The administrator has verified that the connector server has network connectivity to the internet. Which additional check should be performed first to resolve the connector status?

A.Reinstall the Webex App on all user devices.

B.Verify that the connector's service account credentials have not expired.

C.Check the Expressway-E certificate expiration date.

D.Restart the Cisco Unified Communications Manager service.

AnswerB

The connector uses a service account to authenticate with the Webex cloud. If the credentials expire or are changed, the connector cannot authenticate and will show as 'Down'. Checking and updating the service account credentials is a primary troubleshooting step for a down connector, especially when network connectivity is confirmed.

Why this answer

When a Hybrid Services connector shows as 'Down' and network connectivity is confirmed, the most common cause is an authentication issue. The connector uses a service account to connect to the Webex cloud; if those credentials expire or are incorrect, the connector cannot establish a session. Verifying and updating the credentials is the logical first step before investigating other potential causes.

Exam trap

The trap here is assuming that network connectivity is the only requirement and overlooking that the connector also depends on valid service account credentials.

Full explanation →

232

MCQmedium

A Cisco IOS XE voice gateway is configured with two POTS dial peers pointing to the same FXO port. Dial peer 10 matches destination pattern 911, and dial peer 20 matches destination pattern 9T. An inbound PSTN call arrives and is matched to a VoIP dial peer that forwards it to the gateway's FXO trunk. Which dial peer will the gateway use for the outbound POTS leg, and why?

A.Neither dial peer is selected; the gateway rejects the call because two POTS peers cannot share the same voice port.

B.Dial peer 20, because the gateway selects the peer with the lowest preference value configured under the dial peer.

C.Dial peer 10, because the most specific match (longest explicit digit string) is selected before the wildcard pattern.

D.Dial peer 20, because wildcard patterns always take precedence over literal patterns to allow flexible routing.

AnswerC

Cisco IOS dial-peer matching prefers the most specific destination pattern. The explicit pattern 911 has no wildcards, so it is considered a longer, more specific match than 9T. The gateway selects dial peer 10 for the outbound POTS leg, ensuring emergency calls use the dedicated peer and any special configuration attached to it.

Why this answer

Dial-peer matching on Cisco IOS voice gateways uses a specificity comparison: an exact digit string outranks a pattern containing wildcards. Because 911 is fully explicit and 9T contains the T wildcard, the gateway selects the peer with destination pattern 911 for the outbound POTS leg. This behavior lets engineers attach distinct settings, such as different trunk groups or calling-party transformations, to emergency calls while still allowing all other dialed numbers to use the general pattern.

Exam trap

The trap here is assuming that a wildcard pattern or a lower preference value overrides a more specific literal pattern during dial-peer selection.

Full explanation →

233

MCQmedium

You are monitoring RTP traffic using a protocol analyzer. You observe that the Payload Type field in the RTP header is set to 0. Which codec is currently in use?

A.G.729

B.Opus

C.G.711 PCMU

D.G.711 PCMA

AnswerC

Payload Type 0 is statically mapped to G.711 µ-law (PCMU) in the RTP audio/video profile, so the analyser reading confirms PCMU is in use. The PT field is the header's codec identifier, and value 0 is reserved for this 64 kbps narrowband codec.

Why this answer

In the RTP header, the Payload Type (PT) field value of 0 indicates the use of the G.711 PCMU (μ-law) codec. This is defined in RFC 3551, which assigns static payload type 0 to PCMU. Therefore, the correct answer is G.711 PCMU.

Exam trap

CLCOR often tests the static payload type assignments, and candidates may confuse PT 0 (PCMU) with PT 8 (PCMA) or assume dynamic types for all codecs.

How to eliminate wrong answers

Option A is wrong because G.729 is assigned a dynamic payload type (typically 18) and is not represented by PT 0. Option B is wrong because Opus uses dynamic payload types (e.g., 96-127) and is not assigned a static PT. Option D is wrong because G.711 PCMA (A-law) is assigned static payload type 8, not 0.

Full explanation →

234

Multi-Selectmedium

Which TWO tasks must be performed when setting up Webex Hybrid Calendar service with Microsoft Exchange?

Select 2 answers

A.Create a dedicated DNS zone for Exchange

B.Configure a SIP trunk between Exchange and Webex

C.Install an H.323 gateway in the DMZ

D.Grant the service account ApplicationImpersonation rights

E.Enable the Hybrid Calendar service in Control Hub

AnswersD, E

ApplicationImpersonation lets the service account act on behalf of users to read and write their calendars, which the Hybrid Calendar connector requires for meeting scheduling. Without this Exchange role assignment, calendar operations fail despite correct Control Hub configuration.

Why this answer

Option D is correct because the Webex Hybrid Calendar service connects to Microsoft Exchange on-premises via Exchange Web Services (EWS) using a dedicated service account, and that account must be granted the ApplicationImpersonation management role so it can act on behalf of users to read and manage their calendar data. Option E is correct because the Hybrid Calendar service must be activated and configured in Control Hub (the Webex administrative portal), where you enable the service and supply the Exchange environment details and service account credentials that tie the on-premises deployment to the cloud service. Options A, B, and C are incorrect: no dedicated DNS zone for Exchange is required, Hybrid Calendar does not use a SIP trunk between Exchange and Webex, and no H.323 gateway in the DMZ is involved since the integration relies on HTTPS/EWS rather than telephony signaling.

Exam trap

CLCOR often tests the confusion between Hybrid Calendar (EWS/impersonation) and Hybrid Calling (SIP trunk), causing candidates to select SIP-related options for calendar questions.

Full explanation →

235

MCQeasy

A collaboration administrator is configuring a new site in Cisco Unified Communications Manager. The site has a 2 Mbps WAN link, and the administrator wants to prevent more concurrent G.711 calls than the link can support by defining the maximum audio bandwidth allowed. Which configuration element in Cisco Unified Communications Manager should the administrator configure?

A.SIP Profile

B.Device Pool

C.Location

D.Region

AnswerC

A Location defines the total audio and video bandwidth available for calls to and from devices assigned to it. When the aggregate bandwidth of existing calls plus the new call would exceed the configured limit, CUCM rejects the call with a not-enough-bandwidth cause. This is the correct object for capping concurrent calls at a site with a constrained WAN link.

Why this answer

Locations in Cisco Unified Communications Manager are the call admission control mechanism for hub-and-spoke WAN designs, tracking aggregate audio and video bandwidth per site. Assigning devices to a Location with a defined maximum audio bandwidth causes CUCM to deny calls that would exceed the limit, which is precisely what the administrator needs to protect a 2 Mbps link from oversubscription.

Exam trap

The trap here is confusing Regions, which choose codecs between sites, with Locations, which actually count and limit aggregate bandwidth for call admission control.

Full explanation →

236

MCQmedium

A Cisco IOS XE voice gateway routes inbound PSTN calls to Cisco Unified Communications Manager. The gateway is configured with dial-peer voice 100 pots, which has no incoming called-number command, and dial-peer voice 200 voip, which points to the CUCM subscriber. An engineer needs the gateway to terminate incoming POTS calls on the POTS dial peer and match outbound calls to the VoIP dial peer based on the called number. Which command must be added to dial-peer voice 100 pots to ensure inbound calls are matched correctly?

A.incoming called-number .

B.answer-address .

C.destination-pattern .

D.incoming called-number 200

AnswerA

For inbound POTS calls, the gateway matches the dial peer using the incoming called-number command. Without a specific incoming called-number, the POTS dial peer may not be selected for incoming calls. Adding incoming called-number . matches any called number, allowing the POTS dial peer to terminate the inbound call before outbound dial-peer matching occurs.

Why this answer

Inbound POTS dial peers are matched using the incoming called-number command, which examines the called number of the incoming call. Without it, the gateway may not select the POTS dial peer, leading to call failure. Using incoming called-number . matches any called number, ensuring the POTS dial peer terminates the call.

The VoIP dial peer is then used for outbound matching based on destination-pattern.

Exam trap

The trap here is confusing inbound matching commands with outbound matching commands, such as using destination-pattern or answer-address for inbound POTS calls.

Full explanation →

237

MCQmedium

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?

A.NetFlow

B.SNMP

C.IP SLA

D.SPAN

AnswerC

IP SLA actively generates synthetic probes to measure packet loss, latency, and jitter along the path to the CUCM cluster, directly satisfying the endpoint readiness requirement. Its jitter operation reports round-trip delay and inter-packet variation, giving the precise metrics needed to validate voice-quality thresholds before deploying the TelePresence endpoint.

Why this answer

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.

Exam trap

CLCOR often tests the confusion between monitoring tools like NetFlow and performance measurement tools like IP SLA, where candidates might choose NetFlow for latency measurement when it does not provide that.

How to eliminate wrong answers

Option A is wrong because NetFlow provides traffic flow data but does not measure latency and jitter directly; it's used for traffic analysis and accounting. Option B is wrong because SNMP is used for monitoring device health and interface statistics, not for measuring end-to-end network performance metrics like jitter. Option D is wrong because SPAN is used for port mirroring to capture packets for analysis, but it does not generate performance metrics; it requires manual analysis.

Full explanation →

238

MCQhard

You are designing a dial plan with a centralized PSTN egress. What must you configure to ensure that local extensions can be dialed directly across sites?

A.A shared device pool.

B.A translation pattern for each extension.

C.A CSS that includes all site-specific partitions.

D.A route pattern that includes all extensions.

AnswerC

A CSS containing every site's partitions lets each phone search all partitions, so dialling a remote site's extension matches the correct partition and routes over the centralized PSTN egress. Without the shared CSS, cross-site extension dialling fails.

Why this answer

In a centralized PSTN egress design, each site's extensions reside in site-specific partitions. To allow any site to dial another site's extensions directly, the calling devices need a Calling Search Space (CSS) that includes all site-specific partitions. This enables the dial plan to match the dialed extension in the appropriate partition and route the call internally.

Exam trap

CLCOR often tests the role of CSS and partitions in inter-site dialing; candidates may think a shared device pool or translation patterns are needed, but the correct answer is a CSS that includes all relevant partitions.

How to eliminate wrong answers

Option A is wrong because a shared device pool does not affect dialing permissions; it only groups devices for settings like region and CM group. Option B is wrong because a translation pattern per extension is not scalable and unnecessary; a CSS with partitions is the standard method. Option D is wrong because a route pattern that includes all extensions would route calls to a gateway, not enable internal extension dialing across sites; route patterns are for external destinations.

Full explanation →

239

MCQeasy

Which protocol is required for the Webex Hybrid Directory Connector to communicate with on-premises Active Directory?

A.LDAP

B.XML

C.SIP

D.SNMP

AnswerA

The Webex Hybrid Directory Connector queries on-premises Active Directory using LDAP to synchronise user and group objects into the Webex cloud. LDAP is the native directory access protocol Active Directory exposes, satisfying the connector's requirement to read directory data from the on-premises domain controllers.

Why this answer

The Webex Hybrid Directory Connector synchronizes user data from on-premises Active Directory to the Webex cloud, and it uses LDAP (Lightweight Directory Access Protocol) to query and read directory information from Active Directory. LDAP is the standard protocol for accessing and maintaining distributed directory information services.

Exam trap

CLCOR often tests the protocol used for directory synchronization, and candidates may confuse LDAP with other protocols like SIP or XML due to the hybrid context.

How to eliminate wrong answers

Option B is wrong because XML is a markup language used for data representation, not a directory access protocol, and while it may be used in some API payloads, it is not the protocol for AD communication. Option C is wrong because SIP is a signaling protocol for VoIP and real-time communication, not for directory synchronization. Option D is wrong because SNMP is used for network management and monitoring, not for querying Active Directory.

Full explanation →

240

MCQhard

You notice high latency on calls placed through a Local Gateway. Which metric should you inspect in the Webex Control Hub to confirm the issue is with the network path?

A.Network latency and jitter.

B.Calling license usage.

C.User status.

D.Device firmware version.

AnswerA

Network latency and jitter metrics in Control Hub isolate the transport path between endpoints and the Local Gateway, distinguishing path degradation from application or media-processing faults, which is exactly what the stem asks you to confirm.

Why this answer

Network latency and jitter are key metrics that measure the quality of the network path. High latency indicates delay in packet delivery, while jitter measures variation in delay, both of which directly affect voice quality. Inspecting these metrics in Webex Control Hub can confirm if the issue is with the network path.

Exam trap

The trap is selecting metrics that seem related to call quality but are not network-related, such as license usage or firmware version. Candidates must focus on the specific symptom: high latency on calls, which points directly to network metrics.

How to eliminate wrong answers

Option B is wrong because calling license usage relates to licensing consumption, not network performance. Option C is wrong because user status indicates whether a user is available or busy, not network latency. Option D is wrong because device firmware version is about software compatibility, not real-time network conditions.

Full explanation →

241

MCQeasy

What is the purpose of the 'Webex Service' account in Active Directory when using Directory Connector?

A.To encrypt the synchronization traffic.

B.To read directory objects for synchronization.

C.To manage the user's email password.

D.To bypass the firewall.

AnswerB

The Webex Service account is granted read access to directory objects so Directory Connector can synchronise users and groups into Webex. This read-only permission satisfies the stem's synchronisation purpose, without granting write or administrative rights in Active Directory.

Why this answer

The Webex Service account in Active Directory is used by Directory Connector to read (bind and query) directory objects such as users, groups, and attributes for synchronization to Webex. It requires read permissions on the directory partitions being synced. This service account is the identity Directory Connector uses to authenticate to AD and pull the data needed for user provisioning.

Exam trap

CLCOR often tests the purpose of the AD service account — candidates confuse it with encryption or password management, missing that its sole role is read access for directory synchronization.

How to eliminate wrong answers

Option A is wrong because encrypting synchronization traffic is handled by TLS/SSL between Directory Connector and Webex, not by the service account, which is an AD identity. Option C is wrong because managing a user's email password is not the service account's role — Directory Connector does not manage passwords; it reads directory attributes. Option D is wrong because bypassing the firewall is a network configuration concern (ports/proxies), not something an AD service account can do.

Full explanation →

242

Multi-Selectmedium

Which THREE network conditions contribute to poor VoIP call quality in an RTP stream?

Select 3 answers

A.Latency

B.High MTU

C.Jitter

D.Packet Loss

E.Fast switching

AnswersA, C, D

Delay causes conversation overlaps.

Why this answer

Latency (A) is correct because one-way delay above roughly 150 ms degrades RTP voice interactivity, causing talk-over and echo that the jitter buffer cannot fix. Jitter (C) is correct because variable inter-packet arrival times force the jitter buffer to either underrun (clipping) or grow (added delay), directly harming voice quality. Packet Loss (D) is correct because dropped RTP packets cannot be retransmitted in real time, so codecs conceal the gaps, producing robotic or choppy audio.

High MTU (B) is not inherently harmful—fragmentation is the real risk—and Fast switching (E) is a LAN forwarding behavior, not a WAN impairment affecting RTP quality.

Exam trap

CLCOR often tests whether candidates confuse general network performance factors (MTU, switching mode) with the specific RTP impairments defined by RFC 3550 — only latency, jitter, and packet loss belong in that list.

Full explanation →

243

MCQhard

A CUBE is configured for H.450.3 call forwarding, but it is not working. What is the most likely cause?

A.Dial-peer not defined

B.Incompatible signaling protocol

C.Codec mismatch

D.SIP profile missing

AnswerB

H.450.3 supplementary services require matching signalling on both call legs. If the CUBE's inbound and outbound dial peers negotiate different protocols, such as H.323 against SIP, the forwarding request cannot be translated, so forwarding silently fails.

Why this answer

H.450.3 is a supplementary service for call diversion (call forwarding) defined within the H.323 protocol suite. For H.450.3 to function, the signaling protocol on the dial-peer must be H.323 — it cannot work over SIP or other signaling protocols because H.450.x supplementary services are H.323-specific constructs. If the CUBE is configured with SIP signaling on the relevant dial-peers, H.450.3 call forwarding will never trigger, making incompatible signaling protocol the most likely cause.

Exam trap

CLCOR often tests the assumption that H.450.x supplementary services are protocol-agnostic; candidates pick dial-peer or codec issues when the real blocker is that H.450.3 only operates over H.323 signaling, not SIP.

How to eliminate wrong answers

Option A is wrong because a missing dial-peer would prevent the call from routing at all, not specifically break H.450.3 forwarding — and the question states forwarding is configured but 'not working,' implying routing exists. Option C is wrong because a codec mismatch causes one-way audio or media negotiation failure, not a failure of the H.450.3 supplementary service signaling. Option D is wrong because a missing SIP profile would affect SIP-specific features like early offer or header manipulation, and H.450.3 is not a SIP feature — it is an H.323 supplementary service.

Full explanation →

244

Multi-Selectmedium

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

Select 2 answers

A.Call routing in CUCM

B.Media interworking

C.IP phone registration

D.Voice mail storage

E.Topology hiding

AnswersB, E

The CUBE terminates signalling and media on separate legs, so it can transcode between dissimilar codecs and repacketise flows, letting endpoints with mismatched capabilities interoperate. This satisfies the stem's requirement to identify a function the CUBE performs as a Session Border Controller.

Why this answer

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.

Exam trap

CLCOR often tests the misconception that CUBE performs CUCM functions like call routing or phone registration, when in fact CUBE is a border element focused on interworking, security, and topology hiding.

Full explanation →

245

MCQmedium

A company has deployed Webex Hybrid Message Service to integrate its on-premises Cisco Unified Communications Manager IM and Presence Service with Webex. Users complain that their presence status is not updating in Webex when they change it in the on-premises Jabber client. The administrator confirms that the Expressway-C and Expressway-E are registered and that the Hybrid Message Service connector is running. Which component should the administrator verify next to ensure presence federation is working?

A.Cisco Unity Connection voicemail ports

B.Webex Control Hub user account license assignment

C.Cisco Unified Communications Manager IM and Presence Service SIP federation configuration

D.Cisco Expressway-C certificate trust store for HTTPS

AnswerC

Hybrid Message Service relies on SIP federation between the on-premises IM and Presence Service and the Webex cloud. If the SIP federation is misconfigured or the federation routing is not properly set up, presence updates will not be exchanged even though the connectors are healthy. Verifying the SIP federation configuration on IM and Presence is the logical next step to isolate the presence synchronization issue.

Why this answer

Hybrid Message Service uses SIP federation to connect the on-premises IM and Presence Service with the Webex cloud. Presence information flows over this federation, so if presence updates are not appearing, the SIP federation configuration on IM and Presence is the most likely culprit. The Expressway connectors and Hybrid Message Service connector only provide the transport and service registration; they do not replace the need for correct SIP federation settings.

Exam trap

The trap here is focusing on the hybrid connector health and overlooking that presence federation specifically requires SIP federation to be configured and operational on the on-premises IM and Presence Service.

Full explanation →

246

MCQmedium

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?

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

B.Configure CBWFQ with a bandwidth guarantee per branch class and add LLQ for voice in the same policy-map.

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

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

AnswerB

Class-based weighted fair queueing gives each branch class a guaranteed minimum bandwidth and shares excess proportionally, which prevents one branch from monopolizing the link. Adding low-latency queueing for voice within the same policy-map ensures voice bearer traffic gets strict-priority treatment, satisfying both the fairness and voice-priority requirements in this hub-and-spoke design.

Why this answer

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.

Exam trap

The trap here is treating priority queueing or shaping alone as sufficient for fairness across many branches.

Full explanation →

247

Multi-Selectmedium

Which TWO of the following are considered best practices for implementing QoS in a Cisco collaboration network?

Select 2 answers

A.Mark traffic as close to the source as possible

B.Trust all incoming traffic by default

C.Only mark traffic at the WAN edge

D.Extend the trust boundary to the access layer switch

E.Disable all QoS policies to improve performance

AnswersA, D

Marking at the source lets the originating endpoint or access switch set DSCP values before any congestion or queuing occurs, so downstream devices can classify and prioritise on trusted markings rather than re-inspecting every flow. This satisfies the stem's best-practise requirement by preserving classification integrity end to end.

Why this answer

Option A is correct because Cisco QoS best practice is to classify and mark traffic as close to the source as possible, typically on the endpoint or the first-hop access switch, so that the DSCP value is preserved end-to-end and every downstream device can apply consistent per-hop behavior without re-classifying. Option D is correct because extending the trust boundary to the access layer switch allows that switch to trust the DSCP markings from Cisco IP phones and other endpoints, enforcing classification at the network edge rather than re-marking later. Option B is wrong because trusting all incoming traffic by default lets users or rogue devices set arbitrary DSCP values, which can abuse priority queues and break QoS guarantees.

Option C is wrong because marking only at the WAN edge is too late—traffic has already traversed the LAN unclassified, and the WAN edge cannot reliably distinguish applications after congestion or encryption. Option E is wrong because disabling QoS removes classification, queuing, and policing, which degrades voice and video quality rather than improving performance.

Exam trap

CLCOR often tests whether candidates understand the trust boundary concept — the trap is thinking QoS marking should happen only at the WAN edge or that trusting all traffic simplifies design.

Full explanation →

248

MCQeasy

Which Cisco feature is used to ensure voice traffic is prioritized over web traffic on a Cisco access switch?

A.SNMP Traps

B.Quality of Service (QoS)

C.Spanning Tree Protocol

D.VLAN Trunking

AnswerB

Quality of Service (QoS) classifies and marks voice packets, then schedules them into a priority queue ahead of web traffic, satisfying the requirement to prioritise voice on the access switch. Cisco access switches use QoS mechanisms such as classification, marking, and queuing to guarantee voice gets strict priority treatment.

Why this answer

Quality of Service (QoS) is the Cisco feature set used to classify, mark, and prioritize voice traffic over less delay-sensitive traffic like web browsing on an access switch. QoS mechanisms such as classification, marking (DSCP EF for voice), queuing, and policing ensure voice packets are forwarded with low latency and jitter. This is the standard answer for prioritizing voice over data on a Cisco access switch.

Exam trap

CLCOR often tests the misconception that VLANs or STP provide prioritization, when only QoS mechanisms actually classify and prioritize voice traffic.

How to eliminate wrong answers

Option A is wrong because SNMP Traps are used for monitoring and alerting, not for traffic prioritization. Option C is wrong because Spanning Tree Protocol prevents Layer 2 loops and manages redundant paths, but does not prioritize traffic types. Option D is wrong because VLAN Trunking (802.1Q) carries multiple VLANs over a link but does not itself provide prioritization; QoS must still be applied to prioritize voice within or across VLANs.

Full explanation →

249

Multi-Selecteasy

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

Select 2 answers

A.An on-premises Active Directory server.

B.A Cisco Unified Border Element (CUBE).

C.A SIP trunk connecting to Webex Calling.

D.A dedicated fiber-optic internet connection.

E.A cloud-based PSTN provider.

AnswersB, C

A Cisco Unified Border Element provides the session border controller function that a Cisco Local Gateway deployment requires, interworking SIP signalling and media between the enterprise collaboration environment and the external PSTN or ITSP trunk. Without this demarcation and protocol-normalisation role, the Local Gateway cannot terminate or secure the external voice connection.

Why this answer

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.

Exam trap

CLCOR often tests component roles — candidates pick Active Directory or a cloud PSTN provider because they sound like 'required' enterprise components, missing that CUBE and the Webex Calling SIP trunk are the two mandatory LGW building blocks.

Full explanation →

250

Multi-Selecthard

Which THREE of the following items are verified during the Webex Hybrid Service connector installation? (Choose three)

Select 3 answers

A.The local PBX manufacturer.

B.Active Directory service account privileges.

C.User's home network speed.

D.Outbound HTTPS connection to the cloud.

E.SSL/TLS certificate trust chain.

AnswersB, D, E

The installer confirms the specified Active Directory service account holds the required privileges to read and synchronise directory objects for the hybrid service. Insufficient rights cause verification failure, satisfying the directory-access prerequisite checked during connector setup.

Why this answer

During the Webex Hybrid Service connector installation, the installer validates the Active Directory service account privileges (B) because the connector needs a properly privileged AD account to sync users and groups to the Webex cloud. It also verifies the outbound HTTPS connection to the cloud (D), since the connector must reach Webex services over TCP 443 for registration and ongoing communication. The SSL/TLS certificate trust chain (E) is checked as well, ensuring the connector trusts the cloud endpoint's certificate and can establish secure TLS sessions.

The local PBX manufacturer (A) is not verified by the connector installer, as it is irrelevant to connector prerequisites, and the user's home network speed (C) is not tested because the connector is a server-side component whose bandwidth requirements are not validated during installation.

Exam trap

The trap is selecting plausible-sounding but irrelevant items like PBX manufacturer or home network speed — the exam expects candidates to know that the connector validates AD privileges, cloud HTTPS reachability, and certificate trust, not physical PBX or user-side factors.

Full explanation →

251

MCQhard

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

A.They must be configured with a Unified Communications traversal zone.

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

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

D.They must be in the same subnet.

AnswerA

The 'Unified Communications' traversal zone is mandatory for Hybrid Call Service interworking.

Why this answer

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.

Exam trap

CLCOR often tests the misconception that Expressway-C needs a public IP or that both nodes must be in the same subnet, when in fact Expressway-C is internal and Expressway-E is in the DMZ with a public IP.

How to eliminate wrong answers

Option B is wrong because while certificates are important for trust, the Expressway pair does not need to use the same certificate authority as CUCM nodes; they can use different CAs as long as trust is established. Option C is wrong because Expressway-C is deployed in the internal network and does not require a public IP address; only Expressway-E needs a public IP for external connectivity. Option D is wrong because Expressway-C and Expressway-E are typically deployed in different network segments (internal vs.

DMZ), not the same subnet.

Full explanation →

252

MCQmedium

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

A.media flow-around

B.media pass-through

C.media-bypass

D.media flow-through

AnswerA

The media flow-around command configures the CUBE to pass media directly between endpoints rather than anchoring it through the gateway. This satisfies the requirement that RTP flows bypass the CUBE, leaving only signalling processed by the router.

Why this answer

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.

Exam trap

CLCOR often tests the confusion between 'media flow-around' and 'media flow-through' — candidates may pick 'flow-through' thinking it means bypass, but it actually anchors media on CUBE.

How to eliminate wrong answers

Option B is wrong because 'media pass-through' is not a valid CUBE command; the actual command is 'media flow-through' which forces media to traverse the CUBE. Option C is wrong because 'media-bypass' is not a recognized Cisco IOS voice command. Option D is wrong because 'media flow-through' does the opposite — it anchors media on the CUBE rather than allowing direct endpoint-to-endpoint flow.

Full explanation →

253

MCQmedium

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?

A.Cisco SIP Proxy

B.Cisco Presence Engine

C.Cisco XCP Router

D.Cisco TFTP

AnswerB

The Cisco Presence Engine is the service that aggregates and publishes user availability state, so it must run for users to see each other's status. Without it, the IM and Presence server handles messaging but cannot supply presence information.

Why this answer

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.

Exam trap

CLCOR often tests the misconception that the XCP Router or SIP Proxy provides presence, when the Presence Engine is the specific service responsible for presence state processing.

How to eliminate wrong answers

Option A is wrong because Cisco SIP Proxy is not the presence-processing service; it handles SIP signaling for the IM and Presence node but does not itself manage presence state. Option C is wrong because Cisco XCP Router is the underlying messaging/connection manager that routes XMPP traffic between services, but it is not the service that specifically implements presence logic. Option D is wrong because Cisco TFTP is a CUCM service for delivering configuration files and firmware to endpoints, unrelated to IM and Presence.

Full explanation →

254

MCQhard

An administrator is deploying Webex Hybrid Calendar Service with an on-premises Microsoft Exchange 2019 environment. Users report that their Webex meetings are not being automatically added to their Outlook calendars. The administrator has already verified that the Expressway-C and Expressway-E connectors are registered and healthy in Control Hub. Which additional component must be installed and configured to enable the calendar integration?

A.Webex Hybrid Calendar Service connector (Calendar Connector) on a Windows server

B.Webex Hybrid Media Service node on the Expressway-C

C.Cisco Directory Connector on a domain-joined Windows server

D.Cisco Expressway-E with a public TLS certificate

AnswerA

The Calendar Connector is a separate on-premises component that communicates with the Exchange environment to read and write calendar items, and it registers with the Webex cloud to enable Hybrid Calendar Service. Even when Expressway connectors are healthy, the Calendar Connector is required for Exchange integration, so its absence explains why meetings are not appearing in Outlook calendars.

Why this answer

Hybrid Calendar Service requires a dedicated Calendar Connector to integrate with the on-premises Exchange environment. The Expressway pair handles secure connectivity to the Webex cloud, but the Calendar Connector is the component that actually reads and writes Exchange calendar data to insert Webex meeting details. Without it, meetings will not appear on users' Outlook calendars even if the Expressway connectors are healthy.

Exam trap

The trap here is assuming that healthy Expressway connectors are sufficient for all hybrid services, when each hybrid service such as Calendar requires its own dedicated on-premises connector.

Full explanation →

255

Multi-Selecthard

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.)

Select 2 answers

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

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

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

D.Use the show class-map command to confirm the match criteria and verify the video class is referencing the correct DSCP or ACL.

E.Run the show policy-map interface command to view per-class packet counts and drop statistics on the WAN interface.

AnswersD, E

The show class-map command displays the match statements configured in each class-map. Verifying that the video class matches the correct DSCP value, such as AF41, or the correct access list confirms that the classification logic itself is correct before relying on interface statistics, which is essential when video is not being placed in the expected queue.

Why this answer

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.

Exam trap

The trap here is assuming that call-level or interface-level commands reveal per-class QoS treatment when only policy and class-map output does.

Full explanation →

256

MCQeasy

In a Cisco CUCM cluster, which node is responsible for managing the database and performing configuration updates?

A.Publisher

B.Subscriber

C.TFTP node

D.Music on Hold (MoH) node

AnswerA

The Publisher is the sole node holding the master, read-write database. All configuration changes are written there first, then replicated to subscribers. This makes it the authoritative source for database management and configuration updates across the CUCM cluster.

Why this answer

The Publisher node in a CUCM cluster is the only node that owns the master database and is responsible for all configuration changes. Subscribers receive replicated copies of the database and are read-only for configuration purposes.

Exam trap

CLCOR often tests the assumption that any node can perform configuration changes, when only the Publisher has write access to the master database.

How to eliminate wrong answers

Option B is wrong because Subscribers only maintain replicated copies of the database and cannot perform configuration updates; they serve call processing and redundancy. Option C is wrong because the TFTP node (often colocated with the Publisher) only serves configuration files and firmware to endpoints, not database management. Option D is wrong because the MoH node only streams music on hold audio streams and has no database or configuration role.

Full explanation →

257

MCQmedium

A network engineer is configuring LLQ on a Cisco IOS router to protect voice traffic. The voice class is assigned a priority of 512 kbps. During periods of congestion, which mechanism ensures that voice packets are not delayed by lower-priority traffic?

A.Weighted fair queueing allocates bandwidth equally among all flows, including voice.

B.Traffic shaping buffers excess voice packets and sends them later.

C.The priority queue is serviced only when other queues are empty, but no policer is applied.

D.The priority queue is serviced before other queues, and a policer drops packets that exceed the configured rate.

AnswerD

LLQ creates a strict priority queue that is always serviced first, ensuring minimal delay for voice. A built-in policer enforces the configured rate, dropping excess packets to prevent the priority queue from starving other queues. This combination provides low latency and jitter for voice while maintaining fairness for other traffic.

Why this answer

LLQ provides a strict priority queue that is serviced before other queues, ensuring voice packets experience minimal delay and jitter. The built-in policer enforces the configured bandwidth limit, dropping excess packets to prevent the priority queue from consuming all bandwidth and starving other traffic. This dual mechanism makes LLQ ideal for real-time voice.

Exam trap

The trap here is assuming that LLQ without a policer is sufficient, but the policer is essential to prevent voice from starving other queues.

Full explanation →

258

MCQmedium

In a Webex hybrid environment, what is the primary role of the Expressway-C?

A.To manage public DNS records

B.To act as a SIP proxy for internal phones

C.To provide a secure connector for Hybrid Services

D.To replace the CUCM server

AnswerC

Expressway-C establishes outbound-only connections to Webex cloud services, so no inbound firewall ports are opened. This satisfies the hybrid constraint: it registers and proxies on-premises resources—call connectors, calendar connectors, messaging—through a secure tunnel, with Expressway-E handling external traversal. That connector role is its primary function in hybrid deployments.

Why this answer

Expressway-C is deployed inside the enterprise network and acts as the secure connector that initiates outbound-only connections to Expressway-E, which then relays traffic to Webex cloud services. This connector role is what enables Hybrid Services (Calendar, Message, Directory, Management, Call) to traverse the firewall without inbound pinholes. It does not manage DNS or replace CUCM; it is purely a traversal/connector component in the hybrid architecture.

Exam trap

CLCOR often tests the distinction between Expressway-C (internal secure connector) and Expressway-E (DMZ traversal server), causing candidates to confuse the two roles or assume Expressway-C handles DNS/SIP duties it does not own.

How to eliminate wrong answers

Option A is wrong because public DNS records are managed by the organization's DNS provider or Cisco Webex's cloud DNS, not by Expressway-C. Option B is wrong because while Expressway-C can proxy SIP for Mobile and Remote Access (MRA) registrations, its primary role in the Hybrid Services context is the secure connector, not a generic SIP proxy for internal phones. Option D is wrong because CUCM remains the on-premises call control; Expressway-C never replaces CUCM—it complements it by brokering hybrid service traffic.

Full explanation →

259

MCQeasy

Which component in CUBE is responsible for ensuring that the call session is terminated correctly?

A.Dial-peer

B.Voice translation-profile

C.SIP Stack

D.Voice-class codec

AnswerC

The SIP Stack handles SIP signalling messages, including BYE requests that tear down sessions. It ensures the call session terminates correctly by processing the BYE transaction and confirming the dialog ends cleanly, satisfying the stem's requirement for correct session termination within CUBE.

Why this answer

The SIP Stack in CUBE is the protocol engine that handles SIP signaling end-to-end, including session setup, in-dialog requests, and proper session teardown via BYE transactions. It ensures that call sessions are terminated correctly by managing the SIP dialog state machine and sending/receiving the appropriate messages.

Exam trap

CLCOR often tests the distinction between signaling-plane components (SIP stack) and policy/routing components (dial-peers, translation profiles) — candidates pick dial-peer because it is the most visible CUBE construct.

How to eliminate wrong answers

Option A is wrong because a dial-peer is a routing construct that matches inbound/outbound calls and applies policies; it does not manage SIP dialog state or session teardown. Option B is wrong because a voice translation-profile manipulates calling/called numbers and redirect numbers — it has no role in session termination. Option D is wrong because a voice-class codec defines media codec preferences for SDP negotiation, not signaling session lifecycle.

Full explanation →

Cisco Implementing Cisco Collaboration Core Technologies (CLCOR, 350-801, v2.0 blueprint, CCNP Collaboration) (CLCOR) (CLCOR) — Questions 226–259