Cisco 350-801 Exam Dumps & Practice Test Questions

Question 1:

In the Cisco Collaboration architecture, what are two primary roles of Cisco Expressway when enabling secure communication and remote connectivity for users? (Choose two.)

A. Provides telephony redundancy for remote branch sites
B. Registers MGCP-based gateways to CUCM
C. Enables secure external communication between businesses
D. Manages customer support and contact center interactions
E. Facilitates secure firewall traversal for remote access

Correct Answers: C and E

Explanation:

Cisco Expressway is a foundational component of the Cisco Collaboration architecture, designed to extend unified communications securely beyond the internal enterprise network. It plays a significant role in enabling secure collaboration between businesses and providing remote connectivity for users across firewalls without requiring traditional VPN setups.

One of its core capabilities is secure business-to-business (B2B) communication (Option C). Cisco Expressway allows organizations to connect with external entities—such as partners, vendors, or clients—using standards-based protocols and encrypted signaling/media. It supports secure SIP trunking and federation, making it possible for companies to establish voice, video, messaging, and conferencing services with external parties while still maintaining robust security postures.

Another central function is secure firewall traversal for remote devices (Option E). Cisco Expressway consists of two components: Expressway-C (Core) and Expressway-E (Edge). Expressway-C resides within the enterprise network and communicates with collaboration services like Cisco Unified Communications Manager (CUCM), while Expressway-E is placed in the DMZ and interfaces with external users or devices. This configuration enables mobile users—such as teleworkers or employees using smartphones and tablets—to access collaboration services like voice, video, instant messaging, and voicemail without needing to establish a VPN. The firewall traversal is handled transparently and securely, simplifying remote access while preserving security policies.

Now, let’s look at the incorrect options:

• A (Survivable Remote Site Telephony): This is handled by Cisco SRST, which ensures continued telephony services at branch sites during WAN failures. It is not part of Cisco Expressway's function.

• B (MGCP Gateway Registration): MGCP (Media Gateway Control Protocol) is typically managed directly by CUCM for voice gateways. Cisco Expressway does not manage gateway registration.

• D (Customer Interaction Management Services): While valuable, this falls under Cisco Contact Center solutions, not Cisco Expressway. Expressway’s focus is not on managing customer interactions but rather on enabling secure connectivity.

In summary, Cisco Expressway is designed to extend unified communications to external users and systems securely, through B2B collaboration and firewall traversal, making C and E the correct choices.

Question 2:

A network engineer is assigned to allow employees to use their personal mobile devices to connect to the corporate phone system via the internet, with minimal configuration required on the user end. 

Which solution best addresses this goal?

A. Cisco Express Mobility
B. Cisco Expressway-C and Expressway-E
C. Cisco Unified Border Element (CUBE)
D. Cisco Unified IM and Presence

Correct Answer: B

Explanation:

When an organization seeks to extend its unified communications system to employees using their own mobile devices over the internet, Cisco Expressway-C and Expressway-E offer an ideal solution that ensures secure, simple, and efficient access. These two components are explicitly designed for firewall traversal and remote collaboration access without requiring users to configure complex VPNs or network settings.

Cisco Expressway-C, deployed inside the enterprise network, interfaces directly with collaboration applications such as Cisco Unified Communications Manager (CUCM), Unity Connection, and IM and Presence. It manages internal signaling and serves as the central hub for secure communication routing.

Cisco Expressway-E, positioned in the DMZ or outside the enterprise firewall, establishes the secure interface between external users and the internal collaboration system. It handles NAT, security enforcement, and encryption, enabling remote endpoints to register and communicate securely as if they were physically located on the internal network.

Together, Expressway-C and Expressway-E simplify remote connectivity, allowing mobile users to leverage applications like Cisco Jabber or Webex without manual configurations or VPN tunnels. Users can access voice, video, messaging, voicemail, and directory services by simply launching an app—an experience that meets the requirement of low setup complexity.

Reviewing the other options:

• A (Cisco Express Mobility): This feature enables seamless Layer 3 roaming in wireless LAN environments but is not designed to facilitate remote telephony access over the internet.

• C (Cisco Unified Border Element): CUBE serves as a SIP gateway, interconnecting enterprise and service provider voice networks. It does not handle remote user access or simplify mobile device configuration.

• D (Cisco Unified IM and Presence): While this supports chat, presence status, and federation, it does not enable remote phone registration or firewall traversal. It is a complementary service, not a standalone remote access solution.

In summary, Cisco Expressway-C and Expressway-E form the most effective solution for secure, minimal-configuration remote connectivity, making B the correct answer.

Question 3:

A large enterprise plans to host an immersive video conference involving five geographically dispersed offices, each equipped with Cisco TelePresence IX5000 Series systems. To ensure a high-quality, seamless telepresence experience across all sites — with synchronized video, spatial audio, and multiple display streams.

Which media resource must be included in the design to fully support multi-stream, multi-screen conferencing?

A. Cisco PVDM4-128 (Packet Voice DSP Module)
B. Software conference bridge via Cisco Unified Communications Manager
C. Cisco Webex Meetings Server (on-premises deployment)
D. Cisco Meeting Server (CMS)

Correct Answer: D

Explanation:

When deploying a multi-site immersive video conferencing solution using Cisco TelePresence IX5000 Series systems, the media infrastructure must be capable of handling rich, high-fidelity media streams, including multiple synchronized video feeds and spatial audio. Each IX5000 system is equipped with three large displays and high-definition cameras to deliver a lifelike, immersive experience. This level of sophistication requires a media resource specifically engineered for such environments.

The Cisco Meeting Server (CMS) is the only option that meets the necessary requirements. CMS is designed to handle complex, multi-screen immersive conferences by supporting features like continuous presence, dynamic video layouts, and advanced media composition. It ensures all participants see each other as if they were in the same physical room, regardless of their geographical location. CMS manages multiple simultaneous streams from each IX5000 endpoint and stitches them into coherent, immersive video sessions.

Let’s examine why the other options are insufficient:

• Option A (Cisco PVDM4-128) is a Digital Signal Processor (DSP) module primarily used for voice processing and basic media transcoding. It lacks the advanced video stream management and scalability required for immersive systems like the IX5000.

• Option B, a software-based conference bridge on Cisco Unified Communications Manager (CUCM), is designed for audio or basic video conferencing. It doesn’t support the multi-stream immersive capabilities needed for TelePresence.

• Option C, Cisco Webex Meetings Server, is targeted toward web-based, general-purpose video meetings. While it supports video, it lacks the deep integration and specialized handling of immersive TelePresence video streams required for an IX5000 deployment.

Therefore, Cisco Meeting Server is essential in this design to unlock the full immersive capabilities of the IX5000 endpoints. CMS provides seamless conferencing with intelligent video switching, precise stream synchronization, and layout optimization — making it the only viable solution to deliver the immersive experience the client expects.

Question 4:

An engineer is designing a SIP-based voice network that includes two Cisco Unified Border Element (CUBE) routers to manage SIP traffic. The design requires cube1.abc.com to handle 60% of the calls and cube2.abc.com to handle 40%. DNS A records already exist for both. 

To achieve this traffic distribution using DNS, what two SRV record entries should be created?

A. _sip._udp.abc.com 60 IN SRV 2 60 5060 cube1.abc.com
B. _sip._udp.abc.com 60 IN SRV 60 1 5060 cube1.abc.com
C. _sip._udp.abc.com 60 IN SRV 1 40 5060 cube2.abc.com
D. _sip._udp.abc.com 60 IN SRV 3 60 5060 cube2.abc.com
E. _sip._udp.abc.com 60 IN SRV 1 60 5060 cube1.abc.com

Correct Answers: E and C

Explanation:

DNS SRV records are widely used in SIP-based environments to distribute load among multiple servers. In this scenario, the requirement is to direct 60% of SIP call traffic to cube1.abc.com and 40% to cube2.abc.com. DNS SRV records include key parameters such as priority and weight that influence how clients select which server to contact.

Priority defines the order in which servers are contacted. A lower number indicates higher priority. However, since we want both routers to share the load rather than one being a backup, the same priority value (e.g., 1) should be used for both SRV records.
Weight controls the probability that a server will be chosen among those with equal priority. The higher the weight, the more frequently that server is selected.

To meet the required load distribution:

• cube1.abc.com needs an SRV record with priority 1 and weight 60.

• cube2.abc.com needs an SRV record with priority 1 and weight 40.

This means:

• Option E correctly configures cube1 with _sip._udp.abc.com 60 IN SRV 1 60 5060 cube1.abc.com

• Option C correctly configures cube2 with _sip._udp.abc.com 60 IN SRV 1 40 5060 cube2.abc.com

Together, these entries ensure SIP clients will probabilistically select cube1 60% of the time and cube2 40% of the time, based on the defined weights. This is a commonly used method for achieving load balancing without requiring more complex hardware-based solutions.

Incorrect options misconfigure either the priority (e.g., A and D use values other than 1, which implies failover behavior) or swap the weight and priority fields (as in B), which results in misbehavior and prevents the intended load distribution.

In summary, using matching priorities and weighted SRV records is the correct and standards-based method for achieving SIP load balancing across multiple CUBE routers in this design.

Question 5:

A network engineer is setting up Cisco Expressway in a collaboration environment to support secure communications between internal and external users and to ensure interoperability across signaling protocols.

Which two features are core capabilities offered by the Cisco Expressway platform to enhance unified collaboration and compatibility? (Choose two.)

A) Seamless interworking between SIP and H.323 signaling protocols
B) Centralized endpoint registration for internal and external devices
C) Support for Intercluster Extension Mobility across different CUCM clusters
D) Native voice and video transcoding services for media adaptation
E) Full-featured voice and video conferencing capabilities for multi-party calls

Correct Answers: B and A

Explanation:

The Cisco Expressway Series, which includes Expressway-C (Core) and Expressway-E (Edge), plays a pivotal role in extending enterprise collaboration beyond traditional boundaries. Its design allows secure communication between internal and remote users without the need for a VPN, while also facilitating protocol interoperability — especially critical in mixed environments.

One of the standout features of the Expressway platform is protocol interworking. Many enterprises operate with legacy H.323-based video endpoints while also adopting modern SIP-based systems. The Expressway Series can seamlessly translate signaling between SIP and H.323, ensuring these devices communicate without manual conversion or gateway devices. This capability, Option A, is crucial for organizations undergoing gradual protocol migrations.

Another core function is Mobile and Remote Access (MRA), which allows Cisco Jabber, Webex clients, and supported endpoints to register to Cisco Unified Communications Manager (CUCM) even when they are offsite. This secure registration uses HTTPS signaling through Expressway-E and Expressway-C to eliminate VPN dependency. This validates Option B, as the Expressway architecture centralizes secure endpoint registration for both internal and external devices.

As for the incorrect options:

• Option C (Intercluster Extension Mobility): This is a CUCM-native feature that enables user roaming across clusters but is not managed by Expressway.

• Option D (Transcoding): Media adaptation, such as codec transcoding, is handled by DSP resources or Media Termination Points, not by Expressway.

• Option E (Multi-party conferencing): While Expressway can route media, the media mixing and conferencing features are provided by Cisco Meeting Server (CMS) or TelePresence Server, not Expressway.

In summary, Cisco Expressway’s strengths lie in secure external access and signaling protocol interworking, making Options A and B the correct choices. These features significantly enhance hybrid work enablement and multi-protocol environments without relying on legacy VPN infrastructures or introducing media processing overhead.

Question 6:

A collaboration engineer is troubleshooting a call failure issue where an incoming call from the PSTN, using the G.711 codec, cannot connect to an internal IP phone that only supports the G.729 codec. The mismatch in codecs prevents the call from completing.

Which media resource must be implemented on both CUBE and CUCM to convert between these codecs and allow the call to proceed?

A) Transcoder
B) Conference Bridge (CFB)
C) Music on Hold (MOH) server
D) Media Termination Point (MTP)

Correct Answer: A

Explanation:

In unified communications environments, successful media negotiation between endpoints depends heavily on codec compatibility. In this scenario, a mismatch exists where a call from the PSTN arrives using the G.711 codec, but the recipient IP phone only supports G.729. Since these two codecs are incompatible without media conversion, the call fails due to the inability to establish a mutual codec for RTP (Real-time Transport Protocol) communication.

The solution lies in deploying a transcoder — a special media resource that can dynamically convert audio streams from one codec to another. In this case, it will convert G.711 to G.729 (or vice versa), depending on call direction. Transcoders are essential in scenarios involving low-bandwidth endpoints, codec-limited IP phones, or interworking between systems with different codec policies.

Transcoders can be deployed in hardware using Digital Signal Processors (DSPs) or virtually in some environments. On Cisco platforms, transcoders must be configured and registered with CUCM and CUBE to be available during call setup. When a codec mismatch is detected during SIP call negotiation, CUCM or CUBE inserts the transcoder into the media path to resolve the conflict, allowing the call to continue successfully.

The other answer options do not serve this purpose:

• Option B (Conference Bridge): Used to mix multiple audio streams in conference calls, not for translating codecs.

• Option C (Music on Hold Server): Plays audio to callers on hold but has no role in codec conversion.

• Option D (Media Termination Point): Used to facilitate DTMF relay, early offer scenarios, and protocol interworking, but not for codec transcoding.

In conclusion, the correct media resource to resolve this codec incompatibility is a transcoder. Without it, communication between endpoints using incompatible codecs such as G.711 and G.729 would be impossible. Proper configuration on both CUCM and CUBE ensures that this resource is available in the call path when needed.

Question 7:

What service parameter must a Cisco engineer enable in CUCM to ensure that a multiparty ad hoc conference call is automatically terminated when the original creator disconnects?

A. Drop Ad Hoc Conference
B. H.225 Block Setup Destination
C. Block OffNet To OffNet Transfer
D. Enterprise Feature Access Code for Conference

Correct Answer: A

Explanation:

In a Cisco Unified Communications Manager (CUCM) environment, an ad hoc conference is a spontaneous multi-user voice session initiated by a user on the fly, typically without prior scheduling. A common concern for administrators is ensuring that such conferences don't continue indefinitely once the original initiator has left the call. To address this, Cisco CUCM offers a configurable service parameter called "Drop Ad Hoc Conference."

By enabling the Drop Ad Hoc Conference parameter, CUCM ensures that when the person who initiated the conference hangs up, the entire session is terminated for all participants. This is particularly beneficial for maintaining system resource efficiency (such as freeing up conference bridge capacity) and preventing unmonitored conversations from persisting.

This behavior is useful in organizations that require tight control over conference calls due to compliance, privacy, or bandwidth management policies. It also simplifies user expectations: when the meeting host leaves, the call ends for everyone.

Let’s review why the other options are not applicable:

• B. H.225 Block Setup Destination: This is related to H.323 protocol call signaling and is used to block specific call destinations. It has no influence on the behavior of ad hoc conferences.

• C. Block OffNet To OffNet Transfer: This setting prevents users from transferring a call from one external endpoint to another (off-net), helping prevent toll fraud. It does not manage conferencing behavior.

• D. Enterprise Feature Access Code for Conference: This allows users to dial a specific code to initiate a conference. While it facilitates conference creation, it does not affect what happens if the conference host disconnects.

Configuration Steps:

1. Log into Cisco Unified CM Administration.

2. Navigate to System > Service Parameters.

3. Select the server running the Cisco CallManager service.

4. Locate Drop Ad Hoc Conference and set it to True.

5. Save the configuration.

In conclusion, enabling Drop Ad Hoc Conference ensures that ad hoc conference calls end when the initiating party leaves, aligning with organizational policies and avoiding unnecessary use of resources.

Question 8:

After removing a user from the LDAP directory, the user still appears in CUCM as an “Inactive LDAP Synchronized User.” What should the administrator do to completely remove the user from CUCM?

A. Manually delete the user from CUCM
B. Restart the DirSync service
C. Trigger a manual LDAP synchronization
D. Wait for the garbage collection process

Correct Answer: C

Explanation:

In environments where Cisco Unified Communications Manager (CUCM) is integrated with an external LDAP directory—like Microsoft Active Directory—user management is typically handled through LDAP synchronization. When an account is deleted in the LDAP directory, CUCM doesn’t immediately remove the user. Instead, that account is flagged as an “Inactive LDAP Synchronized User.”

This inactive status means CUCM has detected that the user no longer exists in the LDAP source, but the record is retained until a formal cleanup occurs. To expedite the removal of this user from the CUCM user database, administrators must manually trigger a directory synchronization.

Manual LDAP synchronization updates the CUCM database with the current state of the LDAP directory. When it completes, CUCM removes entries for users that no longer exist in the external LDAP source. This step is critical for maintaining database hygiene, avoiding potential licensing issues, and ensuring that only active users are consuming resources.

Why the other options don’t work:

• A. Manually delete the user directly from CUCM: Not possible for LDAP-synced users, as CUCM locks their profiles from manual editing or deletion.

• B. Restarting the DirSync service: This merely restarts the background service and does not trigger a sync or user cleanup.

• D. Wait 24 hours: CUCM has an automated garbage collection process, but it’s not immediate and can be unpredictable depending on system schedules and load. Relying on it is not recommended for timely user removal.

Steps to Perform a Manual Sync:

1. Log in to Cisco Unified CM Administration.

2. Go to System > LDAP > LDAP Directory.

3. Locate the appropriate LDAP configuration profile.

4. Click Perform Full Sync Now (or Synchronize Now, depending on your CUCM version).

Once the sync is complete, any users no longer found in LDAP will be purged from CUCM. This ensures the system reflects the current state of the directory and improves user record accuracy.

To summarize, the proper way to remove a deleted LDAP user from CUCM is to initiate a manual directory synchronization.

Question 9:

A customer uses Cisco Unity Connection with LDAP integration via Active Directory to manage voicemail user details. As the Unity Connection administrator, you’ve been asked to update the first name of a user. 

Since Unity Connection synchronizes user data from LDAP, where should you make the change to ensure it correctly appears in Unity Connection?

A) In Cisco Unity Connection
B) In Cisco Unified Communications Manager end user profile
C) In Active Directory
D) In Cisco IM and Presence Server

Correct Answer: C

Explanation:

In environments where Cisco Unity Connection is integrated with LDAP (commonly Microsoft Active Directory), user account data such as first and last names, email addresses, and other personal attributes are not managed directly within Unity Connection. Instead, this information is synchronized from the external LDAP directory.

When LDAP synchronization is enabled, Unity Connection treats the user identity fields as read-only. This means fields like First Name, Last Name, or Display Name cannot be changed directly through the Unity Connection administrative interface. They appear grayed out, preventing any edits within Unity Connection. The rationale is to enforce centralized user management, ensuring all Cisco collaboration components use a single source of truth—LDAP.

To update a user’s first name:

1. The administrator must open Active Directory Users and Computers on a domain controller.

2. Locate and edit the relevant user object.

3. Update the First Name field in the user’s profile.

4. Save the changes.

Once the update is made in Active Directory, Cisco Unity Connection will reflect the change automatically during the next scheduled LDAP synchronization. If the change needs to appear immediately, the administrator can manually trigger a full directory sync from the Unity Connection interface.

The other options are incorrect for the following reasons:

• A (Cisco Unity Connection): LDAP-synced fields cannot be edited directly in Unity Connection.

• B (CUCM): Although CUCM may also use LDAP for its own user synchronization, changes made there do not propagate back to Unity Connection. The authoritative source must be the LDAP directory.

• D (IM and Presence): This component handles messaging and presence functions, not voicemail user identity.

In conclusion, when Unity Connection is LDAP-integrated, all changes to user identity data—such as modifying a user's first name—must be performed in Active Directory, which then synchronizes the updated information into Unity Connection automatically or via a manual sync.

Question 10:

An engineer is deploying Cisco SIP-based IP phones in a CUCM-managed network. To ensure accurate time display on the phones, they must synchronize with a specific time server. 

What configuration is required to make sure the phones receive time updates from the correct source?

A) Define a Phone NTP Reference in the associated Date/Time Group
B) Place the device in the appropriate Region configuration
C) Change the time format from 24-hour to 12-hour display
D) Modify the time zone setting to Etc/GMT+8

Correct Answer: A

Explanation:

Cisco SIP phones rely on CUCM for time synchronization, but not directly from the CUCM server itself. Instead, phones reference a Network Time Protocol (NTP) server that CUCM defines within the Date/Time Group configuration. This ensures consistency and accuracy in time display across all phones, which is critical for environments requiring precise timekeeping (e.g., financial institutions, healthcare).

The Date/Time Group in CUCM allows administrators to specify:

• Time Zone (e.g., America/New_York)

• Time Display Format (12-hour or 24-hour)

• Phone NTP References — a list of NTP servers that phones can use for clock synchronization

To ensure that SIP phones get the correct time:

1. Navigate to System > Date/Time Group in CUCM.

2. Either create a new group or edit an existing one.

3. Specify a valid and reachable Phone NTP Reference (e.g., internal NTP server or public NTP like time.google.com).

4. Assign this Date/Time Group to the Device Pool used by the SIP phones.

5. Restart the phones if necessary.

Once configured, the phones will obtain the IP address or hostname of the NTP server during their registration process and sync their internal clocks accordingly.

Let’s evaluate the incorrect options:

• B (Region): This defines codec preferences and bandwidth limits for calls between different regions but has no impact on time settings.

• C (Time Format): This is a purely cosmetic change that affects how time is displayed, not how it is synchronized or sourced.

• D (Time Zone): While the time zone determines the offset from UTC and is important for correct local time, it doesn’t configure where the time comes from.

To summarize, the only way to correctly direct Cisco SIP phones to use a specific NTP server for time synchronization is by adding a Phone NTP Reference to the Date/Time Group associated with the device’s Device Pool. This ensures proper alignment with the desired time source across the phone network.