Cisco 350-801 Practice Test Questions, Exam Dumps

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350-801 Cisco Practice Test Questions and Exam Dumps

Question No 1:

Which two functionalities does Cisco Expressway provide in the Cisco Collaboration architecture? (Choose two.)

A. Survivable Remote Site Telephony functionality
B. MGCP gateway registration
C. Secure business-to-business communications
D. Customer interaction management services
E. Secure firewall traversal for remote devices

Answer:

The correct answers are C. Secure business-to-business communications and E. Secure firewall traversal for remote devices.

Explanation:

Cisco Expressway is a critical component in the Cisco Collaboration architecture, primarily designed to enable secure and seamless communication across enterprise boundaries, allowing for secure connections between remote devices and internal networks. Cisco Expressway provides the following key functionalities:

Secure business-to-business communications (Option C): Cisco Expressway plays a key role in facilitating secure business-to-business communications by enabling secure communication with external parties, including suppliers, partners, and customers. Using Cisco Expressway-E, businesses can securely extend collaboration services to external users without compromising security. This is achieved through technologies like borderless collaboration and WebEx meeting integration, allowing organizations to securely connect with business partners outside the firewall.
Secure firewall traversal for remote devices (Option E): Cisco Expressway is widely used for secure firewall traversal, enabling remote devices such as smartphones, tablets, and laptops to securely connect to the enterprise network without direct access through a firewall. The Cisco Expressway-C and Cisco Expressway-E provide a seamless and secure connection between remote endpoints and the internal collaboration environment by managing firewall traversal for both voice and video calls, as well as instant messaging services. This helps ensure remote workers can securely access collaboration services without compromising security policies.

Why Other Options Are Incorrect:

A. Survivable Remote Site Telephony functionality: While Cisco Expressway is part of a larger collaboration solution, Survivable Remote Site Telephony (SRST) is a feature typically handled by Cisco Unified Communications Manager (CUCM) or Cisco Unified Survivable Remote Site Telephony (SRST). It is not a primary function of Cisco Expressway.
B. MGCP gateway registration: MGCP (Media Gateway Control Protocol) is also related to Cisco Unified Communications Manager for controlling gateways, but it is not a core function of Cisco Expressway.
D. Customer interaction management services: Customer interaction management services, such as those provided by Cisco Contact Center solutions, are separate from the primary role of Cisco Expressway, which focuses more on secure communications and traversal.

Thus, the correct functionalities of Cisco Expressway in the Cisco Collaboration architecture are C and E.

Question No 2:

An engineer is tasked with extending the corporate phone system to mobile users who will connect through the internet using their own devices. One of the key requirements is to make the setup as simple as possible for the end users. Which infrastructure component would best achieve this goal?

A. Cisco Express Mobility
B. Cisco Expressway-C and Expressway-E
C. Cisco Unified Border Element
D. Cisco Unified Instant Messaging and Presence

Answer:

The correct answer is B. Cisco Expressway-C and Expressway-E.

Explanation:

Cisco Expressway-C and Expressway-E are essential components in extending a corporate phone system to remote users, especially when those users are connecting from their mobile devices over the internet. These components are designed to provide secure firewall traversal and seamless connectivity for mobile users, enabling them to access the corporate communication system without requiring complex configurations.

Here’s how Cisco Expressway-C and Expressway-E work:

Expressway-C is typically deployed within the corporate network and integrates with existing collaboration infrastructure, such as Cisco Unified Communications Manager (CUCM) and Cisco Unity Connection. It acts as a gatekeeper for communication traffic within the internal network.
Expressway-E is placed outside the corporate firewall, allowing secure communication between remote devices and the corporate network. Expressway-E provides a secure and straightforward path for mobile users to access the corporate phone system, and it handles all the complexities of firewall traversal, security, and NAT (Network Address Translation).

These two components together simplify the user experience by eliminating the need for users to configure VPNs or manage complex settings. All the user needs is an internet connection, and their mobile devices can securely access the phone system without additional configuration.

Why Other Options Are Incorrect:

A. Cisco Express Mobility: Cisco Express Mobility is typically used in wireless LAN environments to provide seamless roaming between access points. It does not specifically address the goal of extending the phone system to mobile users over the internet.
C. Cisco Unified Border Element: While Cisco Unified Border Element (CUBE) is useful for connecting internal and external SIP (Session Initiation Protocol) systems, it does not simplify the connection for mobile users in the way that Expressway-C and Expressway-E do. CUBE is more focused on interconnecting VoIP systems rather than providing secure remote access.
D. Cisco Unified Instant Messaging and Presence: This tool facilitates real-time messaging and presence information, but it is not directly responsible for connecting mobile users to a corporate phone system. It does not address the specific need for seamless mobile device integration over the internet.

In summary, Cisco Expressway-C and Expressway-E provide the necessary infrastructure to extend a corporate phone system to mobile users in a simple and secure manner, making them the best choice for this requirement.

Question No 3:

A corporate client is planning to set up a high-end immersive video conference involving five Cisco TelePresence IX5000 Series systems located at different geographical offices. To ensure the full immersive video and audio experience during the conference, which media resource must be incorporated into the design to properly manage the multi-screen, multi-stream capabilities and deliver optimal quality and functionality?

A. Cisco PVDM4-128 (Packet Voice Digital Signal Processor Module)
B. Software-based conference bridge integrated with Cisco Unified Communications Manager (CUCM)
C. Cisco Webex Meetings Server (on-premises Webex solution)
D. Cisco Meeting Server (CMS) (integrated multiparty conferencing platform)

Correct Answer:

D. Cisco Meeting Server

Explanation:

When designing a conference architecture involving Cisco TelePresence IX5000 Series systems, especially with multiple immersive endpoints like in this five-location scenario, it is critical to use a media resource that supports immersive telepresence capabilities. The IX5000 Series endpoints are designed to deliver a highly immersive experience using multiple high-definition video streams, spatial audio, and precise video switching based on participant activity.

Cisco Meeting Server (CMS) is the required media resource in this context. CMS is a powerful, scalable conferencing solution that supports multi-screen immersive video conferencing, dynamic screen layouts, and continuous presence — all essential for fully leveraging the immersive capabilities of the IX5000 Series. CMS can intelligently manage the multiple video streams from each IX5000 system and compose a synchronized, life-like meeting experience across all sites.

Other options do not meet these advanced requirements:

Cisco PVDM4-128 is a DSP module used for traditional audio transcoding and basic video support, not suitable for full immersive telepresence.
Software conference bridges on CUCM can support basic video conferencing but lack the capacity for multi-screen immersive conferencing.
Cisco Webex Meetings Server is designed for general web-based meetings and lacks the specialized integration and media handling necessary for immersive TelePresence rooms.

Thus, to ensure the full utilization of the IX5000 systems' immersive functionality during the conference, Cisco Meeting Server must be deployed as the primary media resource.

Question No 4:

An engineer is tasked with designing a load balancing solution for a SIP-based voice network that uses two Cisco Unified Border Element (CUBE) routers to handle inbound and outbound SIP traffic.
The network requirements specify that:

The first CUBE router (cube1.abc.com) must handle 60% of all SIP call traffic.
The second CUBE router (cube2.abc.com) must handle 40% of all SIP call traffic.

The organization has already created the necessary DNS A records for both CUBE routers.
To complete the design, the engineer needs to configure appropriate DNS SRV (Service) records to distribute the load accordingly based on these traffic percentages.

Which two SRV record configurations are needed to correctly achieve the specified load balancing behavior?(Choose two.)

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. _sip._udp.abc.com 60 IN SRV 1 60 5060 cube1.abc.com
C. _sip._udp.abc.com 60 IN SRV 1 40 5060 cube2.abc.com

Explanation:

In a DNS SRV record, the priority and weight fields are critical for directing and balancing SIP traffic:

Priority determines the order in which servers are contacted (lower value = higher priority).
Weight is used to distribute the load among servers with the same priority.

Since the goal is to have both routers at the same priority (meaning clients can choose between them for load balancing), the priority for both CUBEs should be identical — typically set to 1.
This ensures the load balancing depends only on the weight.

The weight field specifies how traffic is divided between servers:

cube1.abc.com must handle 60% of the traffic, so its SRV record should have a weight of 60.
cube2.abc.com must handle 40% of the traffic, so its SRV record should have a weight of 40.

Thus, the correct SRV entries would be:

E: _sip._udp.abc.com 60 IN SRV 1 60 5060 cube1.abc.com → Priority 1, Weight 60 for cube1
C: _sip._udp.abc.com 60 IN SRV 1 40 5060 cube2.abc.com → Priority 1, Weight 40 for cube2

This setup ensures that SIP clients will randomly select between the two routers based on the relative weights, achieving approximately 60/40 traffic distribution as designed.

Incorrect answers (like A, B, D) either have wrong priority values or reverse the correct priority-weight structure, disrupting the intended load balancing behavior.

Question No 5:

An engineer is deploying the Cisco Expressway Series in a collaboration architecture to enable secure and seamless communication between internal and external endpoints, as well as to support interoperability between different signaling protocols.

Which two key functions are natively provided by the Cisco Expressway Series to enhance collaboration services and interoperability?(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:

A. Seamless interworking between SIP and H.323 signaling protocols
B. Centralized endpoint registration for internal and external devices

Explanation:

The Cisco Expressway Series (consisting of Expressway-C for internal connections and Expressway-E for external connections) is a powerful platform primarily designed to extend collaboration services beyond the enterprise firewall without requiring a VPN. It supports multiple critical functions:

One of its key roles is interworking between SIP and H.323 protocols (Option A). Many enterprises use a mix of SIP- and H.323-based systems, and Expressway allows them to communicate transparently by translating signaling between these two protocols.

Additionally, the Cisco Expressway supports endpoint registration via Mobile and Remote Access (MRA). With MRA, Cisco Jabber clients, Webex clients, and some Cisco video endpoints can register to Cisco Unified Communications Manager (CUCM) even when they are located outside the corporate network (Option B). This secure registration uses HTTPS without requiring traditional VPN tunnels.

However, Expressway does not provide:

Intercluster Extension Mobility (ICEM) (Option C), which is handled natively by CUCM.
Native transcoding (Option D), which is typically performed by Media Termination Points (MTPs) or DSP resources, not Expressway.
Voice and video conferencing (Option E) at the media mixing level — that is the role of Cisco Meeting Server (CMS) or Cisco TelePresence Servers.

Therefore, the Cisco Expressway focuses on secure collaboration extension, protocol interworking, and mobile remote access, rather than media processing like transcoding or conferencing.

Question No 6:

An engineer is troubleshooting an issue where an incoming off-net call to a corporate user is consistently failing.
Upon investigation, it is determined that the incoming call from the PSTN uses the G.711 codec, while the user's IP phone only supports the G.729 codec.
The call cannot be completed because the codecs are incompatible.

To resolve this issue and allow seamless codec negotiation between the PSTN and the IP phone, which media resource must the engineer configure on both the Cisco Unified Border Element (CUBE) and the Cisco Unified Communications Manager (CUCM)?

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

Correct Answer:

A. Transcoder

Explanation:

In Cisco voice networks, when two endpoints (such as a PSTN gateway and an IP phone) use different audio codecs and are unable to directly negotiate a common codec, a media resource must step in to convert one codec to another so the call can be established successfully.

In this scenario:

The PSTN call arrives using G.711.
The IP phone only supports G.729.
There is a mismatch, and without help, the call fails.

The correct solution is to configure a transcoder (Option A).
A transcoder is a media resource that dynamically converts media streams between different codec types — for example, from G.711 to G.729 — allowing successful communication between devices with incompatible codecs.

The other options are incorrect because:

CFB (Conference Bridge) (Option B) is used to mix multiple audio streams for conferencing, not for codec translation.
MOH (Music on Hold) (Option C) plays music to users on hold but does not perform codec negotiation or conversion.
MTP (Media Termination Point) (Option D) helps in DTMF (dual-tone multi-frequency) relay and early offer scenarios where media channels must be established quickly, but it does not perform codec transcoding.

Thus, to manage codec differences and allow the call to complete between G.711 and G.729 devices, a transcoder must be properly configured and available to the call flow within both the CUBE and CUCM environments.

Question No 7:

An engineer is configuring call behavior in a Cisco Unified Communications Manager (CUCM) deployment.
The requirement is to ensure that when a user who initiated a multiparty (ad hoc) conference call hangs up, the entire conference call is automatically disconnected for all participants.
This prevents conferences from continuing unnecessarily after the initiator leaves the call.

Which service parameter within Cisco Unified Communications Manager should the engineer enable to achieve this behavior?

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. Drop Ad Hoc Conference

Explanation:

In Cisco Unified Communications Manager (CUCM), when users create ad hoc conference calls (calls initiated spontaneously by a user who adds additional participants), administrators sometimes want the conference to end automatically when the initiating user disconnects.
By default, in many CUCM setups, if the initiator of a conference hangs up, the remaining participants can continue speaking — which might not be desirable in all environments, especially where security or resource optimization is critical.

To control this behavior, Cisco provides a service parameter called "Drop Ad Hoc Conference" (Option A).

When the Drop Ad Hoc Conference parameter is enabled, CUCM is instructed to tear down the entire conference if the conference creator (initiator) disconnects.
This ensures that:

No unintended conversations continue without the initiating party.
Resources (like conference bridges) are freed up immediately.
Participants do not remain unknowingly connected to an active call.

The other options do not achieve this behavior:

H.225 Block Setup Destination (Option B) relates to H.323 call signaling and is used to block call setup messages to destinations — it does not affect ad hoc conference behavior.
Block OffNet To OffNet Transfer (Option C) prevents a user from transferring an external call to another external endpoint, enforcing toll-bypass control — unrelated to multiparty conference termination.
Enterprise Feature Access Code for Conference (Option D) refers to configuring feature access codes to initiate or control conference operations, not for automatically dropping conferences when a user disconnects.

Configuration Details:

To enable this feature:

Go to Cisco Unified CM Administration.
Navigate to System > Service Parameters.
Select the appropriate server and the Cisco CallManager service.
Locate the Drop Ad Hoc Conference parameter.
Set it to "True" or enable it based on your call management requirements.

Note: When enabling this, you should also verify your organization's collaboration policy — in some cases, users expect conferences to continue even if the host leaves, especially for scheduled meetings.

In conclusion, to automatically disconnect a multiparty ad hoc call when the initiator hangs up, the engineer must enable the "Drop Ad Hoc Conference" service parameter in CUCM.

Question No 8:

A network administrator has recently deleted a user account from the company's LDAP directory (such as Microsoft Active Directory).
However, upon reviewing the Cisco Unified Communications Manager (CUCM) user database, the administrator notices that the user still appears but is marked as an "Inactive LDAP Synchronized User."

To fully remove this inactive LDAP-synced user from CUCM, what is the next required step that the administrator must perform?

A. Manually delete the user directly from Cisco Unified Communications Manager.
B. Restart the Directory Synchronization (DirSync) service after removing the user from the LDAP directory.
C. Manually trigger a directory synchronization to update the CUCM database and allow the user to be deleted.
D. Wait 24 hours for the automatic garbage collection process to remove the user from the system.

Correct Answer:

C. Manually trigger a directory synchronization to update the CUCM database and allow the user to be deleted.

Explanation:

In a Cisco Unified Communications Manager (CUCM) environment integrated with an external LDAP directory (like Active Directory), users are synchronized from LDAP into CUCM.
This synchronization ensures that user accounts are automatically populated and updated without manual intervention.

When a user is deleted from the LDAP directory, CUCM does not immediately remove the user. Instead, the user status changes to "Inactive LDAP Synchronized User."
This status indicates that the user no longer exists in LDAP, but CUCM has not yet completed the process of fully cleaning up the record.

To remove the user from CUCM after they are marked inactive:

You need to initiate a manual LDAP synchronization (Option C).
Manual synchronization updates CUCM’s internal database, processes inactive records, and removes users who no longer exist in the external LDAP directory.
This synchronization immediately refreshes the user database, reflecting the latest LDAP state and deleting any obsolete user entries.

The other options are incorrect for the following reasons:

A. Deleting the user manually is not permitted for LDAP-synced users directly; CUCM locks LDAP-synced fields to prevent manual editing or deletion.
B. Restarting the DirSync service does not automatically trigger synchronization or user deletion. It only restarts the service process itself.
D. Waiting 24 hours relies on the Garbage Collector process, but this is not reliable for immediate removal and depends on other CUCM internal schedules. Manual sync is faster and more predictable.

How to Execute Manual LDAP Synchronization:

Log in to Cisco Unified CM Administration.
Navigate to System > LDAP > LDAP Directory.
Locate the configured LDAP directory profile.
Click Perform Full Sync Now or Synchronize Now (depending on CUCM version).

After the sync completes, the user should be completely removed from the CUCM database if they no longer exist in LDAP.

In summary, to promptly and cleanly remove the inactive LDAP user from Cisco Unified Communications Manager, the administrator should execute a manual synchronization.

Question No 9:

A customer is using Cisco Unity Connection integrated with an LDAP directory (such as Microsoft Active Directory) to manage voicemail users and authentication.
As a Cisco Unity Connection administrator, you have received a request to change the first name of an existing voicemail (VM) user.

Considering the system is LDAP-integrated and user information is synchronized automatically, where must the first name change be performed so that it correctly updates in Cisco Unity Connection?

A. Directly in Cisco Unity Connection
B. In the Cisco Unified Communications Manager (CUCM) end user profile
C. In the Active Directory (LDAP server)
D. In Cisco IM and Presence Server

Correct Answer:

C. In the Active Directory (LDAP server)

Explanation:

When Cisco Unity Connection is configured to use LDAP integration (typically with Microsoft Active Directory), user accounts and related information such as first name, last name, phone number, and email address are synchronized from LDAP into Unity Connection.
This integration allows centralized user management and ensures consistent, accurate information across the collaboration environment.

Because of this synchronization:

Unity Connection becomes read-only for fields that are controlled by LDAP.
You cannot directly edit LDAP-synced fields like the first name from within Unity Connection.
The system expects all changes to user identity information (such as name updates) to happen at the source, which is the LDAP directory — typically Active Directory.

Therefore, to change the first name of a voicemail user:

You must edit the user’s record in Active Directory.
After updating the information, Unity Connection will automatically update its internal user records during the next scheduled LDAP synchronization.
Alternatively, an administrator can manually trigger a directory synchronization in Unity Connection to immediately pull the changes without waiting for the next scheduled sync.

The other options are incorrect because:

A. Cisco Unity Connection: You cannot manually edit LDAP-synced fields in Unity Connection; fields like first name are grayed out.
B. CUCM end user: Although CUCM may also be LDAP-synced, modifying a user in CUCM does not affect Unity Connection — changes must originate in the LDAP directory.
D. Cisco IM and Presence: This server handles instant messaging and presence information; it is unrelated to voicemail user data.

How to Proceed:

Open Active Directory Users and Computers.
Locate the user account associated with the voicemail user.
Edit the First Name field.
Save the changes.
In Cisco Unity Connection Administration, navigate to LDAP Directory Configuration and either wait for the next sync cycle or click Perform Full Sync Now to manually refresh the user data.

In conclusion, because Cisco Unity Connection relies on LDAP for user identity synchronization, any changes such as modifying a user's first name must be performed directly in Active Directory, ensuring that Unity Connection reflects the update after synchronization.

Question No 10:

An engineer is setting up Cisco SIP IP phones in a Cisco Unified Communications Manager (CUCM) environment and needs to ensure that the phones display the correct time according to a designated time source.
The phones must synchronize their clock settings accurately with a specific time server configured within the system.

Which configuration step is necessary to properly direct the Cisco SIP phones to synchronize their time with the desired source?

A. Add a Phone NTP Reference to the appropriate Date/Time Group assigned to the device.
B. Assign the device to the correct Region configuration in CUCM.
C. Modify the Time Format from 24-hour to 12-hour format.
D. Change the Time Zone setting from America/Los_Angeles to Etc/GMT+8 for the device.

Correct Answer:

A. Add a Phone NTP Reference to the Date/Time Group.

Explanation (around 450 words):

In Cisco Unified Communications Manager (CUCM), SIP phones rely heavily on the CUCM server settings to obtain and synchronize their date and time information.
However, for accurate clock synchronization — especially in environments spanning multiple time zones or requiring compliance with precise timekeeping (such as in financial or healthcare industries) — it's crucial to configure a proper Network Time Protocol (NTP) source.

The correct procedure involves adding a Phone NTP Reference to the Date/Time Group:

A Date/Time Group in CUCM defines the time zone, date/time formats, and most importantly, the NTP reference for a group of phones.
By configuring a Phone NTP Reference within a Date/Time Group, you specify which external NTP server (like a company’s authoritative time server) the SIP phones should synchronize with.
When the phones boot up or re-register, they inherit the NTP server settings from the Date/Time Group they are associated with via their device pool, allowing them to synchronize their internal clocks accurately.

The other options are incorrect because:

B. Assigning the device to a Region: Region configuration in CUCM deals with codec selection and bandwidth settings between sites; it does not affect time synchronization.
C. Changing the Time Format: Adjusting from 24-hour to 12-hour format changes only the visual display of the time on the phone screen — it doesn't configure or synchronize the actual time source.
D. Changing the Time Zone: Setting the time zone (like from America/Los_Angeles to Etc/GMT+8) impacts how the time is displayed relative to UTC, but does not configure or assign the NTP server used for clock synchronization.

Configuration Steps:

In CUCM Administration, navigate to System > Date/Time Group.
Select or create a Date/Time Group.
Add a Phone NTP Reference to the group (an NTP server IP or hostname).
Assign the Date/Time Group to the Device Pool used by the phones.
Save and reset phones if necessary.

Thus, to ensure that a Cisco SIP phone synchronizes its time with a specific source, it is necessary to add a Phone NTP Reference to the Date/Time Group associated with the device.