Free 350-801 Practice Test Questions 2026

Excess traffic is dropped.

Explanation

Your answer, C, is correct.

Traffic policing is a Quality of Service (QoS) mechanism that enforces a configured traffic-rate limit. When traffic exceeds that limit, the policer typically drops excess packets or, depending on its configuration, marks them with a lower-priority value. Unlike traffic shaping, policing does not buffer excess packets for later transmission.

Correct Option

C. Excess traffic is dropped.
When traffic exceeds the configured rate, the policer takes the configured exceed or violate action. In a basic policing configuration, that action is to drop the excess packets immediately, helping enforce the bandwidth limit.

Incorrect Options

- A. All traffic is treated equally —
Incorrect. Policing distinguishes traffic based on its measured rate and configured actions; conforming and excess traffic can receive different treatment.

- B. Excess traffic is retransmitted —
Incorrect. Policing does not retransmit packets. Dropped packets may be retransmitted by higher-layer protocols, such as TCP, but retransmission is not the policer's function.

- D. Excess traffic is queued —
Incorrect. Buffering excess packets and transmitting them later is the characteristic behavior of traffic shaping, not policing.

Reference
- Cisco, QoS Packet Policing — Cisco IOS XE Configuration Guide

Multimedia conferencing

Explanation

Your answer, C, is correct.

Cisco’s 8-class QoS strategy builds on the four-class model by adding four classes: multimedia conferencing, multimedia streaming, network control, and scavenger. Multimedia conferencing supports interactive collaboration applications, including desktop video and multimedia conferencing.

Correct Option

C. Multimedia conferencing
Multimedia conferencing is one of the explicitly defined additional classes in Cisco’s 8-class QoS strategy. It is designed for interactive video and collaboration traffic, which is sensitive to latency, jitter, and packet loss. Cisco commonly recommends AF41 (DSCP 34) for this class, although some designs may use EF depending on the traffic and policy.

Incorrect Options

- A. Assured Forwarding —
Incorrect. Assured Forwarding (AF) is a DiffServ per-hop behavior and a family of DSCP markings, not the name of an application class in the 8-class QoS strategy.

- B. Broadcast video —
Incorrect. Broadcast video is a valid class in Cisco’s broader 12-class QoS model, but it is not one of the four additional classes introduced by the 8-class model.

- D. Real-Time Interactive —
Incorrect. Real-Time Interactive is also a class in the broader 12-class model, not one of the four additional classes in the 8-class strategy.

Reference
1. Cisco Press, QoS Strategy Models — 8-Class QoS Strategy

1

Explanation

Your answer, B, is correct.

Cisco Unified Communications Manager (Unified CM) allows only one DNS SRV entry to be configured as the destination for a SIP trunk when the Destination Address is an SRV option is selected. The DNS SRV record itself can resolve to multiple server destinations, allowing Unified CM to use DNS-based prioritization and weighting for redundancy and load distribution.

Correct Option

B. 1
In the SIP Trunk Configuration window, you can configure either multiple individual destination addresses or a single DNS SRV destination entry. Although up to 16 destination addresses can be configured for a SIP trunk, that limit applies to individual IP addresses or fully qualified domain names—not multiple SRV entries.

Incorrect Options

- A. 4 —
Incorrect. Unified CM does not support four separate DNS SRV entries in the SIP trunk destination field.

- C. 8 —
Incorrect. Eight is not the supported limit for SRV entries.

- D. 16 —
Incorrect. Sixteen is the maximum number of individual destination addresses supported for a SIP trunk, not the number of DNS SRV entries.

Reference
1. Cisco, System Configuration Guide for Unified CM 12.5 — Configure SIP Trunks

10.0.0.2

Explanation
Your answer, C (10.0.0.2), is the expected answer, assuming the exhibit shows that the route pattern matching directory number 5507 points to the SIP trunk whose destination address is 10.0.0.2.

When a user at extension 4401 dials 5507, Unified CM performs digit analysis, matches the configured route pattern, and sends the call to the associated gateway or SIP trunk. The calling extension does not by itself determine the destination IP address; the configured call-routing path does.

Correct Option

C. 10.0.0.2
The destination is the IP address configured for the trunk or route selected by the matching route pattern. I couldn't directly inspect the exhibit image here, so the answer is conditional on the configuration showing 10.0.0.2 as that destination.

Incorrect Options

- A. 10.0.0.3 —
Incorrect if this address belongs to a different trunk or gateway.

- B. 10.0.0.1 —
Incorrect if this is a Unified CM node or another unrelated destination.

- D. 10.0.0.4 —
Incorrect if this address is not associated with the route selected for 5507.

Reference
- Cisco, System Configuration Guide for Unified CM — Configure Call Routing

Service parameter configuration

Explanation

The correct answer is C. Service parameter configuration.

Cisco Unified Communications Manager (Unified CM) allows administrators to configure cluster-wide default bit-rate values through service parameters. These defaults apply to calls involving regions unless specific region relationships override them. This provides a centralized way to control bandwidth usage for audio and video calls across the deployment.

Correct Option

C. Service parameter configuration

To configure the default values:

1. Open Cisco Unified CM Administration.

2. Navigate to System → Service Parameters.

3. Select the Unified CM server.

4. Select the Cisco CallManager service.

5. Find the relevant parameters under Clusterwide Parameters (System – Location and Region).

These parameters define the default maximum bit rates used by region relationships when Use System Default is selected.

Incorrect Options

- A. Region configuration —
Incorrect for the default. This is where administrators configure bit-rate limits for specific regions and relationships. A region can use the system default or have a custom value.

- B. Enterprise Phone configuration —
Incorrect. Enterprise phone settings control phone-wide configuration defaults, not region-based media bit-rate limits.

- D. Enterprise parameters configuration —
Incorrect. Enterprise parameters control system-wide Unified CM behavior, but region bit-rate defaults are configured under Service Parameters.

Reference
- Cisco, System Configuration Guide for Unified CM 15 — Configure Clusterwide Defaults for Regions

Calling party Transformation Mask

Explanation

Your answer, A, is correct.

In Cisco Unified Communications Manager (Unified CM), the Calling Party Transformation Mask changes the calling number presented to the called party on an outbound call. For example, it can transform an internal extension such as 4401 into the organization's public number, such as +1 212 555 0100, when routing a call to the PSTN.

Correct Option

A. Calling Party Transformation Mask
This field is configured in the route pattern settings and manipulates the caller ID number before Unified CM sends the call onward. The actual caller ID can also be affected by settings on a route group or SIP trunk, which may override route-pattern transformations.

Incorrect Options

- B. Route partition —
Incorrect. A partition organizes dial-plan patterns and controls their reachability through calling search spaces. It does not change caller ID.

- C. Called Party Transformation Mask —
Incorrect. This changes the destination number being dialed, not the calling party's caller ID.

- D. Connected Line ID Presentation —
Incorrect. This controls whether the connected party's number is displayed to the calling party; it does not transform the outbound caller ID.

Reference
- Cisco, Cisco Unified Communications Manager System Guide — Calling and Called Party Transformations

restart of the network service

Explanation
Your selected answer, C, is the expected exam answer, but there is an important technical distinction. In Cisco Unified Communications Manager (Unified CM), changing the primary DNS server through the CLI using set network dns primary requires a server reboot for the change to take effect. Cisco's official CLI documentation specifies using utils system restart. It does not simply require restarting the network service.

Correct Option

C. Restart of the network service —
Expected exam answer, but technically imprecise.

The exam likely intends a network-related restart. However, according to Cisco's official documentation, the actual requirement is to reboot the Unified CM node after changing the DNS server IP address.

Incorrect Options

- A. Restart of CallManager service —
Incorrect. Restarting Cisco CallManager alone does not apply the system-level DNS configuration change.

- B. Restart of DirSync service —
Incorrect. DirSync synchronizes directory information and is unrelated to applying the DNS server configuration.

- D. Restart of TFTP service —
Incorrect. TFTP serves phone configuration files; restarting it does not apply the node's DNS settings.

Reference
Cisco, Command Line Interface Reference Guide for Unified CM 15 — Set Commands. The set network dns section explicitly states that changing the DNS server IP address requires a reboot using utils system restart.

Cisco Dual Mode for Android

Explanation
Your answer, B, is correct.

A BOT device represents a Cisco Jabber softphone on an Android device. When creating the device in Cisco Unified Communications Manager (Unified CM), the administrator must select Cisco Dual Mode for Android as the phone type. Cisco uses the prefix BOT in the device name to identify this type of device.

Correct Option

B. Cisco Dual Mode for Android
To configure the device, navigate to Cisco Unified CM Administration → Device → Phone → Add New, select Cisco Dual Mode for Android, and configure the device name, device pool, owner user ID, and directory number as required.

Incorrect Options

- A. Cisco Dual Mode for iPhone —
Incorrect. This phone type creates a TCT device for Cisco Jabber on an iPhone.

- C. Third-party SIP device —
Incorrect. This is used for supported third-party SIP endpoints, not the native Jabber Android softphone device type.

- D. Cisco Unified Client Services Framework —
Incorrect. This creates a CSF device for Cisco Jabber on Windows or macOS.

Reference
Cisco, Jabber for Android — Quick Start Guide. The guide explicitly identifies Cisco Dual Mode for Android as the phone type used for a BOT softphone.

The ISDN Type is modified to the administrators defined value

Explanation
Your answer, A, is correct.

Cisco IOS voice translation rules can modify not only the numeric phone number but also the ISDN numbering type and numbering plan. When a translation rule specifies type values, the gateway can replace the incoming number type with an administrator-defined value. The question’s exhibit shows two rules intended to manipulate these ISDN attributes rather than strip digits from the telephone number.

Correct Option

A. The ISDN Type is modified to the administrator's defined value.
The type keyword in a voice translation rule specifies the matching and replacement numbering types, such as unknown, national, or international. For example:

rule 1 // // type unknown national

This matches a number with type unknown and changes its type to national, without changing the numeric digits.

Incorrect Options

- B. The ISDN Plan is modified to the administrator's defined value —
Incorrect. The plan keyword changes the numbering plan, such as unknown to isdn. That is a separate attribute from the numbering type.

- C. The leading Plus is stripped from the numeric phone number —
Incorrect. The rules shown concern ISDN numbering metadata, not removal of a + character from the phone number.

- D. Any Zero is stripped from the numeric phone number —
Incorrect. Removing zero digits requires a matching and replacement pattern that explicitly targets those digits. The rules described modify the ISDN type rather than the number itself.

Reference
- Cisco, Configure Number Translation with Voice Translation Profiles — explains the type and plan match and replacement parameters

dtmf-relay rtp-nte sip-notify

Explanation
The correct answer is C. dtmf-relay rtp-nte sip-notify.

DTMF (Dual-Tone Multi-Frequency) relay allows telephone keypad tones to be transmitted reliably during VoIP calls. rtp-nte sends DTMF digits out-of-band using RTP Named Telephone Events, while sip-notify uses SIP NOTIFY messages. However, neither is in-band DTMF. In-band DTMF is carried within the audio stream and is the default when no DTMF relay mechanism is configured. Cisco documents that SIP NOTIFY has higher priority than RTP-NTE when both are successfully negotiated.

Correct Option

C. dtmf-relay rtp-nte sip-notify
This is the intended exam answer for enabling two DTMF relay mechanisms, with RTP-NTE listed before SIP NOTIFY. However, the question’s wording is technically misleading: both configured mechanisms are out-of-band, and Cisco’s SIP negotiation gives SIP NOTIFY precedence when both methods are negotiated successfully. So this command does not make in-band DTMF first priority.

Incorrect Options

- A. dtmf-relay cisco-rtp —
Enables Cisco-proprietary RTP DTMF relay, not ordinary in-band audio DTMF. It also does not configure a second relay method.

- B. sip-notify dtmf-relay rtp-nte —
Invalid command syntax as written; the IOS command begins with dtmf-relay.

- D. dtmf-relay sip-kmpl cisco-rtp —
Configures out-of-band SIP KPML and Cisco RTP relay, rather than prioritizing in-band DTMF.

Reference
- Cisco IOS Voice Command Reference — dtmf-relay

150

Explanation
The correct answer is C. DHCP Option 150.

Cisco IP phones use DHCP Option 150 to discover the IP addresses of their TFTP servers, which provide configuration files and firmware. Option 150 supports two TFTP server addresses, allowing an endpoint to try the secondary server if it cannot communicate with the primary one. This provides TFTP redundancy during startup and registration.

Correct Option

C. 150
Configure Option 150 with both TFTP server IP addresses. For example, on a Cisco IOS DHCP pool:

ip dhcp pool VOICE

option 150 ip 10.10.10.10 10.10.10.11

The phone tries the first address and uses the second if it cannot establish communication with the primary TFTP server.

Incorrect Options

- A. 66 —
Provides a single TFTP server address or hostname, not the required list of two server addresses.

- B. 143 —
Not the standard Cisco IP phone DHCP option for providing redundant TFTP server addresses.

- D. 166 —
Not the standard DHCP option used by Cisco IP phones to obtain two TFTP server addresses.

Reference
- Cisco Unified Communications SRND — DHCP Option 150

Enable Auto-Registration.

Explanation
The correct answer is A. Enable Auto-Registration.

Cisco Unified Communications Manager (CUCM) self-provisioning allows end users to provision their own phones with minimal administrator intervention. Auto-Registration must be enabled and configured with the appropriate directory number range and universal device and line templates so that a new phone can register and receive its initial configuration. Cisco’s configuration guide explicitly includes enabling Auto-Registration as a required step in the self-provisioning workflow.

Correct Option

A. Enable Auto-Registration.
The administrator configures Auto-Registration on the CUCM publisher under System > Cisco Unified CM, selects the Universal Device Template and Universal Line Template, and specifies the starting and ending directory numbers. The phone can then auto-register before the end user completes the self-provisioning process. Self-provisioning also requires the appropriate user profile and permission to provision phones.

Incorrect Options

- B. Associate the end user to the Standard CCM Super Users group.
This group grants elevated administrative privileges and is not required for ordinary end-user self-provisioning. Users should have the appropriate standard end-user access.

- C. Associate the end user to a SIP Profile.
SIP profiles are associated with phone devices and define SIP behavior; this is not the required end-user task described in the question.

- D. Disable Auto-Registration.
This prevents new phones from auto-registering, which conflicts with the standard self-provisioning workflow.

Reference
- Cisco Unified CM System Configuration Guide — Configure Self-Provisioning