Free 350-801 Practice Test Questions 2026

Cisco Dual mode for Android

Explanation:
A BOT device is the soft-phone registration used by Cisco Jabber on Android. In Unified Communications Manager the corresponding phone type that must be selected when adding the device is “Cisco Dual Mode for Android”. This phone type supplies the correct device capabilities, SIP profile defaults, and dual-mode (Wi-Fi/mobile) behavior required by the Jabber Android client.
Correct Option:

A. Cisco Dual Mode for Android
In CUCM Administration → Device → Phone → Add New, choose “Cisco Dual Mode for Android”. Name the device with the recommended BOT prefix (e.g., BOTUSERNAME), assign the end user, line, and appropriate SIP profile. The Jabber Android client will then register as this BOT device.

Incorrect Option:

B. Cisco Unified Client Services Framework
CSF is the phone type used by Jabber for Windows / desktop clients, not by Android.

C. third-party SIP device
This generic type lacks the dual-mode and Jabber-specific features required for a BOT registration.

D. Cisco Dual Mode for iPhone
This phone type is exclusively for Jabber on iOS (TCT devices); it is not used for Android.

Reference:
Cisco Jabber for Android Configuration / Deployment Guides: “To configure a BOT device, select the phone type Cisco Dual Mode for Android.”

1200 through 1399 only

Explanation:

The route pattern 1[23]XX is interpreted as follows:

The digit 1 is literal.

The character class [23] matches exactly one digit that is either 2 or 3.

Each X matches any single digit 0-9.

Consequently the pattern expands to the complete ranges 1200-1299 and 1300-1399.

Correct Option:

A. 1200 through 1399 only
All numbers from 1200 to 1299 (when the second digit is 2) and from 1300 to 1399 (when the second digit is 3) match the pattern; no other four-digit numbers do.

Incorrect Option:

B. 1230 through 1239 only
Far too narrow; the pattern does not require the third digit to be 3.

C. 12300 through 12399 only
The pattern is only four digits long, not five.

D. 1200 through 1300 only
The range stops at 1300, but the pattern continues through 1399.

Reference:
Cisco Unified Communications Manager Dial Plan Guide – Route Pattern wildcards: [23] matches a single digit in the set {2,3} and X matches any digit, producing the ranges 1200-1299 and 1300-1399.

Set the appropriate service parameter

Explanation:
Toll-fraud prevention in Cisco Unified Communications Manager includes the ability to block a call that arrives from the PSTN (OffNet) from being transferred to another PSTN number (OffNet). This behavior is controlled by a dedicated cluster-wide service parameter. Setting the parameter to True (the default in most releases) immediately rejects any OffNet-to-OffNet transfer attempt.

Correct Option:

C. Set the appropriate service parameter
In Cisco Unified CM Administration → System → Service Parameters → Cisco CallManager, locate the parameter “Block OffNet to OffNet Transfer” (or the equivalent “Transfer On-Net to Off-Net” / “OffNet to OffNet Transfer” setting) and set it to True. CUCM then enforces the restriction for all devices in the cluster.

Incorrect Option:

A. Implement time-of-day routing
Time-of-day routing limits when calls can be placed; it does not specifically block OffNet-to-OffNet transfers.

B. Use the correct route filters.
Route filters manipulate which dialed digits are matched by a route pattern; they do not control transfer permissions.

D. Enforce ad-hoc conference restrictions
Ad-hoc conference restrictions (Drop Ad Hoc Conference) prevent conferences from remaining active after the initiator leaves; they do not address simple call transfers between two OffNet parties.

Reference:
Cisco Unified Communications Manager System Guide – Service Parameters and Security Guide: the “Block OffNet to OffNet Transfer” service parameter is the mechanism used to restrict OffNet-to-OffNet call transfers and thereby mitigate toll fraud.

Set up is not required.

Explanation:
When Cisco Unity Connection is already configured to listen for both IPv4 and IPv6 traffic on its SIP ports, the dual-stack addressing is handled automatically by the existing port-group and SIP-trunk settings. No additional addressing-mode configuration is required on the Unity Connection side for the SIP integration with Unified Communications Manager to function correctly for both IP versions.

Correct Option:

D. Set up is not required.
Because Unity Connection is listening on both IPv4 and IPv6, the SIP stack accepts and responds to calls on either protocol without further addressing-mode changes. The integration works with the current dual-stack configuration.

Incorrect Option:

A. Set up media ports for each port group to use IPv4.
Forcing media to IPv4-only would disable the IPv6 capability that has already been enabled.

B. Set up for each group to use IPv4 and IPv6.
This step is unnecessary once the system is already listening for both protocols.

C. Set up IPv4 and IPv6 in cisco Unified CM.
While CUCM must also support dual-stack, the question asks specifically about the addressing-mode setting inside Unity Connection.

Reference:
Cisco Unity Connection Administration Guide / SIP Integration Guide: when Unity Connection is configured to accept both IPv4 and IPv6 traffic, no additional addressing-mode configuration is required for the SIP integration with Unified CM.

E.164

Explanation:
A global enterprise with offices on four continents needs a consistent, country-independent numbering scheme that can also interoperate with Microsoft Lync (Skype for Business). The E.164 international numbering plan provides a unique, standardized format (+) that is understood by every PSTN carrier worldwide and is the native addressing method used by Lync/Skype for Business for SIP URI and phone-number normalization. Using E.164 therefore satisfies both the multi-country dial-plan requirement and the Lync integration requirement with a single design.

Correct Option:

D. E.164
Design the dial plan around full E.164 numbers (with the “+” prefix). CUCM, gateways and the Lync backend all normalize and route on E.164, eliminating translation complexity between countries and ensuring seamless federation with Lync.

Incorrect Option:

A. SIP URI
SIP URIs are useful for pure IP endpoints but do not by themselves provide a globally routable numeric dial plan for PSTN connectivity across four continents.

B. TEHO
Tail-End Hop-Off is a call-routing strategy, not a dial-plan numbering type.

C. H.323
H.323 is a signaling protocol, not a dial-plan format.

Reference:
Cisco Collaboration SRND – Dial Plan: for multi-national deployments and Microsoft Lync/Skype for Business integration, the recommended dial-plan type is E.164.

Database replication is not synchronized on the Unified CM nodes.

Explanation:
A stratum value of 16 returned by utils ntp status indicates that the Unified CM publisher is not synchronized to any valid NTP source. CUCM database replication relies on tightly synchronized clocks across all cluster nodes. When the publisher’s clock is unsynchronized (stratum 16), time drift quickly exceeds the allowable threshold and database replication between the publisher and subscribers fails or becomes inconsistent.

Correct Option:

D. Database replication is not synchronized on the Unified CM nodes.
Because the publisher cannot obtain accurate time, the timestamps used by the Informix database replication process become unreliable, resulting in replication status errors (for example, “Replication status = 2” or “Not Synchronized”).

Incorrect Option:

A. The date/time group on all phones default to the time zone of the engineer.
Phone time-zone behavior is controlled by Date/Time Groups, not by the NTP stratum of the CUCM servers.

B. The cluster loses access to port 124 at the firewall.
NTP uses UDP port 123, not 124.

C. Unified CM sends an NTPv4 packet.
Sending NTPv4 packets is normal and does not cause a stratum-16 condition.

Reference:
Cisco Unified Communications Manager NTP Troubleshooting Guide: stratum 16 means the server is unsynchronized; time differences greater than a few minutes break database replication among cluster nodes.

The MWI status is retained after a mailbox is moved from one store to another.

Explanation:
When mailboxes are moved between mailbox stores in Cisco Unity Connection, the system preserves the current Message Waiting Indicator (MWI) state of each mailbox. Active MWIs remain active, and inactive MWIs remain inactive after the move completes. No manual reset or re-synchronization of MWI is required solely because of the store move.

Correct Option:

D. The MWI status is retained after a mailbox is moved from one store to another.
Unity Connection explicitly retains the MWI on/off status for every mailbox that is relocated, so users continue to see the correct lamp indication on their phones without further intervention.

Incorrect Option:

A. The move will fail if MWI status is active.
Active MWI does not block a mailbox move.

B. Moving the mailboxes from one store to another fails if MWI is turned on.
Same as A – the move succeeds regardless of MWI state.

C. If source and target mailbox store are not disabled, MWI status is not retained.
The documentation states that MWI status is retained; the stores must simply be enabled for the move to be allowed.

Reference:
Cisco Unity Connection System Administration Guide – Message Storage: “The MWI status is retained after a mailbox is moved from one mailbox store to another.”

Configure codec transparent on the dial peers.

Explanation:
When video endpoints must be allowed to negotiate media capabilities freely, the Cisco Unified Border Element (CUBE) must not insert itself into the codec selection process. Configuring codec transparent on the inbound and outbound dial-peers instructs CUBE to pass the SDP offer/answer through unchanged, so the two endpoints can agree on any mutually supported video (and audio) codec without CUBE forcing or filtering the list.

Correct Option:

D. Configure codec transparent on the dial peers.
Under both the inbound and outbound VoIP dial-peers enter:

codec transparent

CUBE then leaves the SDP codec attributes intact, enabling end-to-end negotiation by the video endpoints.

Incorrect Option:

A. Configure path-thru content sdp on the voice service.
This command passes certain SDP body content but is not the primary method for transparent codec negotiation.

B. Configure a hardcoded codec on the dial peers.
A hardcoded codec forces a single payload type and prevents free negotiation.

C. Configure a transcoder for video protocols.
A transcoder is used when codecs must be converted; it interferes with, rather than permits, native negotiation.

Reference:
Cisco Unified Border Element Configuration Guide – Codec Transparent: “codec transparent” leaves the codec information contained within the call signaling untouched so that endpoints can negotiate freely.

Voice packets are classified and marked.

The network meets bandwidth requirements.

Explanation:
Jitter is the variation in inter-packet arrival time. Two fundamental network conditions that create or exacerbate jitter are (1) lack of proper QoS classification and marking, which allows voice packets to be queued behind data bursts, and (2) insufficient bandwidth, which causes congestion and variable queuing delay. Ensuring that voice packets are classified and marked (normally EF) and that the network path meets the required bandwidth eliminates the most common sources of jitter.

Correct Option:

A. Voice packets are classified and marked.
Classify RTP/SRTP streams and mark them with DSCP EF (or the appropriate media marking). Edge switches and routers can then place the packets in a priority queue, keeping delay variation low.

C. The network meets bandwidth requirements.
Provision adequate bandwidth (including Layer-2/Layer-3 overhead) on every link the voice path traverses so that queues do not build and introduce variable delay.

Incorrect Option:

B. All devices use wired connections instead of wireless connections.
While wireless can add jitter, it is not a primary area an engineer is required to “check to prevent jitter” in a general voice-quality design; QoS and bandwidth are more universal.

D. Cisco UBE manages voice traffic, not data traffic.
CUBE is a border element; it does not by itself control jitter on the IP network.

E. MTP is enabled on the SIP trunk to Cisco Unified Border Element.
An MTP may be required for certain supplementary services or codec changes, but it does not reduce network jitter.

Reference:
Cisco Collaboration QoS Design Guide / SRND: the two primary methods to control jitter are proper classification/marking of voice packets and ensuring the network meets bandwidth requirements so that priority queuing can keep delay variation within acceptable limits.

Clock source network

Explanation:
On a Cisco IOS XE voice gateway the T1/E1 controller must be told whether it should recover clock from the line or supply clock to the line. When the gateway itself is to act as the clock source (i.e., provide clocking to the attached TDM device or network), the correct controller command is clock source network.

Correct Option:

A. Clock source network
Under the T1/E1 controller configuration mode enter:

clock source network

This configures the controller to generate and supply clocking toward the line.

Incorrect Option:

B. Clock source line
This setting makes the controller recover clock from the received signal (clock slave), the opposite of providing clock.

C. Clock source internal
On IOS XE voice T1/E1 controllers the keyword for supplying clock is network, not internal (the latter is used mainly for data-only or older platforms).

D. Cisco IOS XE TDM gateway T1/E1 controller cannot provide clocking
The controller can provide clocking; the proper command simply has to be used.

Reference:
Cisco IOS XE T1/E1 NIM Configuration Guide and Voice Gateway Clocking documentation: use clock source network on the voice T1/E1 controller when the gateway must supply clocking to the line.

The RTP port is set to 0.

Explanation:
In SIP SDP negotiation the media line (m=) contains the port number on which the endpoint is prepared to receive RTP. When the called device answers with a port value of 0 it is explicitly rejecting the media stream. The calling device therefore cannot establish an RTP path and the call fails to set up media, even though the SIP signaling (INVITE/200 OK) may succeed.

Correct Option:

A. The RTP port is set to 0.
The SDP answer contains an m=audio 0 RTP/AVP \ldots line (or the port field is 0). Per RFC 3264 a port of 0 means the stream is declined; no RTP is exchanged and the call fails to establish media.

Incorrect Option:

B. The payload for G.711ulaw must be 18.
G.711 µ-law uses static payload type 0, not 18 (18 is G.729). This is not the cause of the failure shown.

C. The media stream is set to sendonly.
A sendonly attribute would still allow media in one direction; it does not by itself prevent call establishment.

D. The calling device did not offer a ptime value.
The ptime attribute is optional; its absence does not cause a call to fail.

Reference:
RFC 3264 (SDP Offer/Answer) and Cisco SIP Troubleshooting Guides: an RTP port of 0 in the SDP answer indicates media rejection and is a common reason for call-setup failure between SIP devices.

AF13

Explanation:
Assured Forwarding (AF) PHBs use the second digit of the AF name to indicate drop precedence:

x1 = low drop probability

x2 = medium drop probability

x3 = high drop probability

Therefore, within any AF class, the AF*3 marking has the highest likelihood of being discarded when congestion occurs. Among the listed options AF13 carries the highest drop precedence.

Correct Option:

D. AF13
AF13 belongs to Assured Forwarding class 1 with high drop precedence. In a congested queue that implements WRED or similar, packets marked AF13 are dropped before AF11 or AF12 packets of the same class.

Incorrect Option:

A. AF32
Medium drop precedence in class 3 – lower drop probability than AF13.

B. AF12
Medium drop precedence in class 1 – lower drop probability than AF13.

C. AF11
Low drop precedence in class 1 – the least likely of the AF1x markings to be dropped.

Reference:
RFC 2597 (Assured Forwarding PHB Group) and Cisco QoS Design Guides: the third bit of the AF DSCP (the drop-precedence field) determines relative drop likelihood; AF*3 is the most drop-prone.