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350-801 Questions and Answers

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Questions # 1:

What is the function of DiffServ?

Options:

It automatically identifies traffic

It provides reserve bandwidth by using RSVP

It categorizes traffic.

It guarantees end-to-end bandwidth for traffic.

Answer

Questions # 2:

Which value should be changed when each Cisco UCM node does not allow for more than 5000 phones to be registered?

Options:

Maximum Number of Registered and Unregistered Devices service parameter on each node

Minimum Number of Phones service parameter on each node

Maximum Number of Registered Devices service parameter on each node

Maximum Number of Phones service parameter on the Publisher

Answer

Questions # 3:

Refer to the exhibit.

Question # 3

Which codec should an engineer select for a call made between "Dallas-REG' & "Austin-REG'?

Options:

MP4A-LATM

G.711

OPUS

G.729

Answer

Explanation

The codec preference list for the "Dallas-REG" region is "Factory Default low loss". This list includes the following codecs in order of preference:

G.729

G.711

OPUS

MP4A-LATM

The codec preference list for the "Austin-REG" region is "Factory Default low loss". This list includes the following codecs in order of preference:

G.729

G.711

OPUS

MP4A-LATM

Since both regions have the same codec preference list, the codec that will be used for a call made between "Dallas-REG" and "Austin-REG" is G.729.

G.729 is a narrowband speech codec that was developed by the ITU-T in 1988. It is a low-bitrate codec that provides good quality speech at a bitrate of 8 kbps. G.729 is widely used in VoIP applications and is the default codec for many VoIP systems.

G.711 is a wideband speech codec that was developed by the ITU-T in 1972. It is a high-bitrate codec that provides excellent quality speech at a bitrate of 64 kbps. G.711 is not as widely used as G.729 due to its high bitrate requirements.

OPUS is a lossy audio codec that was developed by the IETF in 2012. It is a low-bitrate codec that provides good quality speech at a bitrate of 6 kbps. OPUS is widely used in VoIP applications and is the default codec for many VoIP systems.

MP4A-LATM is a lossy audio codec that was developed by the IETF in 1999. It is a high-bitrate codec that provides excellent quality speech at a bitrate of 24 kbps. MP4A-LATM is not as widely used as G.729 or OPUS due to its high bitrate requirements.

Questions # 4:

An engineer must add a new user account and mailbox in Cisco Unity Connection. Drag and drop the configuration steps from the left into the correct order on the right.

Question # 4

Options:

Answer

Answer:

Questions # 5:

A company has an excessive number of call transfers to local and long-distance PSTN from Cisco Unity Connection voicemail. Which action in the Cisco Unity Connection restriction table resolves this issue?

Options:

Block PSTN patterns on Default Transfer. Default Outdial. and Default System Transfer.

Implement password complexity on voicemail boxes to prevent accounts from being compromised.

Create a custom restriction table ??????????? and block it.

Create a custom restriction table **********and block it.

Answer

Questions # 6:

An engineer configures a SIP trunk for MWI between a Cisco UCM cluster and Cisco Unity Connection. The Cisco UCM cluster fails to receive the SIP notify messages. Which two SIP trunk settings resolve this issue? (Choose two.)

Options:

accept out-of-dialog refer

accept out-of-band notification

transmit security status

allow changing header

accept unsolicited notification

Answer

A, E

Questions # 7:

An administrator is configuring a new Cisco UCM with PSTN capabilities. Due to

bandwidth constraints, audio compression is used on the codec. DTMF must work as

expected because the customer is calling many call centers where the users must select

options in the call. Where is DTMF out-of-band in a CCM 12.5 with SIP-based gateway

configured?

Options:

in the DTMF setting under SIP profile on the Cisco Unified Border Element

in the dial peer on the Cisco IOS router

in regions on the Cisco UCM where the appropriate codec to use is set

in DTMF settings in the audio codec preference list under regions in the Cisco UCM

Answer

Questions # 8:

Which DSCP marking is represented as 101110 in an IP header?

Options:

CS3

AF41

AF31

Answer

Questions # 9:

Which values must be enabled to set up QoS for video conferencing on a VoIP network?

Options:

CS4 and DSCP 32

AP41 and DSCP 34

AF31 and DSCP 34

CS3 and DSCP 24

Answer

Questions # 10:

When multiple potential patterns are present, which two things are considered when Cisco UCM selects a destination pattern? (Choose two.)

Options:

The pattern matches the shortest explicit prefix.

The pattern does not match the dialed string.

The pattern represents the smallest number of endpoints.

The pattern matches the dialed string.

The pattern represents the largest number of endpoints.

Answer

A, D

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