Exam Dumps 350-801 Practice Free Latest Cisco Practice Tests [Q167-Q185]

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Exam Dumps 350-801 Practice Free Latest Cisco Practice Tests

350-801 Exam Questions | Real 350-801 Practice Dumps

Cisco 350-801 Exam Syllabus Topics:

Section	Weight	Objectives
Security	10%	– Collaboration security fundamentals 1. Endpoint authentication and registration security 2. Secure SIP trunking and certificates 3. TLS/SRTP encryption for voice and video
Quality of Service (QoS)	10%	– Collaboration QoS implementation 1. Classification and marking of voice/video traffic 2. End-to-end QoS deployment models 3. Queuing and congestion management
Network Infrastructure	25%	– Voice and video network infrastructure 1. Gateways and SIP trunking configuration concepts 2. Media resources (transcoders, MTP, conference bridges) 3. IP addressing and routing for collaboration endpoints
Collaboration Applications	15%	– Cisco collaboration application services 1. Unity Connection voicemail services 2. IM and Presence services 3. Cisco Webex integration concepts
Architecture	20%	– Cisco collaboration architecture fundamentals 1. SIP architecture and call signaling flow basics 2. Cisco collaboration components overview (CUCM, IM&P, Unity Connection) 3. On-premises and cloud deployment models
Call Control and Dial Plan	20%	– Call control configuration and management 1. Dial plan design and implementation 2. Cisco Unified Communications Manager (CUCM) call routing 3. SIP and SCCP endpoints management

NEW QUESTION 167
Which two protocols are proxied over an Expressway-E/C pair when a MRA login including phone services is performed? (Choose two.)

SCCP

HTTPS

H.323

SRTP

SIP

https://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/expressway/config_guide/X8-
10/Mobile-Remote-Access-via-Expressway-Deployment-Guide-X8-10.pdf

NEW QUESTION 168
During the Cisco IP Phone registration process, the TFTP download (ails. What are two reasons (or this issue?
(Choose two.)

The DNS server was not specified, which is needed to resolve the DHCP server IP address.

Option 100 string was not specified, or an incorrect Option 100 string was specified.

The Cisco IP Phone does not know the IP address of the TFTP server.

The Cisco IP Phone does not know the IP address of any of the Cisco UCM Subscriber nodes.

Option 150 string was not specified, or an incorrect Option 150 string was specified.

NEW QUESTION 169
An engineer is setting up a new voicemail server, and the requirements state that the integration for call control must be secure. During the configuration the engineer wants to select next Generation Encryption? What is the prerequisite for using this option to ensure an encrypted SIP integration?

TCP connection using port 5061 with explicit TLS1.2 and ACME-signed SIP certificates

secure TLS connection and Tomcat certificates

secure TLS connection and SIP certificates

TCP connection using port 5061 with explicit TLS1.2 and signed SIP certificates

NEW QUESTION 170
Which two configuration elements are part of the Cisco Unified Communications Manager toll-fraud prevention? (Choose two.)

SIP trunk security profile

Calling Search Space

SUBSCRIBE Calling Search Space

feature control policy

partition

Section: Call Control
Explanation/Reference:

NEW QUESTION 171
An administrator must implement toll fraud prevention on cisco UCM using these parameters:
– Enable Forced Authorization Code 112211.
– Set an authorization level of 3 for the route pattern 8005551212.
– Require no access code to dial 10-digit numbers.
How must the route pattern be implemented?

Pattern = 3.800xxxxxx

Pattern = 8005551212.1122113

Pattern = 8005xxxxxx

Pattern = 1122113.8005551212

https://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified- communications-manager-callmanager/81541-fac-config-ex.html

NEW QUESTION 172
An administrator executes the debug isdn q931 command while debugging a failed call.
After a test call is placed, the logs return a disconnect cause code of 1.
What is the cause of this problem?

The destination number rejects the call

The destination number is busy

The media resource is unavailable

The dialed number is not assigned to an endpoint

NEW QUESTION 173
What is the purpose of configuring locations in a Cisco Webex dial plan?

to specify the maximum bandwidth to allocate per Webex call

to choose how many Webex data centers to use for redundancy

to conserve bandwidth by selecting the closest data center

to allow the use of an existing PSTN provider for local calls

In Webex Calling, locations group users or devices by geographical site and are used to manage PSTN access, whether on-premises or cloud connected PSTN. This allows calls to be routed appropriately based on the user’s location and the associated PSTN trunk or gateway.
Locations enable linking to a specific PSTN provider or trunk (such as a Local Gateway or Cloud Connected PSTN), so calls from users in that location can be routed through the correct PSTN connection for local calling.
Locations also support inter-site dialing by using location codes and routing prefixes, helping maintain consistent dialing habits and call routing across sites.
While bandwidth management and data center selection are important, these are not the primary purposes of configuring locations in the dial plan for Webex Calling.
Therefore, configuring locations primarily facilitates local PSTN call routing by associating users with the correct PSTN provider or trunk based on their site.

NEW QUESTION 174
A customer has Cisco Unity Connection that is integrated with LDAP. As a unity connection administrator, you have received a request to change the first name for VM user. Where must the change be performed?

Cisco unity connection

Cisco IM and presence

Cisco Unified Communications Manager end user

Active directory

NEW QUESTION 175
An engineer is configuring a Cisco IOS XE gateway to join three callers together in the same conversation.
Which IOS XE media resource must the engineer configure to accomplish this goal?

IOS XE Hardware MTP

IOS XE Software MTP

IOS XE Hardware Transcoder

IOS XE Hardware Conference Bridge

NEW QUESTION 176
Refer to the exhibit.

An administrator configures a secure SIP trunk on Cisco UCM.
Which value is needed in the secure certificate subject or subject alternate name field to accomplish this task?

The fully qualified domain name of the remote device that is configured on the SIP trunk.

The common name of the Cisco UCM CallManager certificate.

The full qualified domain name of all Cisco UCM nodes that run the CallManager service.

The common name of the remote device certificates.

NEW QUESTION 177
What are two functions of Cisco Expressway in the Collaboration Edge? (Choose two.)

Expressway-C provides encryption (or Mobile and Remote Access but not (or business-to-business communications.

The Expressway-C and Expressway-E pair can enable connectivity from the corporate network to the PSTN via a T1/E1 trunk.

The Expressway-C and Expressway-E pair can interconnect H.323-to-SIP calls for voice.

Expressway-E provides a VPN entry point for Cisco IP phones with a Cisco AnyConnect client using authentication based on certificates.

Expressway-E provides a perimeter network that separates the enterprise network from the Internet.

NEW QUESTION 178
What is the major difference between the two possible Cisco IM and presence high-availability modes?

Balanced mode provides user load balancing and user failover only for manually generated failovers.
Active/standby made provides an unconfigured standby node in the event of an outage, but it does not provide load balancing.

Balanced mode provides user load balancing and user failover in the event of an outage.
Active/standby mode provides an always on standby node in the event of an outage, but it does not provide load balancing.

Balanced mode does not provide user load balancing, but it provides user failover in the event of an outage.
Active/standby mode provides an always on standby node in the event of an outage, but is does not provide load balancing.

Balanced mode provides user load failover in the event of an outage.
Active/standby mode provides an always on standby node in the event of an outage, and is also provides load balancing.

Balanced mode: This mode provides redundant high availability with automatic user load balancing and user failover in the event that one nodes fails because of component failure or power outage.
Active/standby mode: The standby node automatically takes over for the active node if the active node fails. It does not provide automatic load balancing.

NEW QUESTION 179
Given the H.323 gateway configuration and using Cisco best practices, how must the called party transformation pattern be configured to ensure that a proper ISDN type of number is set?

NEW QUESTION 180
An administrator executes the debug isdn q931 command while debugging a failed call.
After a test call is placed, the logs return a disconnect cause code of 1.
What is the cause of this problem?

The destination number rejects the call

The destination number is busy

The media resource is unavailable

The dialed number is not assigned to an endpoint

https://www.cisco.com/c/en/us/support/docs/dial-access/integrated-services-digital-networks-isdn- channel-associated-signaling-cas/8607-isdn-disc- code.html

NEW QUESTION 181
Which action prevent toll fraud in Cisco Unified Communications Manager?

Configure ad hoc conference restriction.

Implement toll fraud restriction in the Cisco IOS router.

Allow off-net to off-net transfers.

Implement route patterns in Cisco Unified CM.

https://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified- communications-manager-express/107626-cme-toll-fraud.html

NEW QUESTION 182

Refer to the exhibit. An engineer configures ISDN on a voice gateway. The provider confirms that the PRI is configured with 10 channels the engineer ordered and is working from the provider side, but the engineer cannot get a B-channel to carry voice. The rest of the configuration for the serial interface and voice network is functioning correctly. Which actions must be taken to carry voice?

The engineer must activate the controller card on the voice gateway before configuring the device.

The engineer used a T1 interface but must use an E1 interface.

The pri-group timeslots command must be 0-9 for the 10 channels because all values on a router start with 0.

The engineer must manually revert the order of using the channels.

NEW QUESTION 183
Which IP Precedence value is used to classify a call signalling packet?

NEW QUESTION 184
An engineer roust configure DTMF relay on a Cisco Unified Border Element by using RFC2833 as the preferred relay mechanism and KPML as the next preferred relay mechanism. The engineer logs in to the CUBE and enters the dial-peer configuration level. Which command should be run at dial-peer configuration level?

dtmf-relay sip-kvmi rtp-nte

dtmf-relay rtp-nte sip-kpml

dtmf-relay sip-kgml rtp-inband

dtmf-relay rtp-inband sip-kvmi

The dtmf-relay command is used to configure DTMF relay on a Cisco Unified Border Element.
The rtp-nte option specifies that RFC2833 is the preferred relay mechanism, and the sip-kpml option specifies that KPML is the next preferred relay mechanism.

NEW QUESTION 185
Refer to the exhibit.

An engineer verifies the configured of an MGCP gateway. The commands are already configured. Which command is necessary to enable MGCP?