350-801 Implementing Cisco Collaboration Core Technologies (350-801 CLCOR) v2.0 Question and Answers

Pattern usage, IPv4 pattern, IPv6 pattern, and description

Pattern usage, IPv4 pattern, and SIP trunk/Route list

Pattern usage, IPv6 pattern, and SIP trunk/Route list

SIP trunk/Route list, description, and IPv4 pattern

G.729

ILBC

G.711ulaw

G.728

G.722

G.711

G.729

G.723

software MTP on Cisco IOS Software

software MTP on Cisco UCM

software transcoder on Cisco UCM

hardware transcoder on Cisco IOS Software

hardware MTP on Cisco IOS Software

to handle calls between Webex Calling and Cisco Calling Plans

to handle calls between Webex Calling and Cloud Connected PSTN

to handle calls between Cisco IJCM and Webex Calling

to handle calls between the Public Switched Telephone Network and Webex Calling

31.2 kbps

38.4 kbps

55.2 kbps

87.2 kbps

10 percent

13 percent

15 percent

20 percent

IP TCP/TLS (A) + SIP TCP/TLS (B) + SIP TLS (C)

SIP TCP/TLS (A) + SIP TCP/TLS (B) + SIP TCP/TLS (C)

SIP TLS (A) + SIP TLS (B) + SIP TLS (C)

SIP TCP/TLS (A) + SIP TLS (B) + SIP TLS (C)

/+.! Route Pattern

\+.! Route pattern

\+.! Translation Pattern

\+1.[2-9]XX[2-9] XXXXXXX called Party Transformation Pattern

H.225 Block Setup Destination

Drop Ad Hoc Conference

Enterprise Feature Access Code for Conference

Block OffNet To OffNet Transfer

jitter increase due to variable delay

discards in the interface of routers and switches

video loss due to video frame corruption

gaps in the voice due to packet loss

platform qos trust extend cos 3

platform qos trust extend

platform qos extend trust

platform qos trust extend cos 5

Cisco United CM Administration > User Management > User Settings > End User CAPF Profle

Cisco Unified CM Administration > User Management > User Settings > User Group

Cisco Unified CM Administration > User Management > User Settings > Access Control Group

Cisco Unified CM Administration > User Management > End User > End User Configuration

+13135551212@192.168.1.3

ciscocm1.jupiter.com

\+13135551212@192.168.1.3

192.168.1.3

Change Cisco UCM SIP trunk DTMF Signaling Method to -sip-notify " .

Change Cisco UCM SIP trunk DTMF Signaling Method to " sip-kpml " .

Change Cisco Unified Border Element configuration to " dtmf-relay sip-notify sip-kprnl " .

Change Cisco Unified Border Element configuration to " dtmf-relay rtp-nte " .

dtmf-relay cisco-rtp

dtmf-relay sip-kpml cisco-rtp

sip-notify dtmf-relay rtp-nte

dtmf-relay rtp-nte sip-notify

Traffic policing can be applied to outbound traffic on an interface.

Traffic policing can be applied to inbound traffic on an interface.

Traffic policing uses scheduling functions to transmit delayed packets.

When traffic rate equals the configured maximum rate, excess traffic is remarked

Traffic policing queues excess packets.

MGCP devices

SCCP devices

pass-through MTPs

H.323 trunks

DX80

It creates a trust relationship with the called party.

It enables the Use Trusted Relay Point setting from the associated common device configuration.

It enables Cisco UCM to insert an MTP or transcoder designated as a TRP.

If TRP is allocated and MTP is also required for the endpoint. calls fail.

AF41 has higher drop precedence than AF42. which has higher drop precedence than AF43.

Voice and video calls are marked with different DSCP values and placed in different queues.

AF43 has higher drop precedence than AF42 but lower drop precedence than AF41.

RTP traffic from voice and video calls is marked EF and placed in the same queue.

A TCP socket is opened to communicate between the DSP farm transcoder and CUBE.

An external API is used by CUBE to access DSP farm resources for transcoding.

The DSPfarm transcoding profile and CUBE must be configured on the same platform.

Transcoding can be used for SRTP pass through calls

Block

Transfer to Number

Reject

Restrict

Restart NTPD on the Cisco UCM server.

Delete and re-add the new NTP server via the Cisco UCM command-line interface

Start the NTP service on the new NTP server

Configure the " reach " value as " 377 " for the new NTP server.

Check on the phone side because it is using an incorrect source port to connect with Cisco UCM server

Check on Cisco UCM side because the CTL fie was not downloaded successfuly

Check on the network to verify why the connectivity is not happening correctly

Check on the phone side because It is attempting to register to an incorrect Cisco UCM server

Change the Calling Search Space of the user s device

Uncheck the Urgent Priority box on the Route Pattern Configuration page in Cisco UCM

Change the Calling Search Space of the user s line

Check the Require Forced Authorization Code box on the Route Pattern Configuration page in Cisco UCM

Change the Route Option on the Route Pattern Configuration page in Cisco UCM

It forwards calls when a line is busy by enabling the call waiting feature.

It forwards calls to the voicemail of a mobile phone when there is no answer.

It provides the ability to forward a Webex meeting to another user.

It chooses a tone to be played when forwarding all calls.

80

443

8080

8443

MRGL A and Audio Source 4

MRGL B and Audio Source 4

MRGL A and Audio Source 2

MRGL B and Audio Source 2

The phone is configured with an incorrect Device Security Profile

The SDP contains a private IP address that is not being NATed correctly

The phone is configured with an incorrect Device Security Profile

The MTP resource Is failing to be allocated within the calling MRG.

linecode ami

linecode b8zs

linecode hdb3

linecode esf

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

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.

Webex Meetings

Webex Messaging

Webex Events

Webex Calling

Option A

Option B

Option C

Option D

transfer rules

message settings

alternate extensions

greetings

Set Enable Activation Code enterprise parameter to True-

Manually provision new phones that have an activation code requirement

Create a Bulk Administration Tool provisioning template.

Generate 16-digit codes by using the Bilk Administration Tool

Enable Auto-Registration.

Associate the end user to the Standard CCM Super Users group

Associate the end user to a SIP Profile.

Disable Auto-Registration.

Option A

Option B

Option C

Option D

A number marked as Main is not eligible to be moved of deleted

A number can only belong to one location

A number must be marked Active to be eligible for deletion.

A number can belong to multiple locations.

A number cannot be moved while marked Active.

Auto-registration is disabled on the Cisco Unified Communications Manager nodes and the phone device does not have a DN configured.

The DHCP settings are incorrectly and the phone does not have an alternate TFTP defined.

The phone cannot download and install the latest firmware.

The DN assigned to the phone is already in use by another SIP phone.

The DN configuration for this phone is shared with SCCP phone, which is not supported.

Verify that the switch port of the phone is enabled.

Verify that the RJ-11 cable is plugged into the PC port.

Disable HSRP on the access layer switch.

Check the RJ-65 cable.

Verify that the phone ' s network can access the option 150 server.

24

32

34

46

Option A

Option B

Option C

Option D

whether auto-registration is enabled in Cisco UCM

whether the Initial Trust List and Certificate Trust List files on the phone are correct

whether the phone ts in the correct VLAN

whether the phone ' s MAC address is correct In Cisco UCM

HTTP

XML

UDP

SIP

Nothing, the PRI is sending keepalives.

Layer 2 is down on the controller.

PRI does not have an IP address configured on the interface.

Layer 1 is down on the controller.

H.323 endpoint registrations

Mobile and Remote Access

SIP gateway for PSTN providers

VTC bridge

Configure a SIP integration with Cisco UCM to sync users.

Configure an SCCP integration with Cisco UCM.

Configure an AXL server to access the Cisco UCM users.

Configure an active directory to sync the users who will have a voicemail box.

Low response time between endpoints

Provisioned bandwidth of the link

Variable bit rate of the video stream

Bursly behavior of video traffic

Deploy RSVP to improve VoIP QoS only where it will have a positive impact on quality and functionality where there is limited bandwidth and frequent network congestion.

Use multiple egress queues to provide priority queuing of RTP voice packet streams and the ability to classify or reclassify traffic and establish a network trust boundary.

Map audio and video streams of video calls (AF41 andAF42) to a class-based queue with weighted random early detection.

Implement IP RTP header compression on the serial interface to reduce the bandwidth required per voice call on point-to-point links.

Use 802 1Q trucking and 802.1p for proper treatment of Layer 2 CoS packet marking on ports with phones connected.

analog gateways

unsecure PCs

Cisco Telepresence endpoints

handheld mobile devices

Cisco IP phones

Use multiple egress queues to provide priority queuing of RTP voice packet streams and the ability to classify or reclassify traffic and establish a network trust boundary.

Use 802, 1Q trunking and 802.1p for proper treatment of Layer 2 CoS packet marking on ports with phones connected.

Implement IP RTP header compression on the serial Interface to reduce the bandwidth required per voice call on point-to-point links.

Deploy RSVP to improve VoIP QoS only where it can have a positive impact on quality and functionality where there is limited bandwidth and frequent network congestion.

Map audio and video streams of video calls (AF41 and AF42) to a class-based queue with weighted random early detection.

CSS

device pool

route list

route pattern

directory handler

call routing table

interview handler

call handler

If a call uses G.729. Cisco UCM subtracts 16k.

If a call uses G.711, Cisco UCM subtracts 64k

If a call uses G.711, Cisco UCM subtracts 80k

If a call uses G.729. Cisco UCM subtracts 24k.

If a call uses G.729. Cisco UCM subtracts 40k

The analog port failed to register to Cisco UCM with an error code 200.

The MGCP Gateway is not communicating with the Cisco UCM.

The analog port is currently shut down.

The analog port is registered to Cisco UCM.

A data file in Abstract Syntax Notation One format must be uploaded to Cisco UCM

A server template must be created in Cisco UCM

A data file in comma-separated values format must be uploaded to Cisco UCM

A data file in Extensible Markup Language format must be uploaded to Cisco UCM

A device template must be created in Cisco UCM

Option A

Option B

Option C

Option D

MP4A-LATM

G.711

OPUS

G.729

It improves monitoring and troubleshooting.

It determines the least cost path between sites.

It optimizes the minimum required bandwidth.

It ensures the best possible user experience

ccm-manager redundant-host

ccm-manager fallback-mgcp

mgcpapp

mgcp call-agent

Perform a web search for the error message

This is an informational message and no action is required

Run the show support command to gather more details

Block the TLS request from the untrusted host at the network layer

local authentication

certificate-based

mutual TLS

Auth-based

Cisco Expressway-E, Cisco IM and Presence Server, and Cisco Video Communication Server

Cisco Unified Border Element, Cisco IM and Presence Server and Cisco Video Communication Server

Cisco Expressway-E, Cisco Expressway-C, and Cisco UCM

Cisco Unified Border Element, Cisco UCM, and Cisco Video Communication Server

UDP161

TCP 161

UDP 162

TCP 80

5 seconds

10 seconds

15 seconds

20 seconds

Cisco CallManager SNMP

SNMP Master Agent

Connection SNMP Agent

Host Resources Agent

Configure access control lists and mark traffic with DSCP values

Classify and mark traffic at the core switch

Classify and mark traffic at the access switch

Configure group policy on the VLAN interface

Cisco SIP Proxy

Cisco XCP Text Conference Manager

Cisco XCP Router

Cisco XCP XMPP Federation Manager

phone auto-registration

phones that are on private VLANs

nonstandard TCP ports

access to the phone settings

Route pattern 56895531 has a route group that contains out of service devices

Route pattern 56895531 has " Block this pattern " enabled

56436223 cannot complete the call due to incorrect class of control configuration

Route pattern for 56896531 has no route partition configured

1,000

2,000

10,000

12,000

Configure a Forced Authorization Code on the international route pattern.

Block international dial patterns in the SIP trunk CSS.

Set Call Forward All CSS to restrict international dial patterns.

Set the Call Classification to OnNet for the international route pattern.

Check Route Next Hop By Calling Party Number on the international route pattern.

secure signaling and media

authentication of SIP messages

reduced authentication times

usage of self-signed certificates

domain models

views models

network models

device models

data models

It collects information from callers by playing a series of questions and recording the answers.

It controls outgoing calls by allowing an administrator to specify the numbers that Cisco Unity Connection dials to transfer calls, notify users of messages, and deliver faxes.

It provides access to a corporate directory by playing an audio list that users and outside callers use to reach users and leave messages.

It answers cans, take messages, and provide menus of options.

digest user

manufacturer

serial number350-801 2023-4

authorization code

REGISTER

SUBSCRIBE

PRACK

NOTIFY

UPDATE

RTP

TLS

TCP

SIP

manipulation of dialed destination

path selection

calling privileges

endpoint addressing

The incoming calling search space of the SIP trunk does not include the partition of the line in the IP phone

The gateway is configured with the wrong device pool

The service parameter related to Öffnet to Offnet Call Transfer is set to TRUE

The IP phone is configured with the wrong region

Set up is not required.

Set up for each group to use IPv4 and IPv6.

Set up media ports for each port group to use IPv4.

Set up IPv4 and IPv6 in Cisco Unified CM.

Add a transcoder that supports G.711ulaw and G.729.

Nothing; both sides support G.729.

Add a media termination point that supports G.711ulaw and G.729.

Nothing both sides support payload type 101.

The end user could incorrectly tag their traffic to advertise their PC as a default gateway.

The end user could incorrectly tag their traffic to bypass firewalls.

There is no reason not to include an end user ' s PC device in a QoS trust boundary.

The end user may incorrectly tag their traffic to be prioritized over other network traffic.

Define Cluster Fully Qualified Domain Name under Servers in Cisco UCM.

Change the Destination Address to a Fully Qualified Domain Name on the SIP trunk.

Define Cluster Fully Qualified Domain Name in Enterprise Parameters.

Set the SIPS URI Handling to True in CallManager Service Parameters.

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.

Restart the Cisco DHCP Monitor Service under Cisco Unified Serviceability

Add the new DHCP server to the primary DNS server

Restart the TFTP service under Cisco Unified Serviceability.

Add a DHCP subnet to the DHCP server under Cisco UCM Administration.

The packets arrive with frame errors.

The packets arrive at varying time intervals.

The packets arrive out of sequence.

The packets never arrive due to tail drop.

The packets arrive at uniform time intervals.

route list

device pool

CSS

route pattern

The cable is unplugged.

The controller shut down.

The clock source is incorrect.

The framing is configured incorrectly.

It only identifies calls as spam if they are from international numbers

It relies solely on a user-maintained block list

It automatically flags any call that originates from an unknown or unlisted number

It relies on the originating and terminating service providers to analyze call patterns and caller ID information

Configure a FAC code in the CSS level and assign the CSS in the route pattern

Configure a route pattern 900xxxxxx with " Block this pattern " enabled

Configure a FAC code with level 5 and enable FAC m a route pattern to 900xxxxxx

Block international dial patterns via CSS in the SIP trunk to PSTN

EF

CS3

AF41

AF31

IOS XE Hardware MTP

IOS XE Software MTP

IOS XE Hardware Transcoder

IOS XE Hardware Conference Bridge

CTI port stales displayed in RTMT monitoring

call trace logs showing the dialed number and matched route pattern

registration status of al IP phones in RTMT

CPU utilization on the Cisco UCM server during call attempts

VPN-based enterprise access for a subset of Cisco Unified IP Phone models

secure reverse proxy firewall traversal connectivity

the ability 10 register third-party SIP or H 323 devices on Cisco UCM without requiring VPN

the ability of Cisco IP Phones to access the enterprise through VPN connection

the ability for remote users and their devices to access and consume enterprise collaboration applications and services

When using media flow-around, the can signaling goes through the Cisco Unified Border Element, but media is not passed through It.

When using media flow-through, call signaling goes through the Cisco Unified Border Element, but media does not

When using media flow-through, the call signaling and media are passed through the Cisco Unified Border Element

When using media flow-through, the call signaling goes through the Cisco Unified Border Element, but media is not passed through When using media flow-around, both call signaling and media do not go through the Cisco Unified Border Element

Change the Attempt Sign in rule to Inactive

Delete the Attempt Sign in rule

Delete the aaroot rule

Change the order of the rules

Incorrect partition

No region relationship

Network congestion

Incorrect QoS

Incompatible codec

accept out-of-dialog refer

accept out-of-band notification

transmit security status

allow changing header

accept unsolicited notification

Authenticate your Entra ID admin account to grant tenant-wide admin consent

Disable auto-sync after the initial configuration completes

Use the Entra ID Wizard App in Control Hub to map user attributes

Manually create user accounts in Webex Control Hub before synchronization

Import Entra ID verified domains Into Control Hub for domain validation

the TFTP server details

the application credentials for Cisco UCM

the security password for Cisco UCM

the FTP server details

PT_NTERNAL

PT_SIPJNTERNAL

PT_SIP_LOCAL

CSS_NTERNAL

The Cisco Discovery Protocol table is shared among devices.

Ping Sweep reports " unmanaged ' state devices.

System logs are collected in a Cisco Prime Collaboration Server.

Simple Network Managed Protocol is enabled on each device to poll specific values periodically.

Maximum Audio Bit Rate must be increased.

Maximum Session Bit Rate for Immersive Video Calls is too low.

Maximum Session Bit Rate for Video Calls is too low.

Cisco 8945 does not have a camera connected.

End users must have a secondary extension.

Cisco Extended Functions service must be running

End users must belong to Standard CCM Admin Users group, the Standard CCM End Users group, and the Standard CCM Self-Provisioning group.

End users must have a primary extension.

End users must be associated to a user profile or feature group template that includes a universal line template and universal device template.

Cisco Discovery Protocol is not enabled, and ping sweep is enabled

DNS is not enabled for the phones

DHCP option 150 is not set up correctly

The IP phone management service is not enabled under Services

Voice traffic must be marked " to DSCP EF.

Loss must be no more man 1 percent.

Voice traffic must be marked to DSCP AF41.

One-way latency must be no more than 200 ms.

Average one-way jitter is greater than 50 ms.

It automatically identifies traffic

It provides reserve bandwidth by using RSVP

It categorizes traffic.

It guarantees end-to-end bandwidth for traffic.

highly secure real-traversal technology to extend organizational reach.

additional visibility of the edge traffic in an organization.

complete endpoint registration and monitoring capabilities for devices that are local and remote.

session-based access to comprehensive collaboration for remote workers, without the need for a separate VPN client.

utilization and adoption metrics of all remotely connected devices.

xStatus HDMI Output

xStatus video Output

xconfiguration video Output

xcommand video status

phantom

replica

hub_none

shadow

Excess traffic is dropped.

Excess traffic is retransmitted.

All traffic is treated equally.

Excess traffic is queued.

iLBC to G.711ulaw

G.728br8 to G.711alaw

G.729r8 to G.711ulaw

G.722 to G.729r8

G.729ar8 to G.711alaw

Set the DELETEUSERDATA=r installation argument

Set the " HKEY_LOCAL_MACHINE,.Software ' «WOW6432Node .CiscoCollabHost -Eula_disable

Set the " HKEY_LCX^M._MACHINE .SoftwareCiscoCollabHo$t€ula Setting registry Eula_disable

Set the DEFAULT^ ' THEMEs-Dark " " installation argument

Set the " /quiet installation argument

Reboot Cisco Unity Connection.

Check the box " Redirecting Diversion Header Delivery - Outbound " , then reset the trunk.

Check the box ' Redirecting Diversion Header Delivery - Outbound " .

Review the conversation manager logs on Cisco Unity Connection.

5 min

10 min

30 min

60 min

TLS-based

certificate-based

registration-based

authentication-based

OAuth-based

certificate provisioning to remote endpoints

multiple line registration

all Cisco Unified Contact Center Express features

peer-to-peer file transfer

Interactive Connectivity Establishment passthrough

Not more than 8%

Not more than 1%

Not more than 3%

Not more than 5%

Set the Allow offNet to OffNet Transfer Cisco CallManager service parameter to false.

Classify an SIP devices as OffNet

Set the Block OffNet to OffNet Transfer CallManager service parameter to true.

Set the Call Classification Cisco CallManager service parameter to OffNet

Classify the SIP trunk to Cisco Unified Border Element as OffNet

one publisher, two TFTP, and two call processing subscribers

one publisher, two TFTP, and four call processing subscribers

one publisher, one TFTP, and one call processing subscriber

one publisher, one TFTP, and two call processing subscribers

The Webex Calling icense must be removed from the users and workspaces

The location must be renamed

The location must be assigned to a workspace

The location must be set as the default location

Add the audio forced command to voice service VoIP on the Cisco Unified Border Element.

Check the Retry Video Call as Audio on the SIP trunk.

Set Video Bandwidth in the Region settings to 0.

Change the Video Capabilities dropdown on the endpoint to Disabled.

Check the Send send-receive SDP in mid-call INVITE check box on the SIP trunk SIP profile.

Configure path-thru content sdp on the voice service.

Configure a hardcoded codec on the dial peers.

Configure a transcoder for video protocols.

Configure codec transparent on the dial peers.

Call policy service

Default zone access rules

TOLLFRAUD_APP

Class of service

\+.001! pre-dot 1 international

\+1.1 none pre-dot 001 international

\+.! pre-dot 00 international

613XXXXXXX none +011 internationa

Drop Ad Hoc Conference

Limit Ad Hoc Conference

Restrict Ad Hoc Conference

Monitor Ad Hoc Conference

voice service voip media flow-around

voice service voip media bulk-stats

voice service voip media anti-trombone

voice service voip media flow-through

m=audio 16444 RTP/AVP 4 101

m=audio 16444 RTP/AVP 15 101

m=audio 16444 RTP/AVP 9 101

m=audio 16444 RTP/AVP 18 101

Drop Ad Hoc Conference set to When Conference Controller Leaves

Advanced Ad Hoc Conference Enabled set to False

Drop Ad Hoc Conference set to When No OnNet Parties Remain in the Conference

Maximum Ad Hoc Conference set to 4

IM and Presence subclusters (configured on Cisco UCM)

Presence Redundancy Groups (configured on Cisco Unified IM and Presence)

IM and Presence subclusters (configured on Cisco Unified IM and Presence)

Presence Redundancy Groups (configured on Cisco UCM)

The SBC FQDN must match the common name in the certificate

Certificate-based trunks cannot coexist with registration-based trunks in the same route group

Wildcard certificates are required for certificate-based trunks

Webex Calling SBC uses a username and password for authentication

multifactor authentication

Active Directory synchronization

attribute-based access control

single sign-on with Google

Client ID/Client Secret

Enforce ad-hoc conference restrictions.

Set the appropriate service parameter.

Implement time-of-day routing.

Use the correct route filters.

Set the " Enable message waiting indicator " on the port group.

The line partition must be inside the rerouting CSS assigned to the Cisco Unity Connection SIP trunk.

The line partition must be inside the inbound CSS assigned to the CUC SIP trunk.

Assign an MWI extension on the mailbox.

Place calls in a queue

Greet callers with recorded prompts.

Filter incoming calls to prevent robocalls.

Identify high-priority calls.

Cisco UCM class of service

Cisco Unified Mobility

Cisco UCM Access Control Group restrictions

Cisco Unified Real-Time Monitoring Tool