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

12

13

15

16

opening greeting, welcome, and default system

caller input, greetings, and transfer

greetings, operator, and closed

opening greeting, operator, and goodbye

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.

Check the box to enable application-level authorization.

Check the box to allow charging header.

Check the box to accept unsolicited notification.

Check the box to transmit security status.

Check the box to accept replaces header.

Cisco CallManager SNMP

SNMP Master Agent

Connection SNMP Agent

Host Resources Agent

G722.1

iLBC

G.711alaw

G.729A

Select “Redirecting Diversion Header Delivery— Inbound’ on the SIP trunk

Enable “Accept presence subscription” on the SIP trunk security profile

Select “Redirecting Diversion Header Delivery – outbound” on the SIP trunk

Enable “Accept unsolicited notification” on the SIP Trunk security profile

9.011#

9.011@

9.011!

9.011*

on a server in the Webex Data Center

on a dedicated on-premises server

on a Cisco Expressway-C connector host server

on an on-premises Microsoft Active Directory server

AF41

CS3

CS4

EF

Option A

Option B

Option C

Option D

66

143

150

166

local authentication

certificate-based

mutual TLS

Auth-based

voice packet loss that is greater than 1 %

excessive network latency

varying delay of received voice packets

varying delay of sent voice packets

G.722

G.711

G.729

G.723

It functions as an analog telephony adapter.

It has remote endpoint enrollment with Certificate Authority Proxy Function.

It gives directory access for remote users via Cisco Directory Integration.

It provides access to on-premises Cisco Unified Communications infrastructure for remote endpoints.

partition

route partem

translation pattern

CSS

route group

The translation rule is not matched because DNIS does not end with a "$".

The translation rule is matched and the translated number is 02553431234.

The translation rule is matched and the translated number is 025553431234.

The translation rule is not matched because DNIS contains"-".

Unsolicited NOTIFY

200 ok

SIP Register

TCP port 5060

6

5

4

3

isdn supp-service name calling

isdn outgoing display-ie

isdn enable did

isdn send display le

\+.001! pre-dot 1 international

\+1.1 none pre-dot 001 international

\+.! pre-dot 00 international

613XXXXXXX none +011 internationa

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

Voice packets are classified and marked

All devices use wired connections instead of wireless connections

The network meets bandwidth requirements

Cisco UBE manages voice traffic, not data traffic

MTP is enable on the sip trunk to cisco Unified Border Elemnet

iLBC to G.711ulaw

G.728br8 to G.711alaw

G.729r8 to G.711ulaw

G.722 to G.729r8

G.729ar8 to G.711alaw

Verify that the correct calling search space is selected for the inbound Calls section

Verify that the run on all active United CM Nodes checkbox is enabled

Verify that the Significant Digits field for inbound Calls is set to All.

Verify that the PSTN Access checkbox is enabled.

The phone and access layer switch negotiate the classification of packets.

Cisco UCM determines how the voice packets are classified.

The voice packets are classified by the phone and the classification is accepted.

The voice packets are not trusted, and the access layer switch reclassifies the packets.

A limit of 900 kb is the total amount of bandwidth used for the sum of calls. The use of compression does not impact the amount of calls because only the codec definition is considered when calculating the CAC in Cisco UCM

A limit of 900 kb is the total amount of actual used RTP traffic so the oversubscription can be done for multiple calls adding its maximum RTP stream requirements.

A limit of 900 kb is the total amount of bandwidth used for the sum of the media and signaling in the calls The system allows for an actual of 20% overhead of IP packet headers This is to compare the use of ISDN lines with no IP overhead.

A limit of 900 kb is the total amount of bandwidth used for the sum of calls of the defined codec including headers and media traffic

8080 / TCP

6970 / UDP

443 /TCP

6970 / TCP

Cisco SIP Proxy

Cisco XCP Text Conference Manager

Cisco XCP Router

Cisco XCP XMPP Federation Manager

Incorrect partition

No region relationship

Network congestion

Incorrect QoS

Incompatible codec

provides high availability for the collaboration infrastructure

provides certificate-based security

reduces latency for audio and video communications

enables the directory address book in Cisco UCM

LAN access ingress and the WAN access ingress

LAN access egress

LAN access ingress

LAN access egress and the WAN access egress

403 Forbidden errors on SIP calls

excessive jitter

unexpected shut-down on Cisco Unified Communications Manager

packet fragmentation

DTMF conversion

SIP early offer

transrating

call queuing

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

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

Change line partition to Partition_A

Change line CSS to only contain Partition_B

Set the user's line CSS to

Set the user's device CSS to

Unified CM sends an NTPv4 packet.

Database replication is not synchronized on the Unified CM nodes.

The cluster loses access to port 124 at the firewall.

The date/time group on all phones defaults to the time zone of the engineer.

MRA

trunk

AAR

H.323

The media resource is unavailable.

The destination number rejects the call.

The destination number is busy.

The dialed number is not assigned to an endpoint.

nongatekeeper-controlled intercluster trunk

SIP intercluster trunk

H.225 trunk

gatekeeper-controlled inter cluster trunk

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.

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.

It provides additional granularity by allowing a user to create classes

It minimizes jitter by implementing a priority queue for voice traffic

It uses a priority queue for voice traffic to avoid jitter.

It provides additional granularity by allowing a user to define custom class

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.

Option A

Option B

Option C

Option D

0

1

3

5

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

On Cisco UCM Administration Page Device > Phone > Phone Configuration page

On Cisco UCM Administrator Page server > Cisco UCM

On Cisco UCM Administrator page system > Device Pool > Cisco UCM group

On Cisco UCM Administrator page system > Enterprise Parameters

H. 323

SIP

SCCP

MGCP

RTP

TLS-based

certificate-based

registration-based

authentication-based

OAuth-based

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.

add voice-codecs 20

add session codec 20

add codec preference 20

add voice-class codec 20

It automatically identifies traffic

It provides reserve bandwidth by using RSVP

It categorizes traffic.

It guarantees end-to-end bandwidth for traffic.

HTTPS

H.323

SIP

SCCP

SRTP

10

12

15

17

expedited forwarding

class selector

assured forwarding

default

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

Change the common phone profile.

Change the SIP dial rules.

Change the SIP profile.

Change the phone security profile.

isdn incoming-voice voice

pri-group timeslots 1-24

card type t1 0 2

voice-port 0/2/0:23

SRST

Route Group

Cisco UCM

Device Pool

SSO

H.323 registration proxy to Cisco Unified Communications Manager

MGCP gateway registration

SRST

show IIdp status

show IIdp traffic

show IIdp

show IIdp run

Option A

Option B

Option C

Option D

From Users, select Users, select New Users, and then click Import from LDAP.

From Users, select Import Users, set Find End Users In to LDAP Directory, and the click Find.

From LDAP. select LDAP Directory Configuration, and then click Perform Full Sync Now.

From Users, select Synch Users, set Find Imported Users in to LDAP Directory, and then dick Find.

Endpoint>Expressway-C>Expressway-E>Cisco UCM

Endpoint>Expressway-E>Expressway-C> Cisco UCM

Endpoint>Expressway-E> Cisco UCM

Endpoint>Expressway-C> Cisco UCM

route partition

transformation pattern

directory number

CTl port

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

Ensure that ad hoc conference calls are dropped if an external user is add.

Call forward settings (ALL/Busy/No Answer) are restricted to internal extensions in the network

Add an additional firewall between the Cisco UCM server and the Expressway Core server.

For extension mobility, logged-out CSS is restricted to internal extensions and emergencies.

Forced authorization code is used to recognize a dialing extension and authorize an international call.

Priority = 100, Weight = 90

Priority = 10, Weight = 5

Priority = 10, Weight = 10

Priority = 5, Weight = 70

iq

message

type

presence

Enable SSH Access under Product Specific Configuration Layout in Cisco UCM

Disable Web Access under Product Specific Configuration Layout in Cisco UCM

Set a username and password under Secure Shell information in Cisco UCM

Enable Settings Access under Product Specific Configuration Layout in Cisco UCM

Enable FIPS Mode under Product Specific Configuration Layout in Cisco UCM

Wart until 10 pm on March 4th when the user is automatically removed from Cisco UCM.

Wait until 10 pm on March 5th when the user is automatically removed from Cisco UCM.

Wait until 3 15 a.m. on March 6th for garbage collection to remove the user from Cisco UCM.

Wait until 315am on March 5th for garbage collection to remove the user from Cisco UCM.

Create a recording that is applied to a service voicemailbox in Cisco UCM.

Configure a directory handler to respond to the calls in Cisco UCM.

Create a call handler template to apply to a call handler in Cisco Unity Connection.

Configure a telephony service in Cisco Unity Connection to play prerecorded messages applying this service to a number.

multifactor authentication

Active Directory synchronization

attribute-based access control

single sign-on with Google

Client ID/Client Secret

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.

phantom

replica

hub_none

shadow

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

5 seconds

10 seconds

15 seconds

20 seconds

The call fails because device mobility is turned on, and the phone is not configured in New York. The engineer must configure which sites the device should be roaming to.

The call egresses in San Francisco because the user uses device mobility and is allowed to roam while still keeping the number and resources assigned in San Francisco.

The call fails because the Standard Local Route Group is being used only it no configuration is set for the device pools.

The call egresses in New York because the device automatically is assigned a New York device pool and uses the local gateway.

IOS XE Hardware MTP

IOS XE Software MTP

IOS XE Hardware Transcoder

IOS XE Hardware Conference Bridge

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.

the TFTP server details

the application credentials for Cisco UCM

the security password for Cisco UCM

the FTP server details

isdn incoming-voice voice

isdn switch-type basic-ni

isdn switch-type basic-qsig

isdn switch-type primary-qsig

The traversal zone on Expressway-c points to Expressway-e through the peer address field on the traversal zone, which specifies the Expressway-e server address. For dual NIC deployments, set the Expressway-e address using an FQDN that resolves the IP address of the internal interface

The external firewall must allow these inbound connections to Expressway: SIP: TCP 5061; HTTPS: TCP 8443; XMPP TCP 5222; media: UDP 36002 to 59999

Do not use a shared address for Expressway-e and Expressway-c, as the firewall

cannot distinguish between them. If static NAT for IP addressing on Expressway-e is used, ensure that any NAT operation on expressway-c does not resolve the same traffic IP address. Shared NAT IS not supported

The internal firewall must allow these inbound and outbound connections between

expressway - c and Expressway-e :sip;HTTPS(tunneled over SSH between C and E.

TCP 2222: TCP 7001: Traversal Media: UDP 2776 to 2777(or 36000 to 36011 for large VM/appliance); XMPP:TCP 7400

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.

Enforce ad-hoc conference restrictions.

Set the appropriate service parameter.

Implement time-of-day routing.

Use the correct route filters.

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

It stands for Call Allocation Control.

It allows for bandwidth restrictions between locations.

It allows for codec restrictions between locations.

It allows for devices making call restrictions between locations based on numbers.

It allows for QoS-based restrictions between locations.

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.

SIP003096D2D5CB.cnf.xml

SEP003096D2D5CB.cnf.xml

SEP003096D2D5CB.cnf

SIP003096D2D5CB.cnf

Scavenger

best effort

voice

call signaling

REGISTER

SUBSCRIBE

PRACK

NOTIFY

UPDATE

G.711ulaw

iLBC

G.711alaw

G.722

The source phone sends a 180 Ringing signal to the Cisco UCM. which sends a 180 Ringing signal to the destination phone. The same process applies to 11 BYE messages.

The source phone must signal to the destination phone that it is ringing, and the destination phone signals back with a 180 Ringing message. The same process applies to 11 BYE messages.

The calls have an MTP in the call path due to different codec support. The calls are subsequently split into two call legs.

The destination phone signals back to the Cisco UCM that it is ringing, and the Cisco UCM signals back to the source phone.

core router

voice VLAN

access switch

distribution switch

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

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

6

7

8

11

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.

24

32

34

46

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

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.

encryption in transit

end-to-end encryption for meetings

Hybrid Data Security end-to-end

encryption of content

voice and video transcoding

endpoint registration

voice and video conferencing

Intercluster extension mobility

Cisco VPN Client

Cisco Umbrella

Cisco Webex

Cisco Unified Communications Mobile and Remote Access

Adjust the T302 timer from the default of 15 seconds to 5 seconds to shorten the interdigit timer

Change the code if the problem persists

Advise the user to hang up and try again

Do not wait for the tones immediately dial the FAC and CMC

Advise the user to press the"#"button after dialing the FAC and CMC codes

VPN connection toward the internal UC resources

SIP header modification

B2B calls

device registration over the Internet

IP to PSTN call connectivity

*

@

!

.

linecode ami

linecode b8zs

linecode hdb3

linecode esf

Confirm the phone load name in the phone configuration.

Assign a universal device template to the phone.

Assign a phone security profile with secure SIP

Set the prepare cluster for rollback to pre-8.0 enterprise parameter to true.

< 10 percent packet loss and < 10 ms jitter

< 1 percent packet loss and < 30 ms jitter

< 9 percent packet loss and < 30 ms jitter

< 3 percent packet loss and < 60 ms jitter

13XX

13[25-8]6

13!#

13[A3-9)6

less than or equal to 150 ms of one-way latency

less than or equal to 60 ms jitter

less than or equal to 300 ms of one-way latency

less than or equal to 10 percent packet loss