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300-815 Implementing Cisco Advanced Call Control and Mobility Services (CLACCM) Questions and Answers

Questions 4

Refer to the exhibit.

300-815 Question 4

An engineer configures Cisco Unified Border Element to connect the enterprise VoIP network with a SIP telephony provider. Calls are not working in either direction. What must be configured in the dial peer 1 to fix the issue?

Options:

A.

answer-address 555 ……..

B.

codec g729

C.

session-protocol sipv2

D.

incoming called number 555…….

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Questions 5

An administrator is configuring a cluster for ILS and wants to limit the amount of entities that Cisco Unified Communications Manager can write to the database for data that is learned through ILS. Which service parameter is used to adjust this limit?

Options:

A.

ILS Max Number of Learned Objects in Database

B.

ILS Active Learned Object Upper Limit

C.

Global Data Service Parameter Limit

D.

Imported Dial Plan Replication Database Object Lower Limit

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Questions 6

An engineer configured extension mobility to allow users to roam between phones while taking their numbers and specific device settings with them. Users are now getting " Error:- XML Error [4| Parse Error " on their phones. Which action will resolve this issue?

Options:

A.

Set service provisioning lo internal.

B.

Lower the number of concurrent requests.

C.

Verify that the URL Is correctly configured.

D.

Set service provisioning to default.

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Questions 7

300-815 Question 7

Refer to the exhibit. An engineer is implementing the Cisco UCME and must set up a dial plan for the 7961 SIP phones to recognize digit strings dialed by users.

Which code snippet completes the configuration?

Options:

A.

" button 1:1 " under voice register pool

B.

" number 1 dialplan 1 " under the voice register pool

C.

" number 1 dp 1 " under the voice register pool

D.

" dialplan 1 " under the voice register pool

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Questions 8

Users in a small location report that audio calls work Tine, but only one video call works before the next one gets a busy signal. Three video conference rooms were recently installed by a third party. The audio and video devices are Cisco endpoints. The site uses video and audio calls, and a network investigation does not show saturation on the link The site is configured with QoS and CoS. All devices are registered on the headquarters Cisco UCM duster. What is the cause of this problem?

Options:

A.

The subzone setting for video is too low.

B.

The zone settings for video are too low.

C.

The location setting for video is too low.

D.

The pipe or link configuration for video is set to only one device.

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Questions 9

A new solution is configured to support internal, local, and international calling. Calling [+44 1111 1111] from one of the registered internal phones does not work. Local and internal calls seem to work without any problems. The configuration has patterns configured to match the failing dialed number [+44 …. ….] . The other configured patterns show [2…] for internal numbers and [555 …] for local numbers. International numbers use E.164 as recommended. What is missing to make this solution work?

Options:

A.

001 or 00 must be used instead of the + sign on Cisco UCM

B.

=+ cannot be used in a route pattern, only in a SIP pattern

C.

\ in front of the +

D.

/ in front of the +

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Questions 10

An engineer is reviewing the dial peer configuration on the Cisco Unified Border Element after users reported that DTMF digits are not recognized when they make an outbound SIP call to a phone system that supports only out-of-band DTMF.

Which configuration under outbound dial peer resolves the issue?

Options:

A.

dtmf-relay sip-notify

B.

dtmf-relay h245-signal

C.

dtmf-relay rtp-nte

D.

dtmf-relay h245-alphanumeric

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Questions 11

Refer to the exhibit.

300-815 Question 11

A no-way audio situation was experienced after the calling party on an outbound call placed the call on hold and then resumed from hold. Which action will resolve the issue?

Options:

A.

Assign an MTP to the SIP trunk.

B.

Restart the SIP trunk on Cisco UCM.

C.

Upgrade the Cisco Unified Border Element to 15.2.

D.

Upgrade the Cisco Unified Border Element device.

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Questions 12

What does VoIP trace in a Cisco Unified Border Element do?

Options:

A.

It stores all call trace data in system memory for failed calls.

B.

It stores all can trace data in system memory (or successful and failed calls

C.

It stores all call trace data in system log files for successful and failed calls.

D.

It stores all can trace data in system file logs for failed calls.

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Questions 13

Refer to the exhibit.

300-815 Question 13

An engineer is troubleshooting an issue where inbound calls to Cisco UCM with early media fail to establish. While investigating the issue, the engineer finds that Cisco UCM is set to require a PRACK. but the Cisco Unified Border Element is not sending it. Which command is causing this issue?

Options:

A.

voice-class sip rel1xx disable

B.

voice-class sip early-media update block

C.

voice-class midcall-signaling block

D.

voice-class sip conn-reuse

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Questions 14

An engineer is troubleshooting an intersite call between two endpoints where the call fails and the message “Not Enough Bandwidth” is displayed. G.729 codec is in use on both sites. First, calls are being properly routed, and the issue happens after the third call is established and the bandwidth utilization between the two sites is under 50%. Which configuration in Cisco UCM must be adjusted to resolve the issue?

Options:

A.

transcoder

B.

location

C.

route pattern

D.

translation pattern

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Questions 15

A company is using a H.323 gateway for PSTN calls from Cisco phones connected to a Cisco UCM cluster. There are reports of problems getting audio on these incoming calls from the PSTN. The phone rings, but nothing is heard and the call disconnects.

Which command troubleshoots this issue?

Options:

A.

debug voice h323 inout

B.

debug h245 pstn

C.

debug h245 asn1

D.

debug voice sip inout

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Questions 16

If all patterns below are configured in Cisco Unified Communications Manager which would be used when dialing the pattern “123”?

Options:

A.

12!

B.

12X (urgent priority set)

C.

1XX (urgent Priority Set)

D.

12[2-5]

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Questions 17

Refer to the exhibit.

300-815 Question 17

The Cisco UCM is configured with four route patterns. When phone A dials 1515, which pattern does Cisco UCM choose as the correct match?

Options:

A.

1XXX

B.

15

C.

!

D.

1[2-3]XX

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Questions 18

ABC company has decided to implement hunt groups to help distribute calls between members. In order to implement this, administrator must configure hunt list, hunt group, and line groups on Cisco UCM. Which distribution algorithms should the administrator implement?

Options:

A.

Top Down, Round Robin, Broadcast

B.

Top Down, Circular, Broadcast

C.

Top Down, Round Robin, Distribute

D.

Sequantial, Circular, Broadcast

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Questions 19

Refer to the exhibit.

300-815 Question 19

ILS has been configured between two hubs using this configuration. The hubs appear to register successfully, but ILS is not functioning as expected. Which configuration step is missing?

Options:

A.

A password has never been set for ILS.

B.

Use TLS Certificates must be selected.

C.

Trust certificates for ILS have not been installed on the clusters

D.

The Cluster IDs have not been set to unique values

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Questions 20

Phone A calls to phone B, but phone B has Call Forward All set to a PSTN number. The route list responsible for the phone B call to the PSTN has a standard local route group configured. Which route group must be used to send the call to the PSTN?

Options:

A.

route group defined in Standard Local Route Group section of Cisco UCM service parameters

B.

route group from the phone B device pool

C.

route group from the phone A device pool

D.

standard local route group defined in the route group configuration

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Questions 21

Refer to the exhibit.

300-815 Question 21

A Cisco Unified Border Element continues to send 180/183 with the required: 100rel header to Cisco UCM. and the call eventually disconnects How is the issue resolved?

Options:

A.

Enable ' SIP ReI1XX Options* and -Early Offer Support " on the SIP Profile Configuration Page in Cisco UCM.

B.

Enable *Early Offer support for voice and video calls " on the SIP Profile Configuration Page in Cisco UCM.

C.

Disable " SIP Rel1XX Options* and ' Early Offer Support* on the SIP Profile Configuration Page m Cisco UCM.

D.

Disable " Send send-receive SDP in mid-call INVITE* on the SIP Profile Configuration Page in Cisco UCM.

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Questions 22

Refer to the exhibit.

300-815 Question 22

A user dials 84969010 and observes that the call is not routed immediately. The administrator notices that after matching the fixed-length translation pattern, the call hits the \+! pattern and waits for interdigit timeout What should be configured to ensure that the call routes out immediately?

Options:

A.

Allow Device Override on the route pattern

B.

Route Next Hop By Calling Party Number on the translation pattern

C.

Do Not Wait For Interdigit Timeout On Subsequent Hops on the translation pattern

D.

Do Not Wait For Interdigit Timeout On Subsequent Hops on the route pattern

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Questions 23

300-815 Question 23

Refer to the exhibit. An administrator is trying to test outbound calls toward the ITSP but cannot complete the call and receives a SIP error. ITSP is consulted, and the issue is that the ANI that is being sent is not the DID provided 8005532447. Which configuration change sends the correct ANI on the INVITE sent to ITSP to fix the error?

Options:

A.

voice class sip-profiles 2

request INVITE sip-header To modify “sip:(.*)@” sip:8005532447@

B.

voice translation-rule 3

rule 1/.*/ /8005532447/

C.

voice class sip-profiles 1

request INVITE sip-header Diversion modify “sip:(.*)@” “sip:8005532447@”

D.

voice translation-rule 4

rule 1/^.*\(8005532447\)/ /\1/

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Questions 24

Exhibit.

300-815 Question 24

Refer to the exhibit. All calls from site A to site B are failing, and the issue has been identified as a media negotiation problem. Which configuration change resolves the issue?

Options:

A.

Disable G.722 on all devices at both sites.

B.

Enable Early Offer on the SIP trunk.

C.

Increase the bandwidth allowance between the RTP_Reg and SJ_Reg regions to 64 kbps.

D.

Create a new audio codec preference list with G.711 U-law 64k as the highest priority and apply it to RTP_Reg and SJ_Reg.

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Questions 25

Refer to the exhibit.

300-815 Question 25

An administrator just Implemented SIP trunking on their Cisco UCM and reports that calls using the SIP trunk are using Media Termination Point resources unnecessarily. Which action resolves the issue?

Options:

A.

Disable SIP Red XX Options.

B.

Change to a range that does not result in MTP.

C.

Change OTMF Signaling Method to " No Preference " .

D.

Change DTMF Signaling Method to " RFC 4733 " .

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Questions 26

An administrator is troubleshooting call failures on an H.323 gateway via the CLI. To see signaling for media and call setup, which debug must the Administrator turn on?

Options:

A.

debug H.323 messages

B.

debug H.224 asn1

C.

debug H.246 asn 1

D.

debug H.225 media

E.

debug H.323 asn 1

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Questions 27

Refer to the exhibit.

300-815 Question 27

An administrator just upgraded to Cisco UCM to version 14 and started SIP implementation with some new SIP trunks. During the testing, an error was reported when making a call. Which action resolves the issue?

Options:

A.

Change to a valid range.

B.

Change DTMF Signaling Method to " RFC 4833*.

C.

Make sure the destination number is configured correctly and the device is registered.

D.

Disable SIP Rel1XX Options.

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Questions 28

An engineer configures several calling party transformation patterns in Cisco UCM so that calling numbers are shown in the +E.164 format. The route patterns match seven-digit internal numbers and prefix the numbers with 1408. The SIP trunk matches 1408XXXXXXX and prefixes the number with +. When users make outbound calls, the calling number still shows as seven digits. What is the cause of this problem?

Options:

A.

The calling transformation on the SIP trunk is applied first, and then the calling transformation on the route patterns is applied.

B.

Calling transformations cannot be configured on both route patterns and a SIP trunk.

C.

The calling transformation on the route patterns is ignored because there is already a calling transformation on the SIP trunk.

D.

The engineer must also apply a calling transformation at the route group details level.

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Questions 29

What is a function of the metadata carried in SIP sessions between the recording client and the recording server?

Options:

A.

It forks RTP media to the recorder.

B.

It provides advanced capabilities, such as speech analytics.

C.

It sets up a new SIP session

D.

It identifies the participant change due to transfers during the call.

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Questions 30

Calls to a particular extension are not routing to voicemail. The user reaches the voicemail system from the handset by pressing the Messages button Which configuration parameter causes this problem?

Options:

A.

The voicemail pilot number for call forwarding is missing from the ephone-dn

B.

The voicemail pilot number is missing from the call handling on Cisco Unity Express

C.

The voicemail pilot number is missing from the telephony service configuration on Cisco UCME

D.

The voicemail pilot number for call forwarding is missing from the ephone

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Questions 31

Refer to the exhibit.

300-815 Question 31

An administrator is troubleshooting a problem in which some outbound calls from an internal network to the Internet telephony service provider are not getting connected, but some others connect successfully. The firewall team found that some call attempts on port 5060 came from an unrecognized IP that has not been defined in the firewall rule. What should the administrator configure in the Cisco Unified Border Element to fix this issue?

Options:

A.

use of port 5061 for SIP secure

B.

access list allowing the firewall IP

C.

ip prefix-list to filter the unwanted IP address

D.

bind signaling and media to the loopback interface

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Questions 32

Some users report having issues dialing some external numbers when traveling to other locations within the company. The company has five locations in five cities in one country and has an egress gateway in each location for TEHO. The configuration has no specific entry stating that the roaming users are using the local gateway, but calls are going out. How is a verification of the call routing in such a specific configuration performed to further identify the problem?

Options:

A.

device mobility

B.

standard local route group

C.

local route groups

D.

TEHO

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Questions 33

The Cisco Unified Communications Manager Dialed Number Analyzer allows analysis of calls from which two devices? (Choose two.)

Options:

A.

translation patterns

B.

device pools

C.

CTI ports

D.

CTI route points

E.

IP phones

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Questions 34

300-815 Question 34

Refer to the exhibit. An engineer deploys CAC to Cisco UCM. UptofiveG.711 calls must be supported on the WAN link between Site A and Site B. Users report that only four concurrent calls are possible between Site A and Site B. Why isn ' t the filth concurrent call successful?

Options:

A.

The G.729 audio codec is negotiated between Site A and B instead of the G.711 codec.

B.

The QoS configuration on the WAN link causes the fifth call to be dropped.

C.

Cisco UCM is using alternative links because of the Weight value.

D.

The Audio Bandwidth value is undersized and must be set to 400 kbps.

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Questions 35

An administrator is configuring Cisco UCM and the system to send *.webex.com traffic to a Cisco UCM Session Ma n agement Edition cluster. The administrator wants to limit which endpoints can reach *.webex.com. Which two items must the administrator configure for the SIP route pattern? (Choose two.)

Options:

A.

calling party transformation

B.

partition of the SIP route pattern

C.

connected party transformation

D.

called party URI transformation

E.

destination SIP trunk of the SIP route pattern

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Questions 36

An administrator is troubleshooting call failures on an H.323 gateway via the CLI. To see signaling for media and call setup, which two debugs must the administrator turn on? (Choose two).

Options:

A.

debug H.245 asn1

B.

debug H.323 message

C.

debug H.225 asn1

D.

debug H.225 media

E.

debug H.323 asn1

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Questions 37

Refer to the exhibit.

300-815 Question 37

Users report high delays when calling national numbers in the United States. An engineer analyzes the route patterns in Cisco UCM. What must be done to resolve this issue?

Options:

A.

Change the Cisco UCM routing logic to the longest-match algorithm.

B.

Select Allow Overlap Sending for the \+! route pattern.

C.

Mark the \+ [2-9JXX[2-91XXXXXX route pattern with urgent priority.

D.

Set the interdigit timeout service parameter value to 0.

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Questions 38

A company is using third-party phones on the internal network at a location behind a Cisco Unified Border Element. These phones seem to work with only two types of codecs, which are high-complexity codecs.

Which configuration ensures that calls are completed with the correct and working media each time?

Options:

A.

Configure a codec preference list on the Cisco Unified Border Element with the voice class codec command, set the codec preference list codecs, and add it to the dial peer.

B.

Configure a codec priority list on the Cisco Unified Border Element with the voice class codec command, set the codec priority list codecs, and add it to the dial-peer voice.

C.

Configure a codec preference list on the Cisco Unified Border Element with the VoIP class codec command, set the codec preference list codecs, and add it to the dial peer.

D.

Configure a codec preference list on the Cisco Unified Border Element with the voice class codec command, set the codec priority list codecs, and add it to the VoIP peer.

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Questions 39

A single site reports that when they dial select numbers, the call connects, but they do not get audio. The administrator finds that the calls are not routing out of the normal gateway but out of another site ' s gateway due to a TEHO configuration. What is the next step to diagnose and solve the issue?

Options:

A.

Verify that IP routing is correct between the gateway and the IP phone.

B.

Verify that the route pattern is not blocking calls to the destination number.

C.

Verify that the dial peer of the gateway has the correct destination pattern configured.

D.

Verify that the route pattern has the correct calling-party transformation mask

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Questions 40

A user in location X dials an extension at location Y. The call travels through a QoS-enabled WAN network, but the user experiences choppy or clipped audio. What is the cause of this issue?

Options:

A.

missing Call Admission Control

B.

codec mismatch

C.

ptime mismatch

D.

phone class of service issue

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Questions 41

When a third-party SIP Phone System is dialed inbound across a Cisco Unified Border Element, DTMF is failing. The third-party vendor accepts only out-of-band DTMF. Which configuration should be added to the outgoing dial peer to resolve this issue?

Options:

A.

dtmf-relay h245-signal

B.

dtmf-relay rtp-nte

C.

dtmf-relay cisco-rtp

D.

dtmf-relay sip-kpml

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Questions 42

An engineer Is configuring a Cisco Unified Border Element. The SIP ITSP requires a DNIS of 10-digits and DTMF tones sent in the RTP stream on the outbound leg of a call. Which code completes this configuration?

A)

300-815 Question 42

B)

300-815 Question 42

C)

300-815 Question 42

D)

300-815 Question 42

Options:

A.

Option A

B.

Option B

C.

Option C

D.

Option D

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Questions 43

When the services key is pressed Cisco Extension Mobility does not show up. What is the cause of the issue?

Options:

A.

The URL configured for Cisco Extension Mobility is not correct.

B.

Cisco Extension Mobility Service is not running.

C.

The phone is not subscribed to Cisco Extension Mobility Service.

D.

Cisco Extension Mobility is not enabled in the Phone Configuration Window (Device > Phone)

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Questions 44

An engineer is implementing survivability for a collaboration environment. The environment is utilizing a centralized Cisco UCM at the headquarters and Cisco IOS-XE gateways in the remote branches. Which action must the engineer take to enable both shared lines and B-ACD for SIP branch phones during a WAN outage?

Options:

A.

Configure mode srst under telephony-service configuration mode.

B.

Configure mode esrst under telephony-service configuration mode.

C.

Configure mode esrst under voice register global configuration mode.

D.

Configure mode esrst under voice service voip configuration mode.

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Questions 45

An administrator manages an environment with mobile users who commonly move between locations and take IP phones to remote sites. When mobile users are at one of the sites, users report issues with call quality, such as the audio cutting in and out. In addition, the affected area has a low bandwidth connection compared to other sites, so the administrator has put these devices in a region with an inter-region restriction of 8 kbps per call. Which feature must the administrator configure to mitigate the issue?

Options:

A.

Location Bandwidth Manager

B.

Extension Mobility

C.

Device Mobility

D.

Extension Mobility Cross Cluster

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Questions 46

300-815 Question 46

Refer to the exhibit. During a troubleshooting session, an engineer was presented with this call flow output diagram after a user reported issues with a call via the SIP ITSP to a counterpart. What is occurring with this call flow?

Options:

A.

The user is experiencing no-way audio after resuming the call from hold.

B.

The user is experiencing one-way audio after resuming the call from hold.

C.

The user call was terminated due to network issues.

D.

The user call proceeded normally after resuming from hold.

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Questions 47

An administrator configured Cisco Unified Mobility to block access to remote destinations for certain caller IDs. A user reports that a blocked caller was able to reach a remote destination. Which action resolves the issue?

Options:

A.

Configure Single Number Reach.

B.

Configure an access list.

C.

Configure a mobility identity.

D.

Configure Mobile Voice Access.

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Questions 48

What is first preference condition matched in a SIP-enabled incoming dial peer?

Options:

A.

incoming uri

B.

target carrier-id

C.

answer-address

D.

incoming called-number

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Questions 49

Refer to the exhibits.

300-815 Question 49

Regions have been configured for all major branches based on the available circuit bandwidth. Some calls from Region A endpoints to Region B endpoints are failing to connect. How is this issue resolved?

Options:

A.

Update the calling search space for affected endpoints to none.

B.

Add a media resource to transcode between available capabilities.

C.

Update all regions to 8 kbps maximum audio bitrate.

D.

Increase the number of available media termination points.

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Questions 50

Which elements does Cisco cloud mobility for collaboration include?

Options:

A.

Cisco Webex Collaboration Cloud Services

B.

Cisco Collaboration Cloud

C.

Cisco Collaboration Cloud and Cisco Webex Collaboration Cloud Services

D.

Cisco Collaboration Cloud and Cisco Mobile and Remote Access Collaboration Cloud Services

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Questions 51

An administrator is implementing a new dial-plan on Cisco Unified Border Element. The administrator must ensure that incoming dial-peers are matched based on the IP address from where the incoming request originates. Which dial-peer configuration should be applied to accomplish this requirement?

Options:

A.

dial-peer voice 1 voip

incoming url via

B.

dial-peer voice 1 voip

incoming url request

C.

dial-peer voice 1 voip

incoming called-number

D.

dial-peer voice 1 voip

incoming url to

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Questions 52

A network engineer designs a new dial plan and wants to block a certain range of numbers (8135100 through 8135105). What is the most specific route pattern that can be configured to block only the numbers in this range?

Options:

A.

813510[012345]

B.

813510[12345]

C.

813510[^0-5]

D.

81XXXXX

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Questions 53

Why would RTP traffic that is sent from the originating endpoint fail to be received on the far endpoint?

Options:

A.

The far end connection data (c=) in the SDP was overwritten by deep packet inspection in the call signaling path.

B.

Cisco Unified Communications Manager invoked media termination point resources.

C.

The RTP traffic is arriving beyond the jitter buffer on the receiving end.

D.

A firewall in the media path is blocking TCP ports 16384-32768.

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Questions 54

Which section under the Real-Time Monitoring Tool allows for reviewing the call flow and signaling for a SIP call in real time?

Options:

A.

Analysis Manager > Inventory > Trace File Repositories

B.

System > Tools > Trace and Log Central

C.

Voice/Video > Session Trace Log View > Real Time Data

D.

Voice/Video > Session Trace Log View > Open From Local Disk

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Questions 55

An engineer needs to configure a translation pattern in Cisco UCM to match dialed numbers that start with 9011 so that the calls can be routed with a single route pattern of VH If the engineer uses translation pattern 9011.! which two called party transformation settings must be configured in Cisco UCM? (Choose two.)

Options:

A.

set Prefix Digits (Outgoing Calls) to +

B.

set Called Party Transform Mask to +!

C.

set Discard Digits to Predot

D.

set Called Party Numbering Plan to International

E.

set Called Party Number Type to International

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Questions 56

An administrator sees the voice register pool 1 command in a Cisco UCME configuration. Which configuration is occurring in this section?

Options:

A.

configuration for a pool of SIP phones (similar to device pool on Cisco UCM)

B.

configuration for a single SIP phone

C.

configuration for SIP registrar service

D.

configuration items common for all SIP phones

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Questions 57

An administrator has configured two route patterns, 9.911 and 9.[2-9]XXXXXX. When a user dials 9911. Cisco UCM waits for the T302 timer before routing the call. How will the administrator force interdigit timeout and route the call as soon as the user has finished dialing 9911, without waiting for the T302 timer to expire?

Options:

A.

decrease the T302 timer in Service Parameters from the default value

B.

enable Urgent Priority on the 9.[2-9]XXXXXX pattern

C.

enable Urgent Priority on the 9.911 pattern

D.

enable Device Override on both route patterns

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Questions 58

A company is using Cisco Jabber on-premises to make B2B calls on video. The calls are using Cisco Expressway-C and Expressway-E and have been configured in Cisco UCM to be able to call any URI on the internet. The Jabber client also has voice enabled and must be able to call local, regional, and international numbers. Where must Cisco UCM be configured to meet this requirement for URI dialing?

Options:

A.

Enter " !# " in the route pattern section tied to a route group and list.

B.

Enter " * " in the SIP route pattern.

C.

Enter " !# " in the SIP route pattern.

D.

Enter " .* " in the route pattern section tied to a route group and list.

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Questions 59

Refer to the exhibit.

300-815 Question 59

In an active SIP call between phone user A and phone user B, phone A initiates a call transfer to phone user C. Which two scenarios are correct? (Choose two.)

Options:

A.

Phone_A sends a SIP-REFER message to the Cisco Unified Communications Manager with Phone_C information in the Refer-To section.

B.

Phone_B sends a SIP-REFER message to the Cisco Unified CM with Phone_C information in the Refer-To section.

C.

As soon as Phone_A presses the Transfer button for the first time, Phone_B hears the MOH and the MOH audio is chosen from Phone_B User Hold MOH Audio Source settings.

D.

As soon as Phone_A presses the Transfer button for the first time, Phone_B hears the music on hold and the MOH audio is chosen from Phone_A Network Hold MOH Audio Source settings.

E.

As soon as Phone_A presses the Transfer button for the first time, Phone_B hears the MOH and the MOH audio is chosen from Phone_A User Hold MOH Audio Source settings.

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Questions 60

Refer to the exhibit.

300-815 Question 60

An administrator is troubleshooting why users are not hearing audio when dialing long distance numbers across their Cisco Unified Border Element. The customer’s carrier has a requirement that dialing long distance requires an access code to be entered. Looking at the exhibit, what two actions can be taken to correct signaling? (Choose two.)

Options:

A.

Enable PRACK.

B.

Enable Early Offer on the Cisco Unified Border Element.

C.

Enable the supplementary-service media-renegotiate command.

D.

Enable Media Flow Around

E.

Enable Mid-Call Signaling Consumption.

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Questions 61

Which configuration must an administrator perform to display Translation Pattern operations in Cisco Unified Communications Manager SDL traces?

Options:

A.

Enable the Detailed Call Analysis option under Enterprise Parameters for Unified CM.

B.

Set up the Digit Analysis Complexity in Service Parameters for Cisco Unified CM to TranslationAndAlternatePatternAnalysis.

C.

Check the Translation Patterns Analysis check box in Micro Traces on the Cisco Unified CM Serviceability page.

D.

By default, the Translation Patterns operations are printed in SDL traces, so no additional configuration is necessary.

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Questions 62

A voice administrator enabled secure media between multiple Cisco UCM clusters that are connected via a SIP trunk.

Which configuration parameter in the SIP trunk must be enabled to allow encrypted media traffic to traverse the SIP trunk?

Options:

A.

Uncheck the RTP Allowed check box in the SIP Trunk configuration

B.

Check the RTP Allowed check box in the SIP Trunk configuration

C.

Check the SRTP Allowed check box in the SIP Trunk configuration

D.

Check the Allow iX Application Media check box in the SIP Profile

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Questions 63

An engineer needs to deploy Cisco Extension Mobility. The engineer already activated the required services, configured the Extension Mobility Phone Service, and created an Extension Mobility device profile that is associated with end users. Which two additional steps are required to complete the configuration? (Choose two.)

Options:

A.

Subscribe IP phones to the Extension Mobility Phone Service.

B.

Consolidate Cisco UCM certificates by using Bulk Certificate Management.

C.

Subscribe directory numbers to the Extension Mobility Phone Service.

D.

Subscribe the device profile to the Extension Mobility Phone Service.

E.

Define the default Device Template on IP phones when users log out.

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Questions 64

An engineer has two cisco UCM Clusters and wants to integrate them using ILS with TLS certificates. Cluster A (pub and 1 subscriber) will be the hub, and Cluster B (pub and 1 subscriber) will be the spoke. Both Clusters have self-signed certificates. The engineer has exchanged Publisher A and subscriber B Tomcat certificates, but the connection fails. What is the cause of the failure?

Options:

A.

The password is incorrect.

B.

Cluster IDs are not unique.

C.

The tomcat certificate from Cluster B must be the publisher.

D.

The engineer needs to exchange the CallManager certificate.

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Questions 65

300-815 Question 65

Refer to the exhibit. An administrator configured a new SIP trunk between Cisco UCM and Cisco Unified Border El e ment. Calls that use this new trunk are failing, and debugs from the Cisco Unified Border Element do not show any signaling received from Cisco UCM. The administrator verified that OPTIONS is enabled on the SIP profile configured on the trunk on Cisco UCM. The route pattern matched for this call is correctly configured to send calls to the Cisco Unified Border Element using this new trunk. What is the cause of this failure?

Options:

A.

The calling search space on the calling line/device does not contain the partition of the route pattern

B.

The called party number is not complete because it contains only 7 digits

C.

A connectivity issue between Cisco UCM and Cisco Unified Border Element prevents Cisco UCM from receiving a response to the OPTIONS it is sending to Cisco Unified Border Element

D.

There are multiple matches for the dialed number, and Cisco UCM is trying to send the call to SEP2834A2824611 rather than Cisco Unified Border Element

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Questions 66

Which call pickup feature allows users to pick up incoming calls in a group that is associated with their own group?

Options:

A.

Other Group Pickup

B.

BLF Call Pickup

C.

Group Call Pickup

D.

Directed Call Pickup

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Questions 67

Refer to the exhibit.

300-815 Question 67

Calls incoming from the provider are not working through newly set up Cisco Unified Border Element. Provider engineers get the 404 Not Found SIP message. Incoming calls are coming from the provider with called number “222333444” and Cisco Unified Communications Manager is expecting the called number to be delivered as “444333222”. The administrator already verified that the IP address of the Cisco Unified CM is set up correctly and there are no dial peers configured other than those shown in the exhibit. Which action must the administrator take to fix the issue?

Options:

A.

Change the destination-pattern on the outgoing dial peer to match “444333222”.

B.

Set up translation-profile on the incoming dial peer to match incoming traffic.

C.

Create specific matching for “222333444” on the incoming dial peer.

D.

Fix the voice translation-rule to match specifically number “222333444” and change it to “444333222”.

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Questions 68

The company Cisco UCM cluster has two different gateways for off-net calls. The current configuration uses 9 as a prefix to get to the main gateway. The secondary gateway is for any calls that start with 9713, but this is not yet configured. The admin does not want to add more route patterns other than the current 9 prefix for the gateway to the Cisco UCM. How must the Cisco UCM be configured to meet the requirements?

Options:

A.

Configure two CSSs. then add a translation pattern with the secondary CSS to send calls with 9713 to the secondary gateway.

B.

Configure two partitions and two CSSs, then add a translation pattern with the secondary CSS to send all calls with 9713 to the secondary gateway.

C.

Create a Standard Route Group to dynamically route calls with prefix 9713 to the secondary gateway.

D.

Configure a Route Group and a Route List to send calls with prefix 9713 to the secondary gateway.

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Questions 69

300-815 Question 69

Refer to the exhibit. Users report that outgoing calls do not work on the new SIP trunk for outgoing calls. The solution consists of a Cisco UCM Cluster linked to a Cisco Unified Border Element where the SIP trunk is terminated. The provider required 10 digits. The logs show a line going toward the Cisco Unified Border Element. Which code snippet must be added to the configuration to meet the requirement?

Options:

A.

request Invite sip-header modify " & lt;sip:1(...)@ " " & lt;sip:9135551\1@ " under the SIP translation profile configuration

B.

sip-header modify " & lt;sip:1(...)@ " " & lt;sip:9135551\1@ " under the voice translation profile configuration

C.

request Invite sip-header Diversion modify " & lt;sip:1(...)@ " " & lt;sip:9135551\1@ " under the SIP profile configuration

D.

request Invite sip-header modify " & lt;sip:1(...)@ " " & lt;sip:9135551\1@ " under the voice translation profile configuration

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Questions 70

The SIP session refresh timer allows the RTP session to stay active during an active call. The Cisco UCM sends either SIP-INVITE or SIP-UPDATE messages in a regular interval of time throughout the active duration of the call. During a troubleshooting session, the engineer finds that the Cisco UCM is sending SIP-UPDATE as the SIP session refresher, and the engineer would like to use SIP-INVITE as the session refresher. What configuration should be made in the Cisco UCM to achieve this?

Options:

A.

Enable SIP ReMXX Options on the SIP profile.

B.

Enable Send send-receive SDP in mid-call INVITE on the SIP profile.

C.

Change Session Refresh Method on the SIP profile to INVITE.

D.

Increase Retry INVITE to 20 seconds on the SIP profile.

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Questions 71

Which IOS command creates a SIP-enabled dial peer?

Options:

A.

voice dial-peer 20 sip

B.

dial-peer voice 20 voip

C.

dial-peer voice 20 pots

D.

dial peer voice 20 sip

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Questions 72

End users at a new site report being unable to hear the remote party when calling or being called by users at headquarters. Calls to and from the PSTN work as expected. To investigate the SIP signaling to troubleshoot the problem, which field can provide a hint for troubleshooting?

Options:

A.

Contact: header of the 200 OK response

B.

Allow: header if the 200 OK response

C.

o= line of SDP content

D.

c= line of SDP content

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Exam Code: 300-815
Exam Name: Implementing Cisco Advanced Call Control and Mobility Services (CLACCM)
Last Update: Jun 5, 2026
Questions: 242

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