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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

How many maximum hops can an ILS update traverse?

Options:

A.

3

B.

6

C.

9

D.

12

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Questions 5

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.225 asn1

C.

debug H.246 asn 1

D.

debug H.225 media

E.

debug H.323 asn 1

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Questions 6

A new deployment is using MVA for a specific user on the sales team, but the user is having issues when dialing DTMF. Which DTMF method must be configured in resolve the issue?

Options:

A.

gateway

B.

out-of-band

C.

channel

D.

in-band

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Questions 7

Refer to the exhibit.

300-815 Question 7

Users report that when they dial the emergency number 9911 from any internal phone, it takes a long time to connect with the emergency operator. Which action resolves this issue?

Options:

A.

Adjust the service parameter T302 timet to the desired value.

B.

Adjust the service parameter T204 timer to the desired value.

C.

Check the Urgent Priority check box under 9.911 pattern.

D.

Point the emergency pattern directly to the PSTN gateway.

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Questions 8

Which description of RTP timestamps or sequence numbers is true?

Options:

A.

The sequence number is used to detect losses.

B.

Timestamps increase by the time “carrying” by a packet.

C.

Sequence numbers increase by four for each RTP packet transmitted.

D.

The timestamp is used to place the incoming audio and video packets in the correct timing order (playout

delay compensation).

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Questions 9

What are two configureation features of the Client matter code setting in the route pattern configuration? (Choose two.)

Options:

A.

The client Matter Code feature supports overlap sending since the Cisco UCM can determine when to prompt the user for the code.

B.

Selecting the Allow Overlap Sending setting disables the Require Client Matter Code setting.

C.

The Client Matter Code feature provides the option to configure Authorization Level susch as in the Forced Authorization Code.

D.

Selecting the Allow Overlap Sending setting allows a user to select the Require Client Matter Code setting.

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Questions 10

Refer to the exhibits.

300-815 Question 10

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 11

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 12

A company was looking at the IT charges and saw many long-distance and international calls primarily to sites in North America and around the world. The administrator wants to optimize the PSTN expense. Which dial plan configuration reduces PSTN connectivity charges by using the IP network to bring the egress point to the PSTN as close as possible to the called number?

Options:

A.

translation patterns

B.

tail end hop off

C.

client matter codes

D.

dial rules

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Questions 13

An organization needs to ensure that the Cisco UCM can provide a resume softkey to users on their desk phones when they disconnect an anchored call on their mobile phone. Which solution must be used to accomplish these goals?

Options:

A.

Set the "Retain Media on Disconnect with PI for Active Call" service parameter to False.

B.

Set the "Retain Media on Disconnect with PI for Active Cal" service parameter to True

C.

Configure a handoff number.

D.

Add Other Dual-Mode Device.

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Questions 14

Where is the dtmf-relay command configured on Cisco Unified Border Element?

Options:

A.

in the voice-class VoIP configuration

B.

in the VoIP dial peer

C.

in global SIP configuration

D.

in the VoIP or POTS dial peers

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Questions 15

Refer to the exhibit.

300-815 Question 15

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 16

Signal number reach call phone that not answered are leaving voicemails on the cell phone rather the corporate mailbox. Which two options will resolve this issue? (Choose two.)

Options:

A.

Check the Enable Extend and Connect checkbox

B.

Check the Enable Unified Mobility features checkbox

C.

Decrease the T302 timer

D.

Decrease the T301 timer Decrease the Answer Too Late timer

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Questions 17

300-815 Question 17

Refer to the exhibit. An administrator troubleshoots an issue where each user's 4-digit directory number is presented as the calling number when dialing out to the PSTN. This 4-digit calling number prevents the external user from calling the internal user back. Which two actions must the administrator perform to resolve the issue? (Choose two.)

Options:

A.

Update the SIP profile on the SIP trunk by checking “enable external presentation name and number”

B.

Change the calling line ID presentation on the SIP trunk to allow.

C.

Create a calling party transformation patter that matches the desired directory number and add a mask to modify the calling number

D.

Create a called party transformation pattern that matches the desired directory number and add a mask to modify the calling number

E.

Add a calling party transformation CSS to the SIP trunk.

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Questions 18

What are the elements for Device Mobility configuration?

Options:

A.

physical location, device pool, and Device Mobility group

B.

device pool, Device Mobility group, and region

C.

physical location. Device Mobility group, and region

D.

device pool, Device Mobility group, and Cisco IP phone

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Questions 19

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 20

An administrator is configuring Meet-me conferencing in a Cisco UCM deployment and has created the Meet-me number and ensured that it is in a partition accessible by all devices. Which two additional steps must the administrator perform? (Choose two.)

Update the softkey template on all phones to ensure that they contain the Meet-me softkey.

Options:

A.

Ensure that conferencing-initiating devices are using a media resource group list that contains at least one Cisco UCM conference bridge.

B.

Enable Meet-me conferencing In enterprise parameters.

C.

Ensure that all devices have G.729 enabled.

D.

Disable Early Media on the SIP profile of all devices that will use Meet-me conferencing.

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Questions 21

Which built-in bridge configuration option can be set on the individual IP phones to ensure that the cluster-wide service parameter is used?

Options:

A.

automatic

B.

default

C.

on

D.

off

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Questions 22

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 23

An engineer is configuring Cisco UCM lo forward parked calls back to the user who parked the call if it is not retrieved after a specified time interval. Which action must be taken to accomplish this task?

Options:

A.

Configure device pools.

B.

Configure service parameters

C.

Configure enterprise softkeys.

D.

Configure class of control.

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Questions 24

After configuring a Cisco CallManager Express with Cisco Unity Express, inbound calls from the PSTN SIP trunk receive a ring tone for 20 seconds and then a busy signal instead of voicemail. Which configuration fixes this problem?

Options:

A.

Router(config)# voice service voip

Router(conf-voi-serv)#allow-connections h323 to h323

B.

Router(config)#dial-peer voice 2 voip

Router(config-dial-peer)#no vad

C.

Router(config)# voice service voip

Router(conf-voi-serv)#allow-connections voice-mail mod

D.

Router(config)# voice service voip

Router(conf-voi-serv)#no supplementary-service sip moved-temporarily

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Questions 25

Refer to the exhibit.

300-815 Question 25

An engineer is troubleshooting an issue with the caller not hearing a PSTN announcement before the SIP call has completed setup. How must the engineer resolve this issue using the reliable provisional response of the SIP?

Options:

A.

voice service voip sip send 180 sdp

B.

voice service voip sip rehxx require 100rel

C.

sip-ua

disable-early-media 180

D.

voice service voip sip no reMxx

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Questions 26

Refer to the exhibit.

300-815 Question 26

A call mode through the Cisco Unified Border Element to pilot 2000 is foiling. What is causing the call to foil?

Options:

A.

No codecs are configured on the dial peers

B.

The Cisco Unified Border Element is not receiving a response to its OPTION keepahves.

C.

The destination pattern is incorrect for the dialed number.

D.

VAD was not disabled on the outgoing dial poor.

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Questions 27

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 28

300-815 Question 28

Refer to the exhibit. An engineer is trying to set up a new deployment using the SIP provider with TCP for signaling. After troubleshooting, the customer notices that the matching incoming dial peer and outgoing dial peer did not generate the INVITE to the SIP provider. Why is the call failing?

Options:

A.

The ITSP is sending FIN.

B.

The ITSP is not sending the ACK.

C.

The ITSP is sending CLOSE state.

D.

The ITSP is not answering the SYN.

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Questions 29

Refer to the exhibit.

300-815 Question 29

A user reports that when they call a specific phone number, no one answers the call, but when they call from a mobile phone, the call is answered. The engineer troubleshooting the issue is expecting the far-end gateway to cut through audio on the 183 Session Progress SIP message. Which SIP Profile configuration element is necessary for the Cisco Unified Communications Manager to send acknowledgement of provisional responses?

Options:

A.

Allow Passthrough of Configured Line Device Caller Information must be enabled.

B.

Accept Audio Codec Preferences in Received Offer must be set to On.

C.

On the SIP Profile, the configuration parameter SIP Rel1XX Options must be set to Send PRACK for all 1xx Messages.

D.

Early Offer for G Clear Calls must be enabled.

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Questions 30

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 thatthe 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 31

When configuring hunt groups, where does the administrator add the individual directory numbers that should be part of the group?

Options:

A.

route group

B.

line group

C.

hunt list

D.

hunt pilot

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Questions 32

A Cisco UCM-based solution uses URI-based calling for Cisco Jabber clients. The solution has several clusters in different locations. The solution being configured must have a secure methodo* exchanging addresses between the clusters. Which action ensures that a configuration complies with the requirements?

Options:

A.

Assign a self-signed certificate to the ILS service trunk.

B.

Apply a secure security profile to the trunk between the clusters.

C.

Select the TLS option on the SIP trunk between the clusters.

D.

Exchange Tomcat certificates

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Questions 33

In Cisco UCM, which tool is used to check SIP traces?

Options:

A.

MTP

B.

CCSIP

C.

RTMT

D.

OS Administration Page

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Questions 34

An administrator troubleshoots call failure in a new deployment and finds that the SIP INVITE messages sent to the service provider contain a diversion header with the user's 4-digit directory number. These 4-digit directory numbers range from 1000 to 9999. The service provider is rejecting the calls because it requires that the diversion header contain 10 digits. Which command on the Cisco Unified Border Element resolves this issue for all users?

A)

300-815 Question 34

B)

300-815 Question 34

C)

300-815 Question 34

D)

300-815 Question 34

Options:

A.

Option A

B.

Option B

C.

Option C

D.

Option D

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Questions 35

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 36

An administrator is configuring a new deployment using Cisco Unified CME. The SCCP phones register without any issues, but SIP phones are not registering. Assume that all other configuration is valid. Which code allows SIP phones to register to Cisco UCME?

Options:

A.

voice service voip

allow-connections sip to h323

B.

voice service voip

sip

bind media source-interface Vlan100

C.

voice service voip

sip

bind control source-interface Vlan100

D.

voice service voip

sip

registrar server expires max 600 min 60

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Questions 37

Users are reporting that several inter-site calls are failing, and the message "not enough bandwidth" is showing on the display. Voice traffic between locations goes through corporate WAN. and Call Admission Control is enabled to limit the number of calls between sites. How is the issue solved without increasing bandwidth utilization on the WAN links?

Options:

A.

Disable Call Admission Control and let the calls use the amount of bandwidth they require.

B.

Configure Call Queuing so that the user waits until there is bandwidth available

C.

Configure AAR to reroute calls that are denied by Call Admission Control through the PSTN.

D.

Reroute all calls through the PSTN and avoid using WAN.

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Questions 38

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 39

An administrator configured Extension Mobility on Cisco UCM. Users report that logins are successful, but the phones do not have an Extension Mobility option after logging in. The administrator verified that Extension Mobility is enabled on the devices and that the log-out profile is valid. Which action must the administrator take to resolve the issue?

Options:

A.

Subscribe all device profiles to the Extension Mobility phone service.

B.

Delete the identity trust list file from the phone(s).

C.

Change the Extension Mobility URL from HHTP to HTTPS.

D.

Restart the Cisco Extension Mobility and Cisco Extension Mobility application services.

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Questions 40

An engineer must configure call queuing under a Hunt Pilot. After the engineer receives the audio file that will be played to callers during queuing, which two steps should be taken to complete the configuration? (Choose two.)

Options:

A.

Assign the uploaded audio file to the hunting Line Group member's "User Hold MOH Audio Source

B.

Assign the uploaded audio file to the hunting Line Group member's "Network Hold MOH Audio Source".

C.

Upload the audio file in "TFTP File Management” via OS Administration GUI

D.

Assign the uploaded audio file to "Network Hold MOH Source & Announcements" under Hunt Pilot's Queuing section.

E.

Upload the audio file in "MOH Audio File Management" via CM Administration GUI

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Questions 41

300-815 Question 41

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 42

Refer to the exhibit.

300-815 Question 42

An administrator is troubleshooting a situation where a call placed from a phone registered to Cisco Unified Communications Manager does not complete. The administrator wants to use the Dialed Number Analyzer on Cisco Unified CM to check which translation pattern the call is matching. However, when logging in to Cisco Unified Serviceability there is no option for Dialed Number Analyzer under the tool menu. Which two steps must be performed to resolve this issue? (Choose two.)

Options:

A.

Restart the subscriber

B.

Activate the Cisco Extended Functions service.

C.

Activate the Cisco CallManager service.

D.

Activate the Cisco Dialed Number Analyzer service.

E.

Activate the Cisco Dialed Number Analyzer Server service.

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Questions 43

An engineer needs to deploy the local route group feature to a Cisco UCM server by using the standard local route group. For each device pool, the engineer needs to specify a different PSTN access for emergency calls. What must the engineer do?

Options:

A.

Add a new standard local route group. Select a route group as the new standard local route group in Device Pools. Create a route pattern of 911 that points to a route list that contains the new standard local route group.

B.

Create a new local route group name. In Device Pools, select a route group for that new local route group. Create a route pattern of 911 that points to a route list that contains the new local route group.

C.

Set a new local route group name. Select a route group as the new local route group in Device Pools. Create a route pattern of 911 that points to the new local route group.

D.

Add a new standard local route group. Select a route group as the new standard local route group in Device Pools. Create a route pattern of 911 that points to the new standard local route group.

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Questions 44

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 45

300-815 Question 45

Refer to the exhibit. When setting up a new connection to Cisco UCM, the engineer must use out-of-band DTMF. Which configuration meets this requirement?

Options:

A.

dtmf-relay h245-alphanumeric

B.

dtmf-relay sip-kpml

C.

dtmf-relay cisco-rtp

D.

dtmf-relay rtp-nte

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Questions 46

Which two configuration parameters are prerequisites to set Native Call Queuing on Cisco Unified Communications Manager? (Choose two.)

Options:

A.

Cisco IP Voice Media Streaming Service must be activated on at least one node in the cluster.

B.

A unicast music on hold audio source must be configured.

C.

Cisco RIS data collector service must be running on the same server as the Cisco CallManager service.

D.

The maximum number of callers allowed in queue must be 10.

E.

The phone button template must have the Queue Status Softkey configured.

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Questions 47

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 48

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 49

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 50

An administrator is configuring Cisco UCM and the system to send *.webex.com traffic to a Cisco UCM Session Management 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 51

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 52

300-815 Question 52

Refer to the exhibit. A collaboration engineer is troubleshooting an issue where a user of a Cisco UCM IP phone reports failed calls when trying to dial out to the PSTN. Which action resolves the issue?

Options:

A.

Assign a calling search space to the line or the device that has access to the route pattern.

B.

Deselect “Block this pattern” on the “Route Option” setting of the route pattern.

C.

Select the “Urgent Priority” setting on the route pattern.

D.

Instruct the user to not dial a “1” before their local area code.

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Questions 53

Users report silence on the line when they try to connect to external voice numbers. The company is using Cisco 8865 phones with Cisco UCM connected to a gateway with two T1 circuits. The MGCP server seems to be responding. Which action must the engineer take to begin to investigate the media for this type of issue?

Options:

A.

Run RTMP on the Cisco router.

B.

Verify that the MGCP service is running under serviceability in Cisco UCM.

C.

Run the debug mgcp media command on the Cisco router.

D.

Run the debug H.323 gateway command on the Cisco router.

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Questions 54

Refer to the exhibit.

300-815 Question 54

A user cannot call number 730. which Is an abbreviated number for 45730. The engineer recently configured a translation pattern In Cisco UCM. The newly configured translation pattern of 7XX prefixes the digits 45. The engineer uses DNA to troubleshoot the issue. What is the reason for this problem?

Options:

A.

The abbreviated partition is missing from the calling search space.

B.

The translation pattern is blocking the call.

C.

The translation pattern Is misconfigured.

D.

Directory number 45730 is unknown to Cisco UCM.

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Questions 55

300-815 Question 55

Refer to the exhibit. A standard local route group is configured for long-distance calls. Calls from building A succeed, but calls from building B fail. On the system. Each building has is own device pool. The DNA tool is used to test the configuration. How is this issue resolved?

Options:

A.

Change the partition of the route pattern

B.

Add a sip trunk inside route group Standard Local Route Group.

C.

Modify the route pattern to add a prefix of 91

D.

Add a local route group on the device pool configuration.

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Questions 56

300-815 Question 56

Refer to the exhibit. An engineer is troubleshooting an issue where inbound Calls are failing after they transferred. The provider reports that update is not supported, and this is causing the calls to fail. Which command should resolve this issue?

Options:

A.

no midcall-signaling passthru

B.

no update-callerId

C.

no contact-passig

D.

rel1xx require “100rel”

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Questions 57

A customer has multisite deployments with a globalized dial plan. The customer wants to route PSTN calls via the gateway assigned to each site. Which two actions will fulfill the requirement? (Choose two.)

Options:

A.

Create one route group for each site and one global route list for PSTN calls that point to the local route group.

B.

Create a route group which has all the gateways and associate it to the device pool of every site.

C.

Create one global route list for PSTN calls that points to one global PSTN route group.

D.

Create a hunt group and assign it to each side route pattern

E.

Assign one route group as a local route group in the device pool of the corresponding site.

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Questions 58

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 59

A new Cisco UCM deployment is configured, and now the customer requires a set of 10 DNSs to park the calls starting with 8. The users must use the softkey to park the call. Which configuration must be performed to achieve this requirement?

A)

300-815 Question 59

B)

300-815 Question 59

C)

300-815 Question 59

D)

300-815 Question 59

Options:

A.

Option A

B.

Option B

C.

Option C

D.

Option D

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Questions 60

An SRST site must use pickup functionality on the Chicago remote site to allow the users to take incoming calls. The pickup command is configured under the call-manager-fallback section on the SRST router What are two results of that configuration? (Choose two.)

Options:

A.

The PickUp soft key is on all SRST phones.

B.

The GPickUp soft key is on all SRST phones.

C.

Calls that come into one directory number can be picked up from another directory number.

D.

Calls that come into a Cisco UCM registered phone can be picked up by a SRST phone.

E.

Calls that ring an unassigned directory number are forwarded to the auto-attendant.

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Questions 61

Refer to the exhibit.

300-815 Question 61

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 62

An administrator is asked to configure egress call routing by applying globalization and localization on Cisco UCM. How should this be accomplished?

Options:

A.

Localize the calling and called numbers to PSTN format and globalize the calling and called numbers in the gateway.

B.

Globalize the calling and called numbers to PSTN format and localize the calling number in the gateway.

C.

Localize the calling and called numbers to E. 164 format and globalize the called number in the gateway.

D.

Globalize the calling and called numbers to E. 164 format and localize the called number in the gateway.

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Questions 63

300-815 Question 63

Refer to the exhibit. An administrator configured Device Mobility but is receiving reports that local calls are failing when a user takes their device from the RTP location to the Charlotte location. The administrator confirmed that the correct subnet is configured under the Device Mobility info page. In addition, the roaming device pool has the correct Device Mobility calling search space selected. Which configuration change resolves the issue?

Options:

A.

Join across lines is not supported with Device Mobility and must be disabled.

B.

The network locale must be changed from none to the Charlotte locale.

C.

The physical location must be updated from none to the appropriate location.

D.

The device must be added to the Device Mobility group.

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Questions 64

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 65

Refer to the exhibit.

300-815 Question 65

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 66

An administrator deployed a third-party H.323 gateway in a voice environment, but users report call failures when using features like call hold or call transfer. What are two reasons that these features fail? (Choose two.)

Options:

A.

The CSS of the transfer initiating line does not contain the partition of the supplementary feature extension (DirectTransfer or MoH Number).

B.

The MTP that is configured for use within the H.323 gateway configuration is configured as a trusted source, but the third-party gateway does not trust the signing root CA certificate of the MTP certificate.

C.

The MTP does not support the negotiated codec, and media renegotiating during the call is not supported.

D.

The Media Resource Group List of the H.323 gateway contains only transcoders and conference bridges but no MTP.

E.

The third-party gateway does not support supplementary features, so Media Termination Point (MTP) must be inserted.

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Exam Code: 300-815
Exam Name: Implementing Cisco Advanced Call Control and Mobility Services (CLACCM)
Last Update: Jun 7, 2025
Questions: 221

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