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300-815 Exam Prep Free

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  • 300-815 Exam Prep Free – 50 Practice Questions to Get You Ready for Exam Day
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300-815 Exam Prep Free – 50 Practice Questions to Get You Ready for Exam Day

Getting ready for the 300-815 certification? Our 300-815 Exam Prep Free resource includes 50 exam-style questions designed to help you practice effectively and feel confident on test day

Effective 300-815 exam prep free is the key to success. With our free practice questions, you can:

  • Get familiar with exam format and question style
  • Identify which topics you’ve mastered—and which need more review
  • Boost your confidence and reduce exam anxiety

Below, you will find 50 realistic 300-815 Exam Prep Free questions that cover key exam topics. These questions are designed to reflect the structure and challenge level of the actual exam, making them perfect for your study routine.

Question 1

An engineer must implement call restriction to toll-free numbers using a class of restriction in a branch Cisco UCME. In which two places is the corlist incoming or cor incoming command configured? (Choose two.)

A. “voice register pool ” configuration mode

B. “ephone-dn ” configuration mode

C. “dial-peer cor custom ” configuration mode

D. “voice register global ” configuration mode

E. “telephony-service ” configuration mode

 


Suggested Answer: AB

 

Question 2

Cisco UCM has 100,000 entries in the database learned through the ILS Service. Parameter ILS Max Number of Learned Objects in Database value is set to 100,000. What will happen to learned data when the service parameter value is reduced to 50,000?

A. Cisco UCM does not write additional ILS learned objects to the database and will delete the last 50,000 entries learned to keep it to the service parameter value.

B. Cisco UCM does not write additional ILS learned objects to the database and keeps the existing database entries.

C. Cisco UCM will overwrite an entry for newly learned data and keep the parameter value at 100,000.

D. Cisco UCM does not write additional ILS learned objects to the database and will delete the first 50,000 entries learned to keep it to the service parameter value.

 


Suggested Answer: B

 

Question 3

Image
Refer to the exhibit. 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.)

A. Enanle 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.

 


Suggested Answer: AB

 

Question 4

A company is experiencing an issue where calls from the Cisco UCM via Cisco Unified Border Element to the ITSP fail with a “500 server internal error” right after a 183 session in progress is sent back from the ITSP. What must be enabled to resolve the issue?

A. delayed offer on Cisco UCM

B. early offer at the CUBE

C. delayed offer on CUBE

D. early offer at the ITSP

 


Suggested Answer: B

 

Question 5

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

A. automatic

B. default

C. on

D. off

 


Suggested Answer: B

 

Question 6

Image
Refer to the exhibit. Within the North American Numbering Plan, gateways located in Ottawa, Canada and marked as "YOW" are assigned to the Calling Party
Transformation CSS NANP_CgPTP, which contains partition NANP_calling_xforms. What is the calling-party number and the numbering type if the calling user
+1613-555-1234 dials the number?

A. calling number 613-555-1234 and numbering type “subscriber”

B. calling number 011-1-613-555-1234 and numbering type “subscriber”

C. calling number 011613-555-1234 and numbering type “international”

D. calling number 613-555-1234 and numbering type “national”

 


Suggested Answer: D

 

Question 7

Image
Refer to the exhibit. A collaboration engineer is troubleshooting a Cisco UCM issue where users report that calls placed to the domain cisco.com are failing and calls to sip.cisco.com are working. Which action resolves the issue?

A. Change the “Route Partition” setting on the *.cisco.com route pattern.

B. Add a SIP route pattern for *.*.

C. Select “Block Pattern” on the *.cisco.com route pattern.

D. Delete the *.cisco.com route pattern.

 


Suggested Answer: B

 

Question 8

55697959.007 |12:20:50.913 |AppInfo |RouteListCdrc::createPartyTransformedCcSetupReqMsg - before DAapplyCdpnXform() preXformCdpn=11112222 preTag=SUBSCRIBER prePos=11112222 crCdpnMask=33334444 crPrefixDigit= crDDI=2
55697959.008 |12:20:50.913 |AppInfo |RouteListCdrc::createPartyTransformedCcSetupReqMsg - after DAapplyCdpnXform() xformCdpn=33334444 xformTag=SUBSCRIBER xformPos=11112222
55697959.009 |12:20:50.913 |AppInfo |RouteListCdrc::transformed cdpn (without unconsumpt digits) = 33334444, unconsumed digit=
Refer to the exhibit. Which INVITE is sent to 10.10.100.123 as a result of this log?

A. 55698034.001 |12:20:50.922 |AppInfo |SIPTcp – wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.100.123 on port 5060 index 41[95992364,NET]INVITE sip:33334444@10.10.100.123:5060 SIP/2.0Via: SIP/2.0/TCP 10.122.200.50:5060;branch=z9hG4bK268d6e4e48f3aeFrom: “1000” ;tag=32412716~41f7To: Date: Thu, 01 Apr 2021 17:20:50 GMTCall-ID: 99878a80-66100f2-265e57-67071d0a@10.122.200.50Supported: timer,resource-priority,replacesMin-SE: 1800 -User-Agent: Cisco-CUCM12.0 –

B. 55698034.001 |12:20:50.922 |AppInfo |SIPTcp – wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.100.123 on port 5060 index 41[95992364,NET]INVITE sip:33334444@10.10.100.123:5060 SIP/2.0Via: SIP/2.0/TCP 10.122.200.50:5060;branch=z9hG4bK268d6e4e48f3aeFrom: “11112222” ;tag=32412716~41f7To: Date: Thu, 01 Apr 2021 17:20:50 GMTCall-ID: 99878a80-66100f2-265e57-67071d0a@10.122.200.50Supported: timer,resource-priority,replacesMin-SE: 1800 -User-Agent: Cisco-CUCM12.0 –

C. 55698034.001 |12:20:50.922 |AppInfo |SIPTcp – wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.100.123 on port 5060 index 41[95992364,NET]INVITE sip:11112222@10.10.100.123:5060 SIP/2.0Via: SIP/2.0/TCP 10.122.200.50:5060;branch=z9hG4bK268d6e4e48f3aeFrom: “1000” ;tag=32412716~41f7To: Date: Thu, 01 Apr 2021 17:20:50 GMTCall-ID: 99878a80-66100f2-265e57-67071d0a@10.122.200.50Supported: timer,resource-priority,replacesMin-SE: 1800 -User-Agent: Cisco-CUCM12.0 –

D. 55698034.001 |12:20:50.922 |AppInfo |SIPTcp – wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.100.123 on port 5060 index 41[95992364,NET]INVITE sip:11112222@10.10.100.123:5060 SIP/2.0Via: SIP/2.0/TCP 10.122.200.50:5060;branch=z9hG4bK268d6e4e48f3aeFrom: “11112222” ;tag=32412716~41f7To: Date: Thu, 01 Apr 2021 17:20:50 GMTCall-ID: 99878a80-66100f2-265e57-67071d0a@10.122.200.50Supported: timer,resource-priority,replacesMin-SE: 1800 -User-Agent: Cisco-CUCM12.0

 


Suggested Answer: C

 

Question 9

Image
Refer to the exhibit. 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?

A. answer-address 555 “¦”¦..

B. codec g729

C. session-protocol sipv2

D. incoming called number 555″¦”¦.

 


Suggested Answer: D

 

Question 10

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?

A. missing Call Admission Control

B. codec mismatch

C. ptime mismatch

D. phone class of service issue

 


Suggested Answer: B

 

Question 11

Image
Refer to the exhibit. 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.)

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.

 


Suggested Answer: AC

 

Question 12

Image
Refer to the exhibit. How many maximum hops can an ILS updarte traverse?

A. 3

B. 6

C. 9

D. 12

 


Suggested Answer: A

 

Question 13

Image
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?

A. voice class sip-profiles 2request INVITE sip-header To modify “sip:(.*)@” sip:8005532447@

B. voice translation-rule 3rule 1/.*/ /8005532447/

C. voice class sip-profiles 1request INVITE sip-header Diversion modify “sip:(.*)@” “sip:8005532447@”

D. voice translation-rule 4rule 1/^.*(8005532447)/ /1/

 


Suggested Answer: B

 

Question 14

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?

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

 


Suggested Answer: C

 

Question 15

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

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.

 


Suggested Answer: D

 

Question 16

Image
Refer to the exhibit. Users report that when they dial to Cisco Unity Connection from an external network, they cannot enter any digits. Assuming only in-band
DTMF is supported, what is a reason for this malfunction?

A. The negotiated RTP port is outside of the range described by RFC, so inband DTMFs do not work.

B. There is SIP Delayed Offer. DTMF is supported only in Early Offer.

C. The rtpmap:0 value for the negotiated codec is marking DTMF as inactive.

D. No DTMF is negotiated.

 


Suggested Answer: D

 

Question 17

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

A. incoming uri

B. target carrier-id

C. answer-address

D. incoming called-number

 


Suggested Answer: A

Reference:
https://www.cisco.com/c/en/us/support/docs/voice/ip-telephony-voice-over-ip-voip/211306-In-Depth-Explanation-of-Cisco-IOS-and-IO.html#anc8

Question 18

CollabCorp is a global company with two clusters, emea.collab.corp and apac.collab.corp. URI dialing is implemented and working in each cluster. The company configured routing between clusters to make inter-cluster calls via URI, but this is not working as expected. Which two configuration elements should be checked to resolve this issue? (Choose two.)

A. intercluster trunk

B. directory URI partition

C. SIP route pattern

D. calling search space and partition

E. SIP trunk

 


Suggested Answer: CE

 

Question 19

Which set of commands binds SIP media and signaling to interface GigabitEthernet0/0 when dial peer 1 is chosen for call routing?

A. dial-peer voice 1 voipvoice-class source interface GigabitEthernet0/0

B. voice service voipbind sip source-interface GigabitEthernet0/0

C. voice service voipsipbind all source-interface GigabitEthernet0/0

D. dial-peer voice 1 voipvoice-class bind control source-interface GigabitEthernet0/0voice-class sip bind media source-interface GigabitEthernet0/0

 


Suggested Answer: C

 

Question 20

Image
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?

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.

 


Suggested Answer: D

 

Question 21

An engineer is configuring a Cisco Collaboration system for SIP endpoints and must enable Survivable Remote Site Telephony for these endpoints. Which code completes this configuration on the SRST gateway?

A. call-manager-fallbackmax-conferences 8 gain –6ip source-address 10.10.10.100 port 2000max-ephones 100max-dn 200

B. telephony-servicemax-conferences 8 gain –6ip source-address 10.10.10.100 port 2000max-ephones 100max-dn 200

C. voice service voipdefault mode secureaddress hidingallow-connections sip to sipsip registrar

D. voice register globaldefault modeno allow-hash-in-dnmax-dn 100max-pool 200

 


Suggested Answer: A

 

Question 22

A user’s phone is already configured for Single Number Reach, and the user wants a feature to move an active call from a mobile phone to a desk phone and vice-versa. As an administrator, which additional configuration should be made to fulfill the user’s request?

A. Use Dialed Number Analyzer to determine if the user extension can dial the mobile phone.

B. Add the mobility key to the softkey template that the desk phone is using.

C. Check to make sure that the Resume softkey option appears on the desk phone.

D. Confirm that the desk phone is subscribed to Cisco Extension Mobility.

 


Suggested Answer: C

 

Question 23

Image
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 its own device pool. The DNA tool is used to test the configuration. How is this issue resolved?

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.

 


Suggested Answer: B

 

Question 24

Image
Refer to the exhibit. An administrator has configured a SIP trunk between two Cisco UCM clusters. For calls that should use the trunk, the calls fail with a fast busy. The administrator checks the Cisco CallManager SDL traces and found that the cluster to which the calling device is registered never sends an INVITE to the destination cluster. The administrator also verifies that all nodes from both clusters are powered on, and the CallManager service is running. How is this issue resolved?

A. The administrator must associate the route pattern with a calling search space the device can dial.

B. The administrator needs to enable OPTIONS pings on the SIP trunks for both clusters.

C. The administrator must allow connectivity so that TCP connections do not fail between the nodes.

D. The administrator needs to disable OPTIONS pings on the SIP trunks for both clusters.

 


Suggested Answer: C

 

Question 25

Configure Call Queuing in Cisco Unified Communications Manager. Where do you set the maximum number of callers in the queue?

A. in the telephony service configuration

B. in the queuing configuration

C. in Cisco Unified CM Enterprise Parameters

D. in Cisco Unified CM Service Parameters

 


Suggested Answer: B

Reference:
https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-communications-manager-callmanager/200453-Configure-CUCM-Native-

Call-Queuing-Featu.html

Question 26

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

A. Top Down, Round Robin, Broadcast

B. Top Down, Circular, Broadcast

C. Top Down, Round Robin, Distribute

D. Sequential, Circular, Broadcast

 


Suggested Answer: B

 

Question 27

Image
Refer to the exhibit. 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?

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.

 


Suggested Answer: C

 

Question 28

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

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.

 


Suggested Answer: D

 

Question 29

An administrator is working on an issue between the customer's Cisco Unified Border Element and the service provider. The provider only wants to see mid-call signaling from the Cisco Unified Border Element for fax calls. Which command must be configured on Cisco Unified Border Element?

A. midcall-signaling passthru

B. no update-callerid

C. midcall-signaling passthru media-change

D. midcall-signaling preserve-codec

 


Suggested Answer: C

 

Question 30

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?

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.

 


Suggested Answer: C

 

Question 31

Image
Refer to the exhibit. DN 1003 was the last to ring during the most recent call. Which hunting method ensures that DN 1005 is presented with the next call when the hunt pilot is dialed?

A. sequential

B. call-blast

C. peer

D. parallel

 


Suggested Answer: A

 

Question 32

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?

A. Location Bandwidth Manager

B. Extension Mobility

C. Device Mobility

D. Extension Mobility Cross Cluster

 


Suggested Answer: A

 

Question 33

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.
Image

B.
Image

C.
Image

D.
Image

 


Suggested Answer: B

 

Question 34

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.)

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

 


Suggested Answer: B

 

Question 35

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?

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 +

 


Suggested Answer: C

 

Question 36

Which elements does Cisco cloud mobility for collaboration include?

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

 


Suggested Answer: D

 

Question 37

A customer must advertise some numbers through GDPR on Call Manager. The Cisco UCM already is advertising some URIs to other clusters with GDPR, but the customer wants to ensure that specific numbers are advertised. Where under the Cisco UCM Admin page > Call Routing > Global Dial Plan Replication must that set of numbers be configured?

A. Route Numbers

B. Advertised Patterns

C. Learned Numbers

D. Route Patterns

 


Suggested Answer: B

 

Question 38

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?

A. dial-peer voice 1 voipincoming uri to

B. dial-peer voice 1 voipincoming called-number

C. dial-peer voice 1 voipincoming uri via

D. dial-peer voice 1 voipincoming uri request

 


Suggested Answer: C

 

Question 39

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

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

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

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

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

 


Suggested Answer: B

 

Question 40

Due to a shortage of physical interfaces on a device, the administrator requires that a loopback for RTP is used. Which command is required when using a loopback interface for RTP?

A. voice-class sip early-offer forced

B. voice-class sip bind control source-interface Loopback0

C. voice-class sip bind media source-interface Loopback0

D. voice-class sip resource priority mode passthrough

 


Suggested Answer: C

 

Question 41

Image
Refer to the exhibit. Which change to the translation rule is needed to strip only the leading 9 from the digit string 9123548?

A. rule 1 /^9(d*)/ /l/

B. rule 1 /^9(.*)/ /l/

C. rule 1 /.*(3548$)/ /l/

D. rule 1 /^9123548/ /1/

 


Suggested Answer: B

 

Question 42

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

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

 


Suggested Answer: C

Reference:
https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-communications-manager-callmanager/213583-procedure-to-analyse-call-
flow-of-sip-ca.html

Question 43

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?

A. voice service voipallow-connections sip to h323

B. voice service voipsipbind media source-interface Vlan100

C. voice service voipsipbind control source-interface Vlan100

D. voice service voipsipregistrar server expires max 600 min 60

 


Suggested Answer: D

 

Question 44

Image
Refer to the exhibit. 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.)

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.

 


Suggested Answer: DE

 

Question 45

Image
Refer to the exhibit. A collaboration engineer is troubleshooting an issue where the PSTN calls of a Cisco UCM IP phone user are not reaching the PSTN gateway. Which action resolves the issue?

A. Change the calling search space of the user’s line or device.

B. Change the “Call Classification” to “OnNet” on the route pattern.

C. Ensure that the user’s phone is assigned to a device pool with the correct local route settings.

D. Deselect “Block this pattern” on the route pattern.

 


Suggested Answer: D

 

Question 46

You see the voice register pool 1 command in your Cisco Unified Communications Manager Express configuration. Which configuration is occurring in this section?

A. configuration for a single SIP phone

B. configuration items common for all SIP phones

C. configuration for a pool of SIP phones (similar to device pool on Cisco Unified Communications Manager)

D. configuration for SIP registrar service

 


Suggested Answer: C

Reference:
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cusrst/admin/sccp_sip_srst/configuration/guide/SCCP_and_SIP_SRST_Admin_Guide/
srst_setting_up_using_sip.html

Question 47

Which description of RTP timestamps or sequence numbers is true?

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).

 


Suggested Answer: D

Reference:
https://www.cs.columbia.edu/~hgs/rtp/faq.html

Question 48

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

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

 


Suggested Answer: B

Reference:
https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/dtmf-relay.html

Question 49

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?

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

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

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

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

 


Suggested Answer: C

 

Question 50

Which two descriptions of the Standard Local Route Group deployment are true? (Choose two.)

A. can be associated under the route group

B. can be associated only under the route list

C. chooses the route group that is configured under the device pool of the calling-party device

D. chooses the route group that is configured under the device pool of the called-party device

E. can be assigned directly to the route pattern

 


Suggested Answer: BD

 

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