Cisco 300-815 Test Engine Practice Test Questions, Exam Dumps

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Official Courses for Cisco 300-815 Exam Preparation

Before taking the exam, you have to master all the above topics. The most rational way to do this is by adopting corresponding Cisco training. These are three programs you can apply for:

• Implementing and Operating Cisco Collaboration Core Technologies (CLCOR)

  If you start from the very beginning, it’s better to first enroll in training for the core test. It lets you prepare for the Cisco CCNP Collaboration from scratch, covering the most basic concepts. During it, you'll prepare for advanced-level positions based on Cisco collaboration technologies deployment and service. You will also acquire the expertise and skills one needs to have to install and deploy common collaboration and networking technologies. Also, it is focused on the areas like configuration, call management, service quality, and other Cisco collaboration applications. It can be followed as 8-day sessions guided by an instructor, 8-day virtual training, or E-learning.

• Implementing and Understanding Cisco Advanced Call Control and Mobility Services (CLACCM)

  During this official course, you will get solid knowledge about how to utilize CU-CM features to simplify the communications system into a modular, lightweight, and reliable collaboration approach. You will also learn about several other features, such as Globalized Call Routing, Cisco Extension Mobility, Cisco Unified Mobility, and many others. Using this course as your guide will bolster your studies for the test. The training is offered in three forms: 5 days of instructor-led classes, 5 days of virtual sessions, or a set of E-learning tools and materials equivalent to 5-day preparation.

• Understanding Cisco Collaboration Foundations (CLFNDU)

  This is one more training course that would make you ready for this test. It provides you with all the information you need to know to maintain a simple Cisco Unified Communications Manager and handle the Session Initiation Protocol (SIP) gateway. It encompasses preliminary settings, the management of systems, customer management, media services management, as well as tools used for troubleshooting and maintenance of CU Communications solutions. This program also has a duration of 5 days regardless of its format, which includes the options such as instructor-led, virtual classroom, and E-learning.

 

NEW QUESTION 70
A user requests a feature to send an active call to the mobile phone number on the physical phone. As an administrator, what should be configured in the Cisco UCM to accomplish this?

• A. A Remote Destination Profile having the same extension as the physical phone's Directory Number, add the mobile phone number to the RDP as a Remote Destination. Add the softkey "Join" to the physical phone's softkey template.
• B. A Remote Destination Profile having the same extension as the physical phone's Directory Number, add the mobile phone number to the RDP as a Remote Destination. Add the softkey "Mobility" to the physical phone's softkey template.
• C. A Remote Destination Profile having the same extension as the Mobile phone's number adds the physical phone's Directory Number to the RDP as a Remote Destination. Add the softkey "Join" to the physical phone's softkey template
• D. A Remote Destination Profile having the same extension as the Mobile phone's number adds the physical phone's Directory Number to the RDP as a Remote Destination. Add the softkey "Mobility" to the physical phone's softkey template.

Answer: B

 

NEW QUESTION 71
Cisco SIP IP telephony is implemented on two floors of your company. Afterward, users report intermittent voice issues in calls established between floors. All calls are established, and sometimes they work well, but sometimes there is one-way audio or no audio. You determine that there is a firewall between the floors, and the administrator reports that it is allowing SIP signaling and UDP ports from 20000 to 22000 bidirectionally.
What are two possible solutions? (Choose two.)

• A. Go to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 20000-22000.
• B. Ask the firewall administrator to change the range of UDP ports to 16384-32767.
• C. Go to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 16384-32767
• D. Ask the firewall administrator to change the ports to TCP.
• E. Go to System Parameters in Cisco Unified Communications Manager and change the range of media ports to 20000-22000.

Answer: B,C

Explanation:
Section: Signaling and Media Protocols
Explanation/Reference: https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/port/9_1_1/ CUCM_BK_T2CA6EDE_00_tcp-port-usage-guide-91/CUCM_BK_T2CA6EDE_00_tcp-port-usage-guide-
91_chapter_01.html

 

NEW QUESTION 72
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 resources priority mode passthrough
• C. voice-class sip bind control source-interface Loopback0
• D. voice-class sip bind media source-interface Loopback0

Answer: D

 

NEW QUESTION 73

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. Activate the Cisco Dialed Number Analyzer Server service.
• B. Activate the Cisco Dialed Number Analyzer service.
• C. Restart the subscriber
• D. Activate the Cisco CallManager service.
• E. Activate the Cisco Extended Functions service.

Answer: A,B

Explanation:
Section: Call Control and Dial Planning

 

NEW QUESTION 74
Which two extended capabilities must be configured on dial peers for fast start-to-early media scenarios (H.323 to SIP interworking)? (Choose two.)

• A. VIDEO
• B. DTMF
• C. BFCP
• D. FAX
• E. AUDIO

Answer: B,C

 

NEW QUESTION 75
Which call pickup feature allows users to pick up incoming calls in a group that is associated with their own group?

• A. BLF Call Pickup
• B. Other Group Pickup
• C. Directed Call Pickup
• D. Group Call Pickup

Answer: B

Explanation:
Reference:
https://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/4_1_3/ccmsys/ a07cpick.html#wp1022865

 

NEW QUESTION 76
Which section under the Real-Time Monitoring Tool allows for reviewing the call flow and signaling for a SIP call in real time?

• A. Voice/Video > Session Trace Log View > Real Time Data
• B. Voice/Video > Session Trace Log View > Open From Local Disk
• C. System > Tools > Trace and Log Central
• D. Analysis Manager > Inventory > Trace File Repositories

Answer: A

 

NEW QUESTION 77
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?

• A. Global Data Service Parameter Limit
• B. Imported Dial Plan Replication Database Object Lower Limit
• C. ILS Max Number of Learned Objects in Database
• D. ILS Active Learned Object Upper Limit

Answer: C

Explanation:
Section: Cisco Unified CM Call Control Features
Explanation/Reference: https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/12_5_1SU1/systemConfig/ cucm_b_system-configuration-guide-1251su1/cucm_b_system-configuration-guide-
1251su1_restructured_chapter_0100011.html#CUCM_TK_I7C708C2_00

 

NEW QUESTION 78
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?

• A. Change the partition of the route pattern
• B. Modify the route pattern to add a prefix of 91
• C. Add a local route group on the device pool configuration.
• D. Add a sip trunk inside route group Standard Local Route Group.

Answer: C

 

NEW QUESTION 79
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 disable OPTIONS pings on the SIP trunks for both clusters.
• C. The administrator needs to enable OPTIONS pings on the SIP trunks for both clusters.
• D. The administrator must allow connectivity so that TCP connections do not fail between the nodes.

Answer: D

 

NEW QUESTION 80
Which IOS command creates a SIP-enabled dial peer?

• A. dial-peer voice 20 pots
• B. dial-peer voice 20 voip
• C. voice dial-peer 20 sip
• D. dial peer voice 20 sip

Answer: B

Explanation:
Reference:
https://www.ciscopress.com/articles/article.asp?p=664148&seqNum=6

 

NEW QUESTION 81
Where is the dtmf-relay command configured on Cisco Unified Border Element?

• A. in the voice-class VoIP configuration
• B. in the VoIP or POTS dial peers
• C. in global SIP configuration
• D. in the VoIP dial peer

Answer: D

 

NEW QUESTION 82

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. Point the emergency pattern directly to the PSTN gateway.
• C. Adjust the service parameter T204 timer to the desired value.
• D. Check the Urgent Priority check box under 9.911 pattern.

Answer: D

Explanation:
Section: Call Control and Dial Planning

 

NEW QUESTION 83
Which two configuration parameters are prerequisites to set Native Call Queuing on Cisco Unified Communications Manager? (Choose two.)

• 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. The phone button template must have the Queue Status Softkey configured.
• D. The maximum number of callers allowed in queue must be 10.
• E. Cisco RIS data collector service must be running on the same server as the Cisco CallManager service.

Answer: A,E

Explanation:
Reference:
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/12_0_1/systemConfig/ cucm_b_system-configuration-guide-1201/cucm_b_system-configuration-guide- 1201_chapter_01001101.html#CUCM_RF_C960BC9A_00

 

NEW QUESTION 84
Where is the dtmf-relay command configured on Cisco Unified Border Element?

• A. in the voice-class VoIP configuration
• B. in the VoIP or POTS dial peers
• C. in global SIP configuration
• D. in the VoIP dial peer

Answer: D

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

 

NEW QUESTION 85

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 rtpmap:0 value for the negotiated codec is marking DTMF as inactive.
• B. The negotiated RTP port is outside of the range described by RFC, so inband DTMFs do not work.
• C. No DTMF is negotiated.
• D. There is SIP Delayed Offer. DTMF is supported only in Early Offer.

Answer: C

Explanation:
Section: Signaling and Media Protocols

 

NEW QUESTION 86
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?

• A. Allow: header if the 200 OK response
• B. c= line of SDP content
• C. o= line of SDP content
• D. Contact: header of the 200 OK response

Answer: B

 

NEW QUESTION 87
Why would RTP traffic that is sent from the originating endpoint fail to be received on the far endpoint?

• A. Cisco Unified Communications Manager invoked media termination point resources.
• B. The far end connection data (c=) in the SDP was overwritten by deep packet inspection in the call signaling path.
• 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.

Answer: D

 

NEW QUESTION 88
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. Localize the calling and called numbers to PSTN format and globalize the calling and called numbers in the gateway.
• C. Globalize the calling and called numbers to PSTN format and localize the calling number in the gateway.
• D. Globalize the calling and called numbers to E. 164 format and localize the called number in the gateway.

Answer: D

 

NEW QUESTION 89
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. Activate the Cisco Dialed Number Analyzer Server service.
• B. Activate the Cisco Dialed Number Analyzer service.
• C. Restart the subscriber
• D. Activate the Cisco CallManager service.
• E. Activate the Cisco Extended Functions service.

Answer: A,B

 

NEW QUESTION 90
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?

• A. c= line of SDP content
• B. o= line of SDP content
• C. Allow: header if the 200 OK response
• D. Contact: header of the 200 OK response

Answer: B

Explanation:
Section: Signaling and Media Protocols

 

NEW QUESTION 91
Which section under the Real-Time Monitoring Tool allows for reviewing the call flow and signaling for a SIP call in real time?

• A. Voice/Video > Session Trace Log View > Real Time Data
• B. Voice/Video > Session Trace Log View > Open From Local Disk
• C. System > Tools > Trace and Log Central
• D. Analysis Manager > Inventory > Trace File Repositories

Answer: A

Explanation:
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

 

NEW QUESTION 92
If all patterns below are configured in Cisco Unified Communications Manager which would be used when dialing the pattern "123"?

• A. 12X (urgent priority set)
• B. 12[2-5]
• C. 1XX (urgent Priority Set)
• D. 12!

Answer: B

 

NEW QUESTION 93
......

What is the duration, language, and format of Cisco 300-815: Implementing Cisco Advanced Call Control and Mobility Services Exam

• Applicability: 3 years
• Format: Multiple selection
• Exam language: English
• Exam duration: 90 minutes

Career Opportunities and Salary Profile

The Cisco 300-815 exam can offer you a lot of opportunities if you pass it with a high result. After getting certified, you will be able to land jobs, such as a Cisco Voice Engineer, a Cisco Network Engineer, a Network Security Engineer, a UC Engineer, a Collaboration Engineer, or a Technical Consulting Engineer. After choosing your path, you can start earning about an average of $109,353 per year.

 

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