300-815 Dumps 2024 - New Cisco 300-815 Exam Questions
Free 300-815 Braindumps Download Updated on Oct 26, 2024 with 176 Questions
To earn the Cisco 300-815 certification, candidates are required to pass the 90-minute exam, which consists of 60-70 questions. 300-815 exam is designed to evaluate the candidate’s understanding of the concepts, technologies, and solutions related to advanced call control and mobility services. Candidates who pass the exam demonstrate their ability to implement and manage complex call control and mobility services solutions, troubleshoot issues, and optimize network performance in a Cisco environment. Implementing Cisco Advanced Call Control and Mobility Services certification is highly valued in the industry and can open up new career opportunities for IT professionals who possess the skills and knowledge required to implement advanced call control and mobility services using Cisco technology.
NEW QUESTION # 12
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 for a pool of SIP phones (similar to device pool on Cisco Unified Communications Manager)
- C. configuration for SIP registrar service
- D. configuration items common for all SIP phones
Answer: B
Explanation:
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
NEW QUESTION # 13
In Cisco Unified Communications Manager globalized call routing is implemented and must confirm that it is correctly implemented without making a call. Which tool do you use for verification?
- A. Dialed Number Analyzer
- B. SDI trace
- C. SDL trace
- D. Real-Time Monitoring Tool
Answer: A
Explanation:
Section: Call Control and Dial Planning
NEW QUESTION # 14
Where on Cisco Unified Communications Manager do you configure the standard local route group for a group of devices?
- A. Call Routing > Emergency Location > Emergency Location (ELIN) Groups
- B. Call Routing > Route/Hunt > Local Route Group Names
- C. System > Location Info
- D. System > Device Pool
Answer: B
NEW QUESTION # 15
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.
- B.
- C.
- D.
Answer: D
NEW QUESTION # 16
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?
- A. debug H.323 messages
- B. debug H.225 asn1
- C. debug H.323 asn 1
- D. debug H.225 media
- E. debug H.246 asn 1
Answer: B
Explanation:
Section: Signaling and Media Protocols
NEW QUESTION # 17
A newly deployed site in Pennsylvania uses SIP-to-SIP legs on a Cisco Unified Border Element. The administrator wants to match the incoming dial peer based on the calling number. Which configuration is needed in the dial-peer?
- A. incoming called-number
- B. destination calling
- C. destination-pattern
- D. answer-address
Answer: A
NEW QUESTION # 18
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?
- A. dtmf-relay h245-alphanumeric
- B. dtmf-relay sip-kpml
- C. dtmf-relay cisco-rtp
- D. dtmf-relay rtp-nte
Answer: B
NEW QUESTION # 19
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 > Open From Local Disk
- B. System > Tools > Trace and Log Central
- C. Analysis Manager > Inventory > Trace File Repositories
- D. Voice/Video > Session Trace Log View > Real Time Data
Answer: D
Explanation:
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 # 20
Which description of RTP timestamps or sequence numbers is true?
- A. Sequence numbers increase by four for each RTP packet transmitted.
- B. Timestamps increase by the time "carrying" by a packet.
- C. The timestamp is used to place the incoming audio and video packets in the correct timing order (playout delay compensation).
- D. The sequence number is used to detect losses.
Answer: C
Explanation:
Reference:
https://www.cs.columbia.edu/~hgs/rtp/faq.html
NEW QUESTION # 21
An administrator discovers that employees are making unauthorized long-distance and international calls from logged-off Extension Mobility phones when the authorized users are away from their desks Which two configurations should the administrator configure in the Cisco UCM to avoid this issue? (Choose two.)
- A. Add the long-distance & international pattern's partitions to the calling search space of the physical phone.
- B. Add the long-distance & international pattern's partitions to the calling search space of the device profile
- C. Remove the long-distance & international pattern's partitions from the calling search space of the physical phone.
- D. Remove the long-distance & international pattern's partitions from the calling search space of the device profile.
- E. Add the long-distance & international pattern's partitions to the calling search space of the physical phone's directory number.
Answer: B,C
NEW QUESTION # 22
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 > Open From Local Disk
- B. System > Tools > Trace and Log Central
- C. Analysis Manager > Inventory > Trace File Repositories
- D. Voice/Video > Session Trace Log View > Real Time Data
Answer: D
NEW QUESTION # 23 
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 011613-555-1234 and numbering type "international"
- C. calling number 613-555-1234 and numbering type "national"
- D. calling number 011-1-613-555-1234 and numbering type "subscriber"
Answer: C
Explanation:
Section: Call Control and Dial Planning
Explanation/Reference:
NEW QUESTION # 24
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?
- A. out-of-band
- B. gateway
- C. in-band
- D. channel
Answer: A
NEW QUESTION # 25
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. incoming called number 555.......
- B. answer-address 555 ........
- C. codec g729
- D. session-protocol sipv2
Answer: D
NEW QUESTION # 26
Refer to the exhibit.
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?
- A. Change the Cisco UCM routing logic to the longest-match algorithm.
- B. Select Allow Overlap Sending for the \+! route pattern.
- C. Set the interdigit timeout service parameter value to 0.
- D. Mark the \+ [2-9JXX[2-91XXXXXX route pattern with urgent priority.
Answer: D
NEW QUESTION # 27 
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. incoming called number 555.......
- D. session-protocol sipv2
Answer: C
Explanation:
Section: Call Control and Dial Planning
NEW QUESTION # 28
Which configuration element of a hunt group allows for changing Calling Party Transformations settings?
- A. line group
- B. hunt list
- C. hunt pilot
- D. route group
Answer: C
Explanation:
Section: Cisco Unified CM Call Control Features
Explanation/Reference:
Reference: https://community.cisco.com/t5/ip-telephony-and-phones/call-alerting-on-hunt-group-as-shared-line/ td-p/2658015
NEW QUESTION # 29
When the services key is pressed Cisco Extension Mobility does not show up. What is the cause of the issue?
- A. Cisco Extension Mobility is not enabled in the Phone Configuration Window (Device > Phone)
- B. The phone is not subscribed to Cisco Extension Mobility Service.
- C. Cisco Extension Mobility Service is not running.
- D. The URL configured for Cisco Extension Mobility is not correct.
Answer: B
NEW QUESTION # 30
An administrator discovers that employees are making unauthorized long-distance and international calls from logged-off Extension Mobility phones when the authorized users are away from their desks.
Which two configurations should the administrator configure in the Cisco UCM to avoid this issue? (Choose two.)
- A. Remove the long-distance & international pattern's partitions from the calling search space of the physical phone's directory number.
- B. Add the long-distance & international pattern's partitions to the calling search space of the physical phone.
- C. Add the long-distance & international pattern's partitions to the calling search space of the device profile
- D. Remove the long-distance & international pattern's partitions from the calling search space of the device profile.
- E. Add the long-distance & international pattern's partitions to the calling search space of the physical phone's directory number.
Answer: A,C
NEW QUESTION # 31
When locations-based Call Admission Control denies the call, which two masks can AAR apply when routing the call through the PSTN? (Choose two.)
- A. external phone number mask
- B. enterrise alternate number mask
- C. +E.164 alternate number mask
- D. called party transform mask
- E. AAR destination mask
Answer: A,E
NEW QUESTION # 32
The Cisco Unified Communications Manager Dialed Number Analyzer allows analysis of calls from which two devices? (Choose two.)
- A. device pools
- B. translation patterns
- C. IP phones
- D. CTI route points
- E. CTI ports
Answer: C,E
Explanation:
Section: Call Control and Dial Planning
Explanation/Reference:
Reference: https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/dna/11_5_1/ CUCM_BK_CBA47A6E_00_cucm-dna-guide-115/CUCM_BK_CBA47A6E_00_cucm-dna-guide-
115_chapter_01.html#CUCM_TP_A5DA99E0_00
NEW QUESTION # 33
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Cisco 300-815 exam measures candidates' abilities to configure, implement, and troubleshoot advanced call control and mobility services. 300-815 exam covers topics such as Cisco Unified Communications Manager (CUCM) call routing and signaling, device registration and mobility, media resources, and Quality of Service (QoS) policies. 300-815 exam also tests candidates' knowledge of Cisco Collaboration applications, including Cisco Unity Connection, Cisco Unified IM and Presence, and Cisco Expressway.
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Updated Certification Exam 300-815 Dumps - Practice Test Questions: https://drive.google.com/open?id=1Cvh_nxUjPD9MPdxVF0Hj2SafWnsv8jkH