Free CCNP Collaboration 300-815 Ultimate Study Guide (Updated 120 Questions)
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To prepare for the Cisco 300-815 exam, candidates can take advantage of a variety of resources, including training courses, study guides, and practice exams. Many of these resources are available through Cisco's official certification program, as well as through third-party training providers. Candidates should also have hands-on experience with the technologies covered on the exam, as this is a crucial component of success.
Cisco 300-815 exam is designed for IT professionals who want to validate their skills in implementing advanced call control and mobility services using Cisco technologies. 300-815 exam is a part of the Cisco Certified Specialist - Collaboration Call Control and Mobility Implementation certification track, which focuses on advanced collaboration solutions. 300-815 exam tests the candidates' knowledge and skills in managing Cisco Unified Communications Manager (CUCM), Cisco Unity Connection, Cisco IM and Presence, and Cisco Expressway series. It also evaluates the candidates' expertise in implementing advanced call control and mobility services, such as SIP trunks, video integration, mobile and remote access, and Cisco Webex Teams integration.
NEW QUESTION # 30
Which configuration element of a hunt group allows for changing Calling Party Transformations settings?
- A. route group
- B. hunt list
- C. line group
- D. hunt pilot
Answer: D
NEW QUESTION # 31
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. ILS Active Learned Object Upper Limit
- B. Global Data Service Parameter Limit
- C. ILS Max Number of Learned Objects in Database
- D. Imported Dial Plan Replication Database Object Lower Limit
Answer: C
NEW QUESTION # 32
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. ptime mismatch
- C. codec mismatch
- D. phone class of service issue
Answer: A
NEW QUESTION # 33
In Cisco Unified Communications Manager, which tool do you use to check SIP traces?
- A. RTMT
- B. OS Administration Page
- C. CCSIP
- D. MTP
Answer: A
Explanation:
Section: Call Control and Dial Planning
NEW QUESTION # 34
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. Voice/Video > Session Trace Log View > Open From Local Disk
- C. Voice/Video > Session Trace Log View > Real Time Data
- D. System > Tools > Trace and Log Central
Answer: C
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 # 35
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 16384-32767
- B. Go to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 20000-22000.
- C. Ask the firewall administrator to change the ports to TCP.
- D. Ask the firewall administrator to change the range of UDP ports to 16384-32767.
- E. Go to System Parameters in Cisco Unified Communications Manager and change the range of media ports to 20000-22000.
Answer: A,D
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 # 36
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?
- A. Reroute all calls through the PSTN and avoid using WAN.
- B. Configure AAR to reroute calls that are denied by Call Admission Control through the PSTN.
- C. Disable Call Admission Control and let the calls use the amount of bandwidth they require.
- D. Configure Call Queuing so that the user waits until there is bandwidth available
Answer: B
NEW QUESTION # 37
In Cisco Unified Communications Manager, which tool do you use to check SIP traces?
- A. RTMT
- B. OS Administration Page
- C. CCSIP
- D. MTP
Answer: A
NEW QUESTION # 38
Refer to the exhibit.
For long-distance calls, users must prefix their dialed number with "91." The translation pattern was created to strip the 91 as the PSTN expects a 10- digit number. The PSTN also requires the calling number to be set to 9195551234. However, the service provider has said calls with a different calling number are being received. How is this issue resolved?
- A. Enable Force Authorization Code on the route pattern.
- B. Enable Use Calling Party's External Phone Number Mask on the translation pattern.
- C. Disable Use Calling Party's External Phone Number Mask on the route pattern.
- D. Change the partition of the translation pattern from none to pstn_pt.
Answer: C
NEW QUESTION # 39
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?
- A. 813510[^0-5]
- B. 813510[012345]
- C. 81XXXXX
- D. 813510[12345]
Answer: B
Explanation:
Section: Call Control and Dial Planning
NEW QUESTION # 40
Which two descriptions of the Standard Local Route Group deployment are true? (Choose two.)
- A. can be assigned directly to the route pattern
- B. can be associated under the route group
- C. chooses the route group that is configured under the device pool of the calling-party device
- D. can be associated only under the route list
- E. chooses the route group that is configured under the device pool of the called-party device
Answer: D,E
NEW QUESTION # 41
Which two descriptions of the Standard Local Route Group deployment are true? (Choose two.)
- A. can be assigned directly to the route pattern
- B. can be associated under the route group
- C. chooses the route group that is configured under the device pool of the calling-party device
- D. can be associated only under the route list
- E. chooses the route group that is configured under the device pool of the called-party device
Answer: D,E
Explanation:
Section: Call Control and Dial Planning
NEW QUESTION # 42
An engineer must configure a secure SIP trunk with a remote provider, with a specific requirement to use port 5065 for inbound and otubound traffic. Which two items must be configured to complete this configuration? (Choose two.)
- A. Incoming Port in SIP Information section of the SIP Trunk configuration.
- B. Incoming Port in SIP Trunk Security Profile configuration
- C. Destination Port in SIP Information section of the SIP Trunk configuration
- D. Incoming Port in Security Information of the SIP Profile configuration.
- E. Destination Port in SIP Trunk Security Profile configuration
Answer: B,C
NEW QUESTION # 43
An IP Telephony administrator is deploying IP phones The administrator has an existing Cisco UCME router with several SCCP & SIP phones registered. The administrator receives a request for a new SIP phone with MAC address 1111 2222.3333 and directory number 2050 to be added in the Cisco UCME. Which two configurations should be added in CME to support this request? (Choose two )
- A. Option B
- B. Option A
- C. Option C
- D. Option D
- E. Option E
Answer: C,D
NEW QUESTION # 44
Refer to the exhibit.
While troubleshooting call failures on the Cisco Unified Border Element, an administrator notices that messages are being sent to the service provider, but there is no response The administrator later learns that this SIP provider does not support PRACK. Which header should be removed from the SIP message to resolve this issue?
- A. Require 100rel
- B. Content-Type: application/sdp
- C. Contact <sip:[email protected]:5060>
- D. Content-Disposition: session:handling=required
Answer: A
NEW QUESTION # 45
Which two statements are correct with respect to the Client Matter Code setting in the route pattern configuration? (Choose two.)
- A. The Client Matter Code feature does support overlap sending because the Cisco Unified Communications Manager can determine when to prompt the user for the code.
- B. If you check the Allow Overlap Sending check box, the Require Client Matter Code check box becomes disabled.
- C. If you check the Allow Overlap Sending check box, you can also check the Require Client Matter Code check box.
- D. The Client Matter Code feature does not support overlap sending because the Cisco Unified CM cannot determine when to prompt the user for the code.
- E. The Client Matter Code has the option to configure Authorization Level such as in the Forced Authorization Code.
Answer: B,D
Explanation:
Reference:
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/10_0_1/ccmfeat/ CUCM_BK_F3AC1C0F_00_cucm-features-services-guide-100/CUCM_BK_F3AC1C0F_00_cucm-features- services-guide-100_chapter_010000.pdf
NEW QUESTION # 46
......
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