[Jan 27, 2022] 300-815 PDF Questions and Testing Engine With 120 Questions
Updated Exam Engine for 300-815 Exam Free Demo & 365 Day Updates
Cisco 300-815 Exam Topics:
| Section | Weight | Objectives |
|---|---|---|
| Signaling and Media Protocols | 20% | - Troubleshoot these elements of a SIP conversation
-Troubleshoot these H.323 protocol elements
|
| Cisco Unified CM Call Control Features | 20% | -Troubleshoot Call Admission Control (exclude RSVP) -Configure ILS, URI synchronization, and GDPR -Configure hunt groups -Configure call queuing -Configure time of day routing -Configure supplementary functions
|
| Call Control and Dial Planning | 25% | - Configure these globalized call routing elements in Cisco Unified CommunicationsManager
-Troubleshoot these globalized call routing elements in Cisco Unified CommunicationsManager
|
| Mobility | 10% | -Configure Cisco Unified Communications Manager Mobility
-Troubleshoot Cisco Unified Communications Manager Mobility
|
| CME/SRST Gateway Technologies | 10% | -Configure Cisco Unified Communications Manager Express for SIP phone registration -Configure Cisco Unified CME dial plans -Implement toll fraud prevention -Configure these advanced Cisco Unified CME features
-Configure SIP SRST gateway |
| Cisco Unified Border Element | 15% | -Configure these Cisco Unified Border Element dial plan elements
-Troubleshoot these Cisco Unified Border Element dial plan elements
|
NEW QUESTION 41
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 dial peer of the gateway has the correct destination pattern configured.
- B. Verify that the route pattern is not blocking calls to the destination number.
- C. Verify that IP routing is correct between the gateway and the IP phone.
- D. Verify that the route pattern has the correct calling-party transformation mask
Answer: A
NEW QUESTION 42
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. Check the Urgent Priority check box under 9.911 pattern.
- B. Adjust the service parameter T204 timer to the desired value.
- C. Adjust the service parameter T302 timet to the desired value.
- D. Point the emergency pattern directly to the PSTN gateway.
Answer: A
NEW QUESTION 43
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 has the option to configure Authorization Level such as in the Forced Authorization 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 feature does support overlap sending because the Cisco Unified Communications Manager can determine when to prompt the user for the 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 44
Refer to the exhibit.
Which change to the translation rule is needed to strip only the leading 9 from the digit string 9123548?
- A. rulel /.*\(3548S\)/^1/
- B. rule 1/^9123548/^1/
- C. rule 1 /^9\(.*\)/A1/
- D. rulel /^9\(\d*\)/^1/
Answer: C
NEW QUESTION 45
Refer to the exhibit.
Calls incoming from the provider are not working through newly set up Cisco Unified Border Element. Provider engineers get the 404 Not Found SIP message. Incoming calls are coming from the provider with called number "222333444" and Cisco Unified Communications Manager is expecting the called number to be delivered as "444333222". The administrator already verified that the IP address of the Cisco Unified CM is set up correctly and there are no dial peers configured other than those shown in the exhibit. Which action must the administrator take to fix the issue?
- A. Fix the voice translation-rule to match specifically number "222333444" and change it to "444333222".
- B. Change the destination-pattern on the outgoing dial peer to match "444333222".
- C. Create specific matching for "222333444" on the incoming dial peer.
- D. Set up translation-profile on the incoming dial peer to match incoming traffic.
Answer: D
NEW QUESTION 46
Which IOS command creates a SIP-enabled dial peer?
- A. dial peer voice 20 sip
- B. dial-peer voice 20 voip
- C. dial-peer voice 20 pots
- D. voice dial-peer 20 sip
Answer: B
NEW QUESTION 47
An engineer must route all SIP calls in the form of <user>@example.com to the SIP trunk gateway corporate local. Which two SIP route patterns can be used to accomplish this task? (Choose two.)
- A. [email protected]
- B. *.*
- C. *@example.com
- D. example.com
- E. gateway.corporate.local
Answer: B,C
NEW QUESTION 48
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. Contact: header of the 200 OK response
- B. c= line of SDP content
- C. Allow: header if the 200 OK response
- D. o= line of SDP content
Answer: D
Explanation:
Section: Signaling and Media Protocols
NEW QUESTION 49
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. Ask the firewall administrator to change the ports to TCP.
- C. Ask the firewall administrator to change the range of UDP ports to 16384-32767.
- D. Go to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 20000-22000.
- E. Go to System Parameters in Cisco Unified Communications Manager and change the range of media ports to 20000-22000.
Answer: A,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 50
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.)
- A. Create a route group which has all the gateways and associate it to the device pool of every site.
- B. Assign one route group as a local route group in the device pool of the corresponding site.
- C. Create a hunt group and assign it to each side route pattern
- D. Create one route group for each site and one global route list for PSTN calls that point to the local route group.
- E. Create one global route list for PSTN calls that points to one global PSTN route group.
Answer: B,D
NEW QUESTION 51
Which two types of distribution algorithm are within a line group? (Choose two.)
- A. top down
- B. circular
- C. random
- D. bottom up
- E. highest preference
Answer: A,B
NEW QUESTION 52
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 011-1-613-555-1234 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 613-555-1234 and numbering type "subscriber"
Answer: C
NEW QUESTION 53
An administrator is troubleshooting call failures on an H.323 gateway via the CLI. To see signaling for media and call setup, which two debugs must the administrator turn on? (Choose two).
- A. debug H.323 message
- B. debug H.225 media
- C. debug H.225 asn1
- D. debug H.323 asn1
- E. debug H.245 asn1
Answer: C,E
NEW QUESTION 54
Refer to the exhibit.
An administrator is troubleshooting a problem in which some outbound calls from an internal network to the Internet telephony service provider are not getting connected, but some others connect successfully. The firewall team found that some call attempts on port 5060 came from an unrecognized IP that has not been defined in the firewall rule. What should the administrator configure in the Cisco Unified Border Element to fix this issue?
- A. bind signaling and media to the loopback interface
- B. ip prefix-list to filter the unwanted IP address
- C. use of port 5061 for SIP secure
- D. access list allowing the firewall IP
Answer: A
NEW QUESTION 55
Refer to the exhibit.
An engineer is troubleshooting a call-establishment problem between Cisco Unified Border Element and Cisco UCM. Which command set corrects the issue?
- A. SIP binding in dial-peer configuration mode:
dial-peer voice 100 volp
voice-class sip bind control source-interface GigabitEthernetO/0/0
voice-class sip bind media source-interface GigabitEthernetO/0/0 - B. SIP binding in SIP configuration mode:
voice service voip sip
bind control source-interface GigabitEthernetO/0/0 bind media source-interface GigabitEthernetO/0/0 - C. SIP binding In SIP configuration mode:
voice service volp
sip
bind control source-Interface GlgabltEthernetO/0/1 bind media source-Interface GlgabltEthernetO/0/1 - D. SIP binding In dial-peer configuration mode:
dial-peer voice 300 voip
voice-class sip bind control source-interface GigabitEthernetO/0/1 voice-class sip bind media source-interface GigabitEthernetO/0/1
Answer: A
NEW QUESTION 56
An engineer has temporarily disabled toll fraud prevention for SIP line calls on a Cisco CME12.6x and must enforce security and toll fraud prevention for the SIP line side on Cisco Unified CME. Which configuration must be used to start this process?
- A. voice service volp
enable Ip address trust list - B. voice service volp
enablo ip address trust authentication - C. voice service volp
ip address trusted authenticate - D. voice service volp
Ip address trusted list
Answer: C
NEW QUESTION 57
Refer to the exhibit.
How many maximum hops can an ILS update traverse?
- A. 0
- B. 1
- C. 2
- D. 3
Answer: D
NEW QUESTION 58
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. The RTP traffic is arriving beyond the jitter buffer on the receiving end.
- C. A firewall in the media path is blocking TCP ports 16384-32768.
- D. Cisco Unified Communications Manager invoked media termination point resources.
Answer: C
NEW QUESTION 59
The Cisco Unified Communications Manager Dialed Number Analyzer allows analysis of calls from which two devices? (Choose two.)
- A. CTI ports
- B. translation patterns
- C. CTI route points
- D. device pools
- E. IP phones
Answer: A,E
Explanation:
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 60
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