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300-815 Test Engine Practice Test Questions, Exam Dumps
NEW QUESTION # 44
Refer to the exhibit.
How many maximum hops can an ILS update traverse?
- A. 0
- B. 1
- C. 2
- D. 3
Answer: B
NEW QUESTION # 45
A customer is using a SIP trunk to route calls to ITSP to decrease the possibility of downtime, the customer invested in a failover device How does the customer ensure reachability to ITSP, so that if one device on ITSP fails, the calls will be routed to another device?
- A. Monitor the link using network management toots, and if it fails, manually change the routing to another working device.
- B. Enable ANAT on the SIP profile.
- C. Enable transmit security status on the SIP security profile
- D. Enable SIP Option Ping on the SIP profile.
Answer: D
NEW QUESTION # 46
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. BFCP
- B. AUDIO
- C. VIDEO
- D. FAX
- E. DTMF
Answer: A,E
NEW QUESTION # 47
The Cisco Unified Communications Manager Dialed Number Analyzer allows analysis of calls from which two devices? (Choose two.)
- A. translation patterns
- B. device pools
- C. CTI ports
- D. CTI route points
- E. IP phones
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 # 48
How does an engineer globalize routing for ingress calls coming from the PSTN to internal DNs?
- A. At the PSTN gateway, put the calling number in E.164 format and the called number in localized (DN) format.
- B. At Cisco Unified CM, put the calling number in E.164 format and the called number in PSTN format.
- C. At Cisco Unified Communications Manager, put the calling number in E.164 format and the called number in E.164 format.
- D. At the PSTN gateway, put the calling number in PSTN format and the called number in DN format.
Answer: B
Explanation:
Section: Call Control and Dial Planning
NEW QUESTION # 49
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.246 asn 1
- B. debug H.225 media
- C. debug H.225 asn1
- D. debug H.323 asn 1
- E. debug H.323 messages
Answer: C
NEW QUESTION # 50
In Cisco Unified Communications Manager, which tool do you use to check SIP traces?
- A. RTMT
- B. CCSIP
- C. OS Administration Page
- D. MTP
Answer: A
NEW QUESTION # 51
When a third-party SIP Phone System is dialed inbound across a Cisco Unified Border Element, DTMF is failing. The third-party vendor accepts only out-of-band DTMF. Which configuration should be added to the outgoing dial peer to resolve this issue?
- A. dtmf-relay h245-signal
- B. dtmf-relay cisco-rtp
- C. dtmf-relay rtp-nte
- D. dtmf-relay sip-kpml
Answer: D
NEW QUESTION # 52
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. ptime mismatch
- B. codec mismatch
- C. phone class of service issue
- D. missing Call Admission Control
Answer: D
NEW QUESTION # 53
The administrator of ABC company is troubleshooting a one-way audio issue for a call that uses H.323 protocol (slow-start mode). The administrator requests that you provide the IP and port information of the Real- Time Transport Protocol traffic that had the one-way audio call.
You gather the H.225 and H.245 messages for one of the one-way audio calls. Where can you find the RTP IP and port information for both sides? (Note: This call flow has not invoked any media resources like MTP or transcoders).
- A. H.245 Open Logical Channel Ack
- B. H.245 Open Logical Channel
- C. H.225 Connect
- D. H.245 Terminal Capability Set
Answer: B
Explanation:
Reference:
http://ccievoicehopeful.blogspot.com/2012/09/h323-notes.html
NEW QUESTION # 54
Refer to the exhibit.
How many maximum hops can an ILS update traverse?
- A. 0
- B. 1
- C. 2
- D. 3
Answer: B
NEW QUESTION # 55
Refer to the exhibit.
A call mode through the Cisco Unified Border Element to pilot 2000 is foiling. What is causing the call to foil?
- A. The destination pattern is incorrect for the dialed number.
- B. No codecs are configured on the dial peers
- C. VAD was not disabled on the outgoing dial poor.
- D. The Cisco Unified Border Element is not receiving a response to its OPTION keepahves.
Answer: A
NEW QUESTION # 56
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 voip
incoming called-number - B. dial-peer voice 1 voip
incoming url request - C. dial-peer voice 1 voip
incoming url to - D. dial-peer voice 1 voip
incoming url via
Answer: D
NEW QUESTION # 57
What are two configureation features of the Client matter code setting in the route pattern configuration? (Choose two.)
- A. Selecting the Allow Overlap Sending setting allows a user to select the Require Client Matter Code setting.
- B. Selecting the Allow Overlap Sending setting disables the Require Client Matter Code setting.
- C. The Client Matter Code feature provides the option to configure Authorization Level susch as in the Forced Authorization Code.
- D. The client Matter Code feature supports overlap sending since the Cisco UCM can determine when to prompt the user for the code.
Answer: B,C
NEW QUESTION # 58
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. Content-Disposition: session:handling=required
- B. Content-Type: application/sdp
- C. Require 100rel
- D. Contact <sip:[email protected]:5060>
Answer: C
NEW QUESTION # 59
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