[2022年09月] 確実合格する有効な方法Cisco試験問題集で300-815試験学習ガイド
300-815問題集とImplementing Cisco Advanced Call Control and Mobility Servicesトレーニングコースでお客様の合格を楽にさせる学習合格試験問題！
質問 56
An administrator is working on an issue between the customer s Cisco Unified Border Element and the service provider. The provider only wants to see mid-call signaling from the Cisco Unified Border Element for fax calls. Which command must be configured on Cisco Unified Border Element?
- A. midcall-signaling preserve-codec
- B. midcall-signaling passthru media-change
- C. no update-callerid
- D. midcall-signallng passthru
正解: B
質問 57
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. Go to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 16384-32767
- 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.
正解: B,D
質問 58
Which call pickup feature allows users to pick up incoming calls in a group that is associated with their own group?
- A. Group Call Pickup
- B. Other Group Pickup
- C. Directed Call Pickup
- D. BLF Call Pickup
正解: B
質問 59
In Cisco Unified Communications Manager, which tool do you use to check SIP traces?
- A. OS Administration Page
- B. MTP
- C. CCSIP
- D. RTMT
正解: D
解説:
Section: Call Control and Dial Planning
質問 60
Refer to the exhibit.
How many maximum hops can an ILS update traverse?
- A. 0
- B. 1
- C. 2
- D. 3
正解: C
質問 61
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. A firewall in the media path is blocking TCP ports 16384-32768.
- C. The RTP traffic is arriving beyond the jitter buffer on the receiving end.
- D. Cisco Unified Communications Manager invoked media termination point resources.
正解: C
質問 62
The Cisco Unified Communications Manager Dialed Number Analyzer allows analysis of calls from which two devices? (Choose two.)
- A. CTI route points
- B. device pools
- C. CTI ports
- D. translation patterns
- E. IP phones
正解: C,E
解説:
Section: Call Control and Dial Planning
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
質問 63
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. enterrise alternate number mask
- B. AAR destination mask
- C. +E.164 alternate number mask
- D. external phone number mask
- E. called party transform mask
正解: B,D
解説:
Reference:
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/collab10/collab10/dialplan.html
質問 64
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. gateway.corporate.local
- B. *.*
- C. example.com
- D. *@example.com
- E. [email protected]
正解: B,D
質問 65
Calls to a particular extension are not routing to voicemail. The user reaches the voicemail system from the handset by pressing the Messages button Which configuration parameter causes this problem?
- A. The voicemail pilot number for call forwarding is missing from the ephone
- B. The voicemail pilot number is missing from the telephony service configuration on Cisco UCME
- C. The voicemail pilot number for call forwarding is missing from the ephone-dn
- D. The voicemail pilot number is missing from the call handling on Cisco Unity Express
正解: C
質問 66
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. phone class of service issue
- C. codec mismatch
- D. ptime mismatch
正解: C
質問 67
Which description of RTP timestamps or sequence numbers is true?
- A. The sequence number is used to detect losses.
- B. The timestamp is used to place the incoming audio and video packets in the correct timing order (playout delay compensation).
- C. Timestamps increase by the time "carrying" by a packet.
- D. Sequence numbers increase by four for each RTP packet transmitted.
正解: B
質問 68
How does an engineer globalize routing for ingress calls coming from the PSTN to internal DNs?
- A. At Cisco Unified Communications Manager, put the calling number in E.164 format and the called number in E.164 format.
- B. At the PSTN gateway, put the calling number in PSTN format and the called number in DN format.
- C. At Cisco Unified CM, put the calling number in E.164 format and the called number in PSTN format.
- D. At the PSTN gateway, put the calling number in E.164 format and the called number in localized (DN) format.
正解: C
解説:
Section: Call Control and Dial Planning
質問 69
An engineer is configuring a call park feature in Cisco Unified Communications Manager Express. Which command does the engineer use to ensure that the call is reverted to the user after 60 seconds?
- A. R2(config-ephone-dn)#park reservation-group 1
- B. R2(config-ephone-dn)#park-slot timeout 30 limit 2 recall alternate 3002
- C. R2(config-ephone-dn)#park-slot timeout 60 limit 2 recall alternate 3002
- D. R2(config-ephone-dn)#park reservation-group 60
正解: C
解説:
Section: Cisco Unified CM Call Control Features
質問 70
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. phone class of service issue
- B. codec mismatch
- C. ptime mismatch
- D. missing Call Admission Control
正解: D
質問 71
What is a component of Cisco Unified Mobility?
- A. Unified IVR
- B. Smart Client Support
- C. Single Number Connect
- D. Mobile Connect
正解: D
質問 72
Which configuration must an administrator perform to display Translation Pattern operations in Cisco Unified Communications Manager SDL traces?
- A. By default, the Translation Patterns operations are printed in SDL traces, so no additional configuration is necessary.
- B. Set up the Digit Analysis Complexity in Service Parameters for Cisco Unified CM to TranslationAndAlternatePatternAnalysis.
- C. Check the Translation Patterns Analysis check box in Micro Traces on the Cisco Unified CM Serviceability page.
- D. Enable the Detailed Call Analysis option under Enterprise Parameters for Unified CM.
正解: A
質問 73
An engineer is troubleshooting local ringback on a Cisco SIP gateway The gateway is not ignoring the SIP 180 response with SDP from the service provider, and the far end device is sending the 180 with SDP to play ringback from the IP address specified m the SDP Which configuration change must be made on the gateway to resolve the issue?
- A. Router(conf-voi-serv)# dlisable-early-media 180
- B. Router(config-sip-ua)# no disable-early-media 180
- C. Router(conftg-sip-ua)# disable-early-media 180
- D. Router(con(-voi-serv)# no disable-early-media 180
正解: C
質問 74
After configuring a Cisco CallManager Express with Cisco Unity Express, inbound calls from the PSTN SIP trunk receive a ring tone for 20 seconds and then a busy signal instead of voicemail. Which configuration fixes this problem?
- A. Router(config)#dial-peer voice 2 voip
Router(config-dial-peer)#no vad - B. Router(config)# voice service voip
Router(conf-voi-serv)#no supplementary-service sip moved-temporarily - C. Router(config)# voice service voip
Router(conf-voi-serv)#allow-connections h323 to h323 - D. Router(config)# voice service voip
Router(conf-voi-serv)#allow-connections voice-mail mod
正解: B
質問 75
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リアル試験問題と解答Cisco 300-815問題集はここに：https://drive.google.com/open?id=1rko5atHZuGywqwb2Yi8hgaLrGj4xIpTD
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