2025年最新の実際の300-815問題集最新Cisco練習テスト問題集ゲット
300-815問題集PDFで300-815リアル試験問題解答
Cisco 300-815試験は、Ciscoの技術と高度なコールコントロールとモビリティサービスの応用における深い理解が必要な難しい試験です。試験は、Ciscoの技術を使用して高度なコールコントロールとモビリティサービスを実装することにおける候補者の知識とスキルをテストする複数選択問題とシミュレーションで構成されています。
質問 # 95
Which call pickup feature allows users to pick up incoming calls in a group that is associated with their own group?
- A. BLF Call Pickup
- B. Other Group Pickup
- C. Group Call Pickup
- D. Directed Call Pickup
正解：B
解説：
https://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/4_1_3/ccmsys/a07cpick.html#wp
1022865
質問 # 96
The SIP session refresh timer allows the RTP session to stay active during an active call. The Cisco UCM sends either SIP-INVITE or SIP-UPDATE messages in a regular interval of time throughout the active duration of the call. During a troubleshooting session, the engineer finds that the Cisco UCM is sending SIP-UPDATE as the SIP session refresher, and the engineer would like to use SIP-INVITE as the session refresher. What configuration should be made in the Cisco UCM to achieve this?
- A. Increase Retry INVITE to 20 seconds on the SIP profile.
- B. Enable SIP ReMXX Options on the SIP profile.
- C. Change Session Refresh Method on the SIP profile to INVITE.
- D. Enable Send send-receive SDP in mid-call INVITE on the SIP profile.
正解：C
質問 # 97
When configuring hunt groups, where do you add the individual directory numbers that will be part of the group?
- A. line group
- B. hunt list
- C. hunt pilot
- D. route group
正解：A
解説：
Reference:
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/12_0_1/systemConfig/ cucm_b_system-configuration-guide-1201/cucm_b_system-configuration-guide-1201_chapter_010101.html
質問 # 98
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 011613-555-1234 and numbering type "international"
- B. calling number 613-555-1234 and numbering type "national"
- C. calling number 613-555-1234 and numbering type "subscriber"
- D. calling number 011-1-613-555-1234 and numbering type "subscriber"
正解：B
質問 # 99
An administrator configured Cisco Unified Mobility to block access to remote destinations for certain caller IDs. A user reports that a blocked caller was able to reach a remote destination. Which action resolves the issue?
- A. Configure a mobility identity.
- B. Configure Mobile Voice Access.
- C. Configure Single Number Reach.
- D. Configure an access list.
正解：D
質問 # 100
Refer to the exhibit.
Refer to the exhibit An engineer deploys a Cisco Unified Border Element for SIP trunk interconnection to an ITSP The engineer performs test calls but there is no audio when the called party picks up the phone. What must the engineer do to resolve the issue?
- A. Disable SIP early-media for 180 responses with SDP
- B. Decrease the SIP max-forwards value to 16.
- C. Disable SIP User Agent bind status (media).
- D. Enable Check media source packets.
正解：B
質問 # 101
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. 81XXXXX
- B. 813510[012345]
- C. 813510[^0-5]
- D. 813510[12345]
正解：B
解説：
Section: Call Control and Dial Planning
質問 # 102
Refer to the exhibit.
A user reports that when they call a specific phone number, no one answers the call, but when they call from a mobile phone, the call is answered. The engineer troubleshooting the issue is expecting the far-end gateway to cut through audio on the 183 Session Progress SIP message. Which SIP Profile configuration element is necessary for the Cisco Unified Communications Manager to send acknowledgement of provisional responses?
- A. Allow Passthrough of Configured Line Device Caller Information must be enabled.
- B. On the SIP Profile, the configuration parameter SIP Rel1XX Options must be set to Send PRACK for all
1xx Messages. - C. Accept Audio Codec Preferences in Received Offer must be set to On.
- D. Early Offer for G Clear Calls must be enabled.
正解：B
質問 # 103
When route patterns are defined as precisely as possible on a Cisco UCM, the control and reliability of the calling plan is increased. A route pattern must be added to support calls to this number range: 9135200 to 9135205. The calls are on-net and no translation patterns are configured. Which configuration meets these requirements?
- A. 913520X
- B. 913520!
- C. 913520[012345]
- D. 913520[1-5]
正解：C
質問 # 104
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. System > Location Info
- C. Call Routing > Route/Hunt > Local Route Group Names
- D. System > Device Pool
正解：C
質問 # 105
Which two statements about Cisco Unified Mobility are true? (Choose two.)
- A. Mobile use of Cisco Jabber requires mobility
- B. Single number reach is a feature of mobility
- C. Mobility just be enabled per user
- D. Mobility is used to support mobile phones in Cisco Unified Communications
- E. Mobility is enabled by default
正解：B、C
質問 # 106
Which two types of authentication are supported for the configuration of Intercluster Lookup Service? (Choose two.)
- A. TLS certificates
- B. TokenID
- C. FQDN of the servers defined in DNS
- D. username and secret key
- E. passwords
正解：A、E
質問 # 107
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. Enable SIP Option Ping on the SIP profile.
- B. Enable transmit security status on the SIP security profile
- C. Monitor the link using network management toots, and if it fails, manually change the routing to another working device.
- D. Enable ANAT on the SIP profile.
正解：A
質問 # 108
Which description of RTP timestamps or sequence numbers is true?
- A. The timestamp is used to place the incoming audio and video packets in the correct timing order (playout
- B. Sequence numbers increase by four for each RTP packet transmitted.
- C. Timestamps increase by the time "carrying" by a packet.
- D. The sequence number is used to detect losses.
正解：A
解説：
delay compensation).
質問 # 109
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. 81XXXXX
- B. 813510[012345]
- C. 813510[^0-5]
- D. 813510[12345]
正解：B
質問 # 110
Refer to the exhibit.
An administrator just Implemented SIP trunking on their Cisco UCM and reports that calls using the SIP trunk are using Media Termination Point resources unnecessarily. Which action resolves the issue?
- A. Change OTMF Signaling Method to "No Preference".
- B. Change DTMF Signaling Method to "RFC 4733".
- C. Disable SIP Red XX Options.
- D. Change to a range that does not result in MTP.
正解：A
質問 # 111
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 System Parameters in Cisco Unified Communications Manager and change the range of media ports to 20000-22000.
- C. Ask the firewall administrator to change the range of UDP ports to 16384-32767.
- D. Ask the firewall administrator to change the ports to TCP.
- E. Go to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 16384-32767
正解：C、E
解説：
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
質問 # 112
An engineer is troubleshooting Cisco Device Mobility and find that the phone has roamed to a building that is assigned to a different device pool but has not changed its device pool accordingly What action resolves the issue?
- A. Set Device CSS under Current Device Mobility Settings.
- B. Set the correct subnet under Device Mobility Info.
- C. Set correct Location under Current Device Mobility Settings
- D. Enable SRST under Current Device Mobility Settings
正解：B
質問 # 113
The Cisco Unified Communications Manager Dialed Number Analyzer allows analysis of calls from which two devices? (Choose two.)
- A. translation patterns
- B. CTI ports
- C. IP phones
- D. device pools
- E. CTI route points
正解：B、C
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 # 114
A company has an SRST gateway running an IOS XE image. The company plans to enable the IPv6 addressing companywide. To enable the IPv6 in a unified SRST gateway to support SIP phones, what are two supported supplementary features for an IPv6 fallback scenario? (Choose two.)
- A. secure SIP lines
- B. T.38 fax relay
- C. SIP trunk
- D. transcoding
- E. three-way conference
正解：B、E
解説：
The following supplementary services are supported as part of IPv6 in Unified SRST IP Phones:
Hold/Resume
Call Forward
Call Transfer
Three-way Conference (with BIB conferencing only)
Line to T1/E1 Trunk and Trunk to Line with Supplementary Service Features Fax to and from PSTN (IPv4 ATA to ISDN T1/E1) for both T.38 Fax Relay and Fax Passthrough
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_sip_isr4000.html
質問 # 115
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