31860Xの無料問題集（62題）、31860X認定試験参考書

Avaya 31860X 問題集

試験コード：31860X

試験名称：Avaya IX™ Calling Design Exam

最近更新時間：2025-04-20

問題と解答：全62問

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質問 1：
You have a customer who requires both SIP Trunking and Remote Workers. After examining the features needed overall, you have determined that 700 Core Suite licenses are required. You still need to quote SBCE licenses, in addition to the entitlements included with Core Suite licenses, to support a total of 120 PSTN SIP trunks and 330 remote workers. The customer has agreed to a 3 remote worker users to 1 session access basis-How many additional standard and advanced a la carte SBCE licenses are needed? (Choose two.)
A. 10 SBCE Advanced Licenses
B. 20 SBCE Standard Licenses
C. 30 SBCE Standard Licenses
D. 20 SBCE Advanced Licenses
正解：A

質問 2：
A customer has an existing single-location Communication Manager solution with G650 gateways and ISDN trunks. You have been asked to design a new location for 200 users. The new location will support IP telephones, basic emergency calling to public safety agencies, and local PSTN trunks. There is a requirement for Standard Local Survivability.
Which design meets the customer's requirements?
A. G430 gateway, IP telephones, MM711 analog media module, and analog central office lines
B. G450 gateway, IP telephones, MM716 analog media module, and analog central office lines
C. G430 gateway, IP telephones, MM710 DS1 media module, and an ISDN-PRI from the service provider
D. G450 gateway, IP telephones, MM710 DS1 media module, and an ISDN-PRI from the service provider
正解：A

質問 3：
On an Avaya Aura 7.x deployment with shuffling turned on within Communication Manager (CM), which statement describes the media processing resources or Digital Signaling Processor (DSP) requirements for a SIP-SIP two-party call between endpoints using the same codec?
A. This call cannot shuffle, and two media-processing (DSP) channels are required for the entire call duration.
B. SIP-SIP two-party calls will use SIP Direct Media, so no media-processing (DSP) channels are required for this call.
C. This call will shuffle after initial call set-up; i.e., one media-processing (DSP) channel is required for the first 20 seconds of the call, for each party.
D. SIP-SIP multi-party calls do not shuffle, and after setup use one media processing (DSP) channel for the entire call..
正解：C

質問 4：
You are designing a Communication Manager (CM) 8 solution with centralized ISDN trunks at the main site and two remote gateway locations. Each remote location will be connected to the main server via dedicated, point-to-point WAN service. Main - CM 8.1 duplex pair with 2-G450s and 250 IP stations Remote 1 - S8300E LSP/G450 with 125 IP stations Remote 2 - S8300E LSP/G430 with 30 IP stations.
To ensure that no one site Is overwhelmed by CM traffic, what is the recommended number of network regions required for this design?
A. 4 regions: 1 at the main, 1 at each remote site, and 1 for the virtual network region
B. 7 regions: 1 at the main, 1 at each remote site, 1 for each site's IP telephones, and one for the virtual network region
C. 3 regions: 1 at the main, and 1 at each remote site
D. 6 regions: 1 at the main, 1 at each remote site, and 1 virtual network region for each site
正解：C

質問 5：
You are creating a new Communication Manager design. You have included Avaya provided servers with G450 gateways. The solution requires 200 - H.323 telephones. Trunk service from the PSTN will be IP trunks.
Shuffling will be enabled. They have also requested call recording via Device Media Call Control (DMCC) application programming interface.
To record all IP endpoints simultaneously, how many DSP resources are required?
A. 600 DSP resources
B. 800 DSP resources
C. 200 DSP resources
D. 400 DSP resources
正解：B

質問 6：
A customer has provided the following requirements for a basic telephone user:
* Two line appearances
* Basic telephony
* "3 rings then cover to voice mail"
* No shared line appearances
* Cheapest telephone
* Possible low power requirements
Which telephone type would be a good choice for the basic user?
A. 9611G
B. J129
C. J179
D. 9641
正解：C

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Avaya IX™ Calling Design 認定 31860X 試験問題:

1. Your customer has a CTI application that requires Avaya Aura Application Enablement Services (AES). To work with AES, how must Communication Manager be operating?

A) As a Feature Server
B) In a survivable remote mode
C) As an Evolution Server
D) In the Avaya Aura Core

2. A customer has 80 remote users, and you expect 25% of the users will be connected at any one time using
G.729. Assuming the following requirements:
* a G.729 codec requires 30 kbps
* a G711 codec requires 80 kbps
* 720p HD video codec requires 512 Kbps
How much bandwidth is needed for a public interface WAN link of the centralized SBCE for the deskphones used by remote workers?

A) 600 Kbps
B) 1600 Kbps
C) 2400 Kbps
D) 6400 Kbps

3. A Sales Engineer is designing an Avaya Aura 8.x solution where 900 video users are being added. There is enough bandwidth for all users to establish video calls at 6 Mbps.
Which video codec will provide the best video quality that the network will support?

A) H.263
B) H.262
C) H.264
D) H.261

4. If a customer has an Avaya Aura 7.0 and shuffling is turned on within Communication Manager, which statement describes the media processing resource (DSP) requirements for a two-party call originated statement describes the media processing resource (DSP) requirements for a two-party call originated by a SIP station and terminated by an .H323 IP Trunk being recoded?

A) SIP-SIP two-calls calls will use SIP Direct Media, so no media-processing channels are required for this call.
B) This call cannot shuffle, and one media-processing channel is required for the entire call.
C) This call will shuffle after initial call set-up; i.e., one media-processing (DSP) channel is required for the first 20 seconds of the call, for each party.
D) Multi-party (i.e. 3-party or greater) calls cannot shuffle, and require media-processing channels for the entire call.

5. You are adding 1400 users to an existing Avaya Aura communication core- The customer wants the same services for alt users, but with this expansion there Is not enough bandwidth for audio and video calls for all.
If the bandwidth parameter Is set lower than the minimum video requirement, what will happen to users Initiating audio and video calls?

A) Audio calls will connect but video calls will not initiate.
B) Audio and video calls will connect for all users.
C) Audio calls will connect and video calls will connect as audio calls.
D) Audio and video calls will connect for some users.

質問と回答：

質問 # 1
正解： B

質問 # 2
正解： D

質問 # 3
正解： B

質問 # 4
正解： B

質問 # 5
正解： D