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Avaya 31861X Exam Syllabus Topics:
| Section | Weight | Objectives |
|---|---|---|
| Licensing, Sizing & Capacity Planning | 20% | - User and endpoint sizing - Redundancy and scalability planning - License models and allocation |
| Design Validation & Best Practices | 15% | - Design review and validation - Avaya design tools and documentation - Resilience and fault tolerance |
| Avaya OneCloud UCaaS Overview & Architecture | 20% | - Avaya OneCloud platform components - Core UCaaS concepts and service models - Avaya IX Calling, Spaces, and Messaging overview |
| Integration & Interoperability | 20% | - Migration considerations from on-premise - SIP trunking and connectivity - Integration with Avaya Aura and third-party systems |
| Solution Design & Requirements Gathering | 25% | - Customer requirement analysis - Network and bandwidth planning - Security and compliance design |
Avaya OneCloud™ UCaaS Design Certified Sample Questions:
Question 1
Your customer is asking about the Multiple Appearance Directory Number (MADN) feature that has been added in support of CS1000 migrations. It includes Single Call Appearance/Arrangement (SCA) which works like Communication Manager bridging, and multiple Call Appearance/Arrange (MCA) which works like bridging on the CS1000.
Which statement about MCA is true?
A. It is only available on Avaya Equinox clients.
B. It makes all calls private.
C. It is tied to a particular call appearance on the primary endpoint.
D. It is only available on Device Adapted sets.
Question 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. 2400 Kbps
C. 1600 Kbps
D. 6400 Kbps
Question 3
A customer wants to prevent additional calls from being established, and to preserve the quality active calls when the capacity limit on the WAN link is reached. The branch inter-office communication use rate is one trunk for every 20 users over the customer's star topology WAN. 600k of the WAN bandwidth is dedicated to voice.
Assuming G.729 (30k bandwidth per call) is used, which Call Admission Control (CAC) limit will meet the customer s requirement?
A. Set the CAC limit to 10 incoming calls and 10 outgoing inter-branch calls at each branch.
B. Set the CAC limit to 2 Mbps just for voice traffic at the headquarters main office WAN connection.
C. Set the CAC limit to 200 intra-branch calls globally in the WAN network.
D. Set the CAC limit to 20 inter-branch calls for each branch location.
Question 4
A customer wants to upgrade their current S8720 Communication Manager 4 (CM) system. They have 2200 users, and a mixture of analog, DCP and IP telephones at their main location. They have six branches with 200-300 stations each served with G700 gateways, each containing an LSP. They want to gain the cost benefits of SIP trunking, and are reluctant to change out the user endpoints, but they want to experiment with a few SIP telephones for some of their remote agents. Survivability and resiliency are essential requirements.
Which elements are required for the solution?
A. Survivable Remote with Branch Session Manager, CM Duplex Core, and Session Border Controller for Enterprise (SBCE)
B. Session Manager (SM), System Manager (SMGR), CM Duplex Core, Survivable Remote with Branch Session Manager, and Session Border Controller for Enterprise (SBCE)
C. Survivable Remote, CM Duplex Core, and Session Border Controller for Enterprise (SBCE)
D. Session Manager (SM), System Manager (SMGR), Survivable Remote with Branch Session Manager, and CM Duplex Core
Question 5
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.
Solutions:
| Question 1 Answer: B | Question 2 Answer: D | Question 3 Answer: D | Question 4 Answer: C | Question 5 Answer: B |




