Passing exam with 72201X latest training materials
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Updated: Aug 30, 2026
No. of Questions: 96 Questions & Answers with Testing Engine
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Avaya 72201X Exam Overview:
| Certification Vendor: | Avaya |
|---|---|
| Exam Name: | Avaya Aura® Core Components Support Certified Exam |
| Exam Number: | 72201X |
| Exam Duration: | 105 minutes |
| Exam Format: | Multiple-choice, Scenario-based |
| Related Certifications: | ACSS-7220 ACIS-7120 |
| Exam Price: | USD 125 |
| Available Languages: | English |
| Real Exam Qty: | 66 |
| Passing Score: | 68% or 700/1000 |
| Certificate Validity Period: | 2 years |
| Recommended Training: | Avaya Aura Core Components Support Training |
| Exam Registration: | Pearson VUE Avaya Learning |
| Sample Questions: | Avaya 72201X Sample Questions |
| Exam Way: | Proctored exam via Pearson VUE test centers or online proctoring |
| Pre Condition: | No mandatory prerequisites; recommended hands-on experience with Avaya Aura |
| Official Syllabus URL: | https://www.avaya-learning.com/ |
Avaya 72201X Exam Syllabus Topics:
| Section | Weight | Objectives |
|---|---|---|
| Maintenance, Troubleshooting & Recovery | 15% | - Fault isolation and resolution - Monitoring and logging - Backup, restore and disaster recovery |
| Fundamental Voice Network Processes & Standards | 20% | - SIP protocol fundamentals - QoS and voice quality - RTP/media streams |
| Avaya Aura Core Architecture | 20% | - System architecture overview - Component roles and interactions - Deployment models |
| Avaya Aura Core Components | 25% | - Communication Manager - System Manager - Media Server and Gateways - Session Manager |
| Call and Message Flows | 20% | - Call routing and session management - SIP trunking and interworking - Registration and authentication |
Avaya Aura® Core Components Support Certified Sample Questions:
Question 1
What happens if Session Manager cannot find a matching SIP Communication Profile for a user in its SIP registry when that user tries to make a SIP call?
A. The call is routed to voicemail to check if the extension is associated with a mailbox.
B. The caller receives a busy tone.
C. The call is sent to Communication Manager to check if the number dialed exists in CM's call routing tables.
D. Network Routing Policy (NRP) is consulted for further routing instructions.
E. Session Manager attempts to create a new SIP Communication Profile for the user.
Question 2
Refer to the Exhibits.
What is the main difference between the SIP call flow made from an Avaya Communicator Remote Worker and the call flow made from a 96X1 telephone Remote Worker?
A. Avaya Aura Communication Manager processes full call model for the SIP-to-Communicator call flow.
B. The 96X1 telephone Remote Worker requires registration to Avaya Aura Session Manager.
C. Avaya Aura Communication Manager processes half-call model for the SIP-to-SIP call flow.
D. The Communicator registers to Avaya Session Border Controller for Enterprise (SBCE).
Question 3
What information can be found when running the display node-names ip command?
A. IP address and type
B. type and link number
C. name and IP address
D. type and IP address
E. name and link number
F. MAC address and IP address
Question 4
Avaya Aura System Manager (SMGR) holds a copy of the Avaya Aura Communication Manager (CM) database, and it is important that the database is kept identical in both SMGR and CM.
If they ever stop being synchronized, what should be done to get them synchronized again?
A. Go to the Inventory > Synchronize > Communication System, and send the database from SMGR to CM.
B. Use the command 'sync database' on both CM and SMGR to synchronize the databases.
C. Go to the Inventory > Synchronize > Communication System, and synchronize the database from CM to SMGR.
D. Issue a save translations all command in CM.
E. Go to the Replication page and issue a Repair of the database on SMGR.
Question 5
Which five statements about media-processing resources (DSPs) are true? (Choose five.)
A. If a direct media path cannot be established between two IP endpoints, the call will fail from release 7.0.
B. CM can use DSP resources for features such as conferencing, music-on-hold, and voice messaging.
C. SIP-SIP multi-party calls will shuffle to establish a direct media path if CM is configured to do so.
D. Two-party calls originated by SIP stations or trunks, and terminated by H.323 trunks, media gateways, or other vendors' H.323 stations, will typically shuffle if CM is configured to do so.
E. H.323-H.323 two-party calls will always shuffle to establish a direct media path if CM is configured to do so.
F. Two-party calls originated by H.323 stations, trunks, or media gateways, and terminated by SIP stations or trunks, cannot shuffle.
G. SIP-SIP two-party calls will always use SIP Direct Media if Communication Manager (CM) is configured to do so.
Solutions:
| Question 1 Answer: D | Question 2 Answer: A | Question 3 Answer: C | Question 4 Answer: C | Question 5 Answer: A,B,E,F,G |
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