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A. Discipline 3 - Prevent Recurrences, Discipline 4 - Determine Root Cause

B. Discipline 3 - Develop Interim Containment Actions, Discipline 5 - Choose and Verify Corrective Actions

C. Discipline 3 - Implement Corrective Actions, Discipline 4 - Determine Root Cause

D. Discipline 3 - Develop Interim Containment Actions, Discipline 4 - Determine Root Cause

E. Discipline 3 - Implement a Work-around, Discipline 4 - Determine Root Cause

A. No trunk-group members have been configured.

B. The VoIP Channel License Limit has not been set on the media-server CM SAT screen.

C. The Media Server is not added as a resource to the call processing entity.

D. The CM license has no CM Media Server VoIP Channels.

E. The Dedicated VoIP Channel Licenses has not been set on the media-server CM SAT screen.

A. Knowledge Management

B. Cause

C. Problem Statement

D. Details/Findings

E. Problem Clarification

F. Troubleshooting steps taken

A. Acknowledged

B. Raised

C. Ignored

D. Active

E. Pending

F. Cleared

G. Closed

H. Resolved

A. SIP-SIP multi-party calls will shuffle to establish a direct media path if CM is configured to do so.

B. CM can use DSP resources for features such as conferencing, music-on-hold, and voice messaging.

C. 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.

D. H.323-H.323 two-party calls will always shuffle to establish a direct media path if CM is configured to do so.

E. SIP-SIP two-party calls will always use SIP Direct Media if Communication Manager (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. If a direct media path cannot be established between two IP endpoints, the call will fail from release 7.0.

A. Execute the SM CLI statapp command.

B. On the SM dashboard, select Status Applications from the Shutdown System pull-down menu.

C. Execute the SM status all command.

D. Use the Linux command ps -ef | grep -i "SM"

E. On the SM dashboard, select Status Applications from the Service State pull-down menu.

A. SIP trunks can be configured on both Avaya Aura Session Manager (SM) and Avaya Aura Communication Manager (CM).

B. SIP User Agents can register to both Avaya Aura Session Manager (SM) and Avaya Aura Communication Manager (CM).

C. Avaya Aura Session Manager (SM) is responsible for routing calls between SIP User Agents.

D. Avaya Aura Media Server (AAMS) connects directly to Avaya Aura Session Manager (SM) using SIP.

E. Avaya Aura Media Server (AAMS) can connect to Avaya Aura Communication Manager (CM) without routing via Avaya Aura Session Manager (SM).

A. Session Manager attempts to create a new SIP Communication Profile for the user.

B. The call is sent to Communication Manager to check if the number dialed exists in CM's call routing tables.

C. The call is routed to voicemail to check if the extension is associated with a mailbox.

D. The caller receives a busy tone.

E. Network Routing Policy (NRP) is consulted for further routing instructions.

A. Create a Bandwidth Share Group in SM

B. Configure SIP authentication between CM and SM

C. Create Locations in Session Manager

D. Create Network Region Groups (NRG) in Communication Manager

E. Match the Network Region used for the SIP users with the Domain in SM

F. Specify the shared bandwidth limit on the Communication Manager (SIP) Entity screen

A. Backup server and default certificates

B. Intermediate certificates and Domain certificates

C. Root or a Certificate Authority (CA) and SIP default certificates

D. Root or a Certificate Authority (CA) certificate and Server Identity certificates

E. Site Root certificates and Security certificates