3313 Avaya Aura Contact Center Maintenance and Troubleshooting Exam

Loading demo links...

Showing 4–6 of 10 questions

Question 4

Avaya Aura® Contact Center supports implementing Secure Real-Time Transport Protocol (SRTP) for voice contacts within the Contact Center. SRTP is an extension to the Real-Time Transport Protocol (RTP) to support secure real-time communications. The primary use of SRTP is to encrypt and authenticate voice over IP on the network.

Before implementing SRTP in Contact Center, you must have TLS on which three links? (Choose three.)

Select all that apply, then click Submit answer.

  • Agent telephones to Communication Manager (CM)

  • Communication Manager (CM) to Contact Center

  • Session Manager (ASM) to Contact Center

  • Contact Center to Avaya Aura® Media Server (AAMS)


Question 5

If announcements are not being played to callers, which troubleshooting steps will you perform? (Choose three.)

Select all that apply, then click Submit answer.

  • From CCMA > Contact Management, ensure that the treatment address includes the correct SIP context for the ANMC, CONF and DIALOG services.

  • Confirm recordings have been uploaded to the CCMS.

  • From CCMA > Configuration > Media services and Routes, ensure that the treatment address includes the correct SIP context for the ANNC, CONF and DIALOG services.

  • Verify that each Media Server (AAMS) is associated with a least one Target Media Server (AAMS).

  • Verify that each Media server (AAMS) is associated with a least one target Media server (AAMS).


Question 6

In a SIP-enabled Avaya Aura® Contact Center (AACC) deployment, a typical incoming call goes through the following sequence of steps:

1. The incoming call arrives at the switch.

2. The switch routes the call to the Contact Center Manager Server (CCMS) based on the routing plan.

What is the next step in the sequence?

Select an option, then click Submit answer.

  • The SIP Gateway Manager suspends the call. No audio path is established until the call is answered by an agent.

  • The call is redirected to a SIP URI on the Session Manager and an H.323 session is established.

  • The call is answered by the SIP Gateway Manager and a Real- Time Transport protocol (RTP) session is established.

  • The CCMS anchors the call on an Avaya Aura® Media server conference port.