Implementing Cisco Advanced Call Control and Mobility Services (CLASSM)

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Question 1 of 245

Exam 300-815: Question 1 - Image 1

Refer to the exhibit. In an active SIP call between phone user A and phone user B, phone A initiates a call transfer to phone user C. Which two scenarios are correct? (Choose two.)

Answer

Suggested Answer

The suggested answer is A, D.

During a SIP call transfer, the transferring phone (Phone A) sends a SIP-REFER message to the Call Manager with the information for the transfer target (Phone C). Therefore, Phone A sends a SIP-REFER message. Secondly, when Phone A initiates the transfer by pressing the Transfer button, Phone B will hear Music on Hold (MOH). The MOH audio source is typically determined by the network hold settings of the transferring phone, which in this case is Phone A. Therefore, the correct scenarios in the exhibit are that Phone A sends the SIP-REFER message to the Call Manager, and Phone B hears the MOH chosen from Phone A's Network Hold MOH Audio Source settings.

Community Votes4 votes
ADSuggested
75%
AC
25%
Question 2 of 245

Exam 300-815: Question 2 - Image 1

Refer to the exhibit. Users report that when they dial to Cisco Unity Connection from an external network, they cannot enter any digits. Assuming only in-bandDTMF is supported, what is a reason for this malfunction?

Answer

Suggested Answer

The suggested answer is D.

No DTMF is negotiated. The exhibit shows the SDP (Session Description Protocol) information for the SIP call setup. For DTMF to work in-band, it's essential that the DTMF tones are negotiated and supported. In this case, the RTP payload types 18 for the codec and 110 for DTMF are listed, but there is no evidence showing an actual negotiation that ensures the DTMF tones will transmit correctly over the network. Therefore, the issue arises due to the lack of DTMF negotiation.

Question 3 of 245

The administrator of ABC company is troubleshooting a one-way audio issue for a call that uses H.323 protocol (slow-start mode). The administrator requests that you provide the IP and port information of the Real-Time Transport Protocol traffic that had the one-way audio call.

You gather the H.225 and H.245 messages for one of the one-way audio calls. Where can you find the RTP IP and port information for both sides? (Note: This call flow has not invoked any media resources like MTP or transcoders).

Answer

Suggested Answer

The suggested answer is D.

To resolve a one-way audio issue in an H.323 protocol call using slow-start mode, it's essential to locate the Real-Time Transport Protocol (RTP) IP and port information. This information can be found in the H.245 Open Logical Channel Acknowledgment (Ack) message. The H.245 Open Logical Channel message initially requests the opening of a communication channel, but it is in the H.245 Open Logical Channel Ack where the endpoint confirms and provides the RTP IP and port details that will be used for the media. Therefore, the correct option is H.245 Open Logical Channel Ack.

Community Votes1 vote
DSuggested
100%
Question 4 of 245

Which two extended capabilities must be configured on dial peers for fast start-to-early media scenarios (H.323 to SIP interworking)? (Choose two.)

Answer

Suggested Answer

The suggested answer is B, E.

For fast start-to-early media scenarios in H.323 to SIP interworking, BFCP (Binary Floor Control Protocol) and AUDIO capabilities must be configured on dial peers. BFCP is needed for content control during the media session, and AUDIO allows for early media transmission, enabling users to hear audio before the call is fully established. These are essential for optimizing call setup times and ensuring a smooth user experience.

Community Votes5 votes
ADMost voted
40%
AC
20%
CE
20%
AE
20%
Question 5 of 245

When you troubleshoot H.323 call setup, which message informs you that the called party is being notified about the call?

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