Implementing Cisco Advanced Call Control and Mobility Services (CLASSM) Exam Questions

Here you have the best Cisco 300-815 practice exam questions.

Some things you may want to keep in mind about this practice exam questions:

You have 167 total questions to study from
Each page has 5 questions, making a total of 34 pages
You can navigate through the pages using the buttons at the bottom
This questions were last updated on September 19, 2024

Question 1 of 167

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.)

A.

Phone_A sends a SIP-REFER message to the Cisco Unified Communications Manager with Phone_C information in the Refer-To section.

B.

Phone_B sends a SIP-REFER message to the Cisco Unified CM with Phone_C information in the Refer-To section.

C.

As soon as Phone_A presses the Transfer button for the first time, Phone_B hears the MOH and the MOH audio is chosen from Phone_B User Hold MOH Audio Source settings.

D.

As soon as Phone_A presses the Transfer button for the first time, Phone_B hears the music on hold and the MOH audio is chosen from Phone_A Network Hold MOH Audio Source settings.

E.

As soon as Phone_A presses the Transfer button for the first time, Phone_B hears the MOH and the MOH audio is chosen from Phone_A User Hold MOH Audio Source settings.

Correct Answer: AD

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.

Question 2 of 167

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-band

DTMF is supported, what is a reason for this malfunction?

A.

The negotiated RTP port is outside of the range described by RFC, so inband DTMFs do not work.

B.

There is SIP Delayed Offer. DTMF is supported only in Early Offer.

C.

The rtpmap:0 value for the negotiated codec is marking DTMF as inactive.

D.

No DTMF is negotiated.

Correct Answer: 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 167

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).

A.

H.245 Terminal Capability Set

B.

H.245 Open Logical Channel

C.

H.225 Connect

D.

H.245 Open Logical Channel Ack

Correct Answer: 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.

Question 4 of 167

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

A.

DTMF

B.

BFCP

C.

VIDEO

D.

FAX

E.

AUDIO

Correct Answer: BE

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.

Question 5 of 167

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

A.

ALERTING

B.

PROCEEDING

C.

CONNECT

D.

RINGING

Correct Answer: A

When troubleshooting H.323 call setup, the 'ALERTING' message informs you that the called party is being notified about the call. This message is sent once the call setup process has reached the called party, indicating that the remote endpoint has initiated the process to alert the user, usually by ringing the phone.