Download Cisco.300-815.NewDumps.2021-03-11.60q.tqb

Vendor:	Cisco
Exam Code:	300-815
Exam Name:	Implementing Cisco Advanced Call Control and Mobility Services (CLASSM)
Date:	Mar 11, 2021
File Size:	2 MB

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Demo Questions

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.)

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

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?

The negotiated RTP port is outside of the range described by RFC, so inband DTMFs do not work.
There is SIP Delayed Offer. DTMF is supported only in Early Offer.
The rtpmap:0 value for the negotiated codec is marking DTMF as inactive.
No DTMF is negotiated.

Correct answer: D

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).