300-815: Implementing Cisco Advanced Call Control and Mobility Services (CLASSM)

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).

H.245 Terminal Capability Set
H.245 Open Logical Channel
H.225 Connect
H.245 Open Logical Channel Ack

Correct answer: B

Explanation:

Reference: http://ccievoicehopeful.blogspot.com/2012/09/h323-notes.html

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

DTMF
BFCP
VIDEO
FAX
AUDIO

Correct answer: AB

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

ALERTING
PROCEEDING
CONNECT
RINGING

Correct answer: C

End users at a new site report being unable to hear the remote party when calling or being called by users at headquarters. Calls to and from the PSTN work as expected. To investigate the SIP signaling to troubleshoot the problem, which field can provide a hint for troubleshooting?

Contact: header of the 200 OK response
Allow: header if the 200 OK response
o= line of SDP content
c= line of SDP content

Correct answer: C

Why would RTP traffic that is sent from the originating endpoint fail to be received on the far endpoint?

The far end connection data (c=) in the SDP was overwritten by deep packet inspection in the call signaling path.
Cisco Unified Communications Manager invoked media termination point resources.
The RTP traffic is arriving beyond the jitter buffer on the receiving end.
A firewall in the media path is blocking TCP ports 16384-32768.

Correct answer: D

An administrator is troubleshooting call failures on an H.323 gateway via the CLI. To see signaling for media and call setup, which debug must the Administrator turn on?

debug H.323 messages
debug H.225 asn1
debug H.246 asn 1
debug H.225 media
debug H.323 asn 1

Correct answer: B

What is first preference condition matched in a SIP-enabled incoming dial peer?

incoming uri
target carrier-id
answer-address
incoming called-number

Correct answer: A

Explanation:

Reference: https://www.cisco.com/c/en/us/support/docs/voice/ip-telephony-voice-over-ip-voip/211306-In-Depth-Explanation-of-Cisco-IOS-and-IO.html#anc8

Cisco SIP IP telephony is implemented on two floors of your company. Afterward, users report intermittent voice issues in calls established between floors. All calls are established, and sometimes they work well, but sometimes there is oneway audio or no audio. You determine that there is a firewall between the floors, and the administrator reports that it is allowing SIP signaling and UDP ports from 20000 to 22000 bidirectionally. What are two possible solutions? (Choose two.)

Go to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 16384-32767
Ask the firewall administrator to change the ports to TCP.
Ask the firewall administrator to change the range of UDP ports to 16384-32767.
Go to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 20000-22000.
Go to System Parameters in Cisco Unified Communications Manager and change the range of media ports to 20000-22000.

Correct answer: AC

Explanation:

Reference: https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/port/9_1_1/CUCM_BK_T2CA6EDE_00_tcp-port-usage-guide-91/CUCM_BK_T2CA6EDE_00_tcp-port-usage-guide-91_chapter_01.html

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

Download Implementing Cisco Advanced Call Control and Mobility Services (CLASSM).300-815.NewDumps.2021-03-11.60q.vcex

How to open VCEX files?

Demo Questions

Question 1

Question 2

Question 3

Question 4

Question 5

Question 6

Question 7

Question 8

Question 9

Question 10

ProfExam at a 20% markdown