cisco 300-815 practice test
Implementing Cisco Advanced Call Control and Mobility Services (CLACCM)
Question 1 Topic 1
Topic 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. What are two results from this action? (Choose two.)
- A. Phone_A sends a SIP-REFER message to the Cisco UCM with Phone_C information in the Refer-To section.
- B. Phone_B sends a SIP-REFER message to the Cisco UCM with Phone_C information in the Refer-To section.
- C. What are two results from this action? (Choose two.) 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.
Answer:
A C
Question 2 Topic 1
Topic 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-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.
Answer:
D
Question 3 Topic 1
Topic 1
An administrator is troubleshooting a one-way audio issue for a call that uses H.323 protocol in slow-start mode. The
administrator requests that the IP and port information of the Real-Time Transport Protocol traffic that had the one-way audio
call is provided. The H.225 and H.245 messages for one of the one-way audio calls are gathered and the call flow has not
invoked any media resources. Where is the RTP IP and port information for both sides found?
- A. H.245 Terminal Capability Set
- B. H.245 Open Logical Channel
- C. H.225 Connect
- D. H.245 Open Logical Channel Ack
Answer:
B
Explanation:
Reference: http://ccievoicehopeful.blogspot.com/2012/09/h323-notes.html
Question 4 Topic 1
Topic 1
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
Answer:
A B
Question 5 Topic 1
Topic 1
When an administrator troubleshoots H.323 call setup, which message gives an alert that the called party is being notified
about the call?
- A. ALERTING
- B. PROCEEDING
- C. CONNECT
- D. RINGING
Answer:
C
Question 6 Topic 1
Topic 1
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?
- A. Contact: header of the 200 OK response
- B. Allow: header if the 200 OK response
- C. o= line of SDP content
- D. c= line of SDP content
Answer:
C
Question 7 Topic 1
Topic 1
Why would RTP traffic that is sent from the originating endpoint fail to be received on the far endpoint?
- A. The far end connection data (c=) in the SDP was overwritten by deep packet inspection in the call signaling path.
- B. Cisco UCM invoked media termination point resources.
- C. The RTP traffic is arriving beyond the jitter buffer on the receiving end.
- D. A firewall in the media path is blocking TCP ports 16384-32768.
Answer:
D
Question 8 Topic 1
Topic 1
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? (Choose two.)
- A. H.323 messages
- B. H.225 asn1
- C. H.246 asn 1
- D. H.225 media
- E. H.323 asn 1
Answer:
B C
Question 9 Topic 1
Topic 1
What is first preference condition matched in a SIP-enabled incoming dial peer?
- A. incoming uri
- B. target carrier-id
- C. answer-address
- D. incoming called-number
Answer:
A
Explanation:
Reference: https://www.cisco.com/c/en/us/support/docs/voice/ip-telephony-voice-over-ipvoip/211306-In-Depth-
Question 10 Topic 1
Topic 1
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 one-way
audio or no audio. It is determined 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 solutions for this issue?
(Choose two.)
- A. Go to the SIP profile assigned to these IP phones in Cisco UCM and change the range of media ports to 16384-32767
- B. Ask the firewall administrator to change the ports to TCP.
- C. Ask the firewall administrator to change the range of UDP ports to 16384-32767.
- D. Go to the SIP profile assigned to these IP phones in Cisco UCM and change the range of media ports to 20000-22000.
- E. Go to System Parameters in Cisco UCM and change the range of media ports to 2000022000.
Answer:
A C
Question 11 Topic 1
Topic 1
Which section under the Real-Time Monitoring Tool allows for reviewing the call flow and signaling for a SIP call in real time?
- A. Analysis Manager > Inventory > Trace File Repositories
- B. System > Tools > Trace and Log Central
- C. Voice/Video > Session Trace Log View > Real Time Data
- D. Voice/Video > Session Trace Log View > Open From Local Disk
Answer:
C
Explanation:
Reference: https://www.cisco.com/c/en/us/support/docs/unified-communications/unifiedcommunications-manager-
callmanager/213583-procedure-to-analyse-call-flow-of-sip-ca.html
Question 12 Topic 1
Topic 1
What is a description of RTP timestamps or sequence numbers?
- A. The sequence number is used to detect losses.
- B. Timestamps increase by the time “carrying” by a packet.
- C. Sequence numbers increase by four for each RTP packet transmitted.
- D. The timestamp is used to place the incoming audio and video packets in the correct timing order (playout delay compensation).
Answer:
D
Explanation:
Reference: https://www.cs.columbia.edu/~hgs/rtp/faq.html
Question 13 Topic 1
Topic 1
A support engineer is troubleshooting a voice network. When conducting a search for call setup details related to calling
search space issues, which trace files should be investigated?
- A. CallManager traces
- B. CTI Manager traces
- C. Cisco IP Manager Assistant
- D. Call logs
Answer:
A
Question 14 Topic 1
Topic 1
Refer to the exhibit. A user reports that when they call a specific phone number, no one answers the call, but when they call
from a mobile phone, the call is answered. The engineer troubleshooting the issue is expecting the far-end gateway to cut
through audio on the 183 Session Progress SIP message. Which SIP Profile configuration element is necessary for the
Cisco Unified Communications Manager to send acknowledgement of provisional responses?
- A. Allow Passthrough of Configured Line Device Caller Information must be enabled.
- B. Accept Audio Codec Preferences in Received Offer must be set to On.
- C. On the SIP Profile, the configuration parameter SIP Rel1XX Options must be set to Send PRACK for all 1xx Messages.
- D. Early Offer for G Clear Calls must be enabled.
Answer:
C
Question 15 Topic 1
Topic 1
Refer to the exhibit. While troubleshooting call failures on the Cisco Unified Border Element, an administrator notices that
messages are being sent to the service provide, but there is no response. The administrator later learns that this SIP
provider does not support PRACK. Which header should be removed from the SIP message to resolve this issue?
- A. Require: 100rel
- B. Content-Type: application/sdp
- C. Contact:
- D. Content-Disposition: session;handling=required
Answer:
A