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| Type | Number of exam questions | Exam name | Exam code |
| Free | 15 | Implementing Cisco Advanced Call Control and Mobility Services (CLACCM) | 300-815 |
Question 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 the two results of this action? (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
User hold: With a user hold, a user presses the Hold button on a phone to explicitly place the caller on hold. If Phone A and Phone B are having a conversation, and the user of Phone B presses the Hold button, Phone B\’s user hold source is
streamed to Phone A.
Network hold: A network hold occurs when a call is placed on hold as part of the processing of a supplementary service, such as a park, transfer, or conference. If Phone B presses the Transfer button to transfer a call, Phone A still gets placed
on hold but hears Phone B\’s network audio source while the rest of the transfer operations are complete.
Question 2:
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 the 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
Question 3:
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
Correct Answer: D
Reference: http://ccievoicehopeful.blogspot.com/2012/09/h323-notes.html
Question 4:
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: AB
Question 5:
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
Correct Answer: A
Question 6:
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
Correct Answer: D
Question 7:
Why would RTP traffic that is sent from the originating endpoint fails 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.
Correct Answer: A
Question 8:
An administrator is troubleshooting call failures on an H.323 gateway via the CLI. To see signaling for media and call setup, which two debugs should the Administrator turn on? (Choose two.)
A. debug H.245 asn1
B. debug H.323 message
C. debug H.225 asn1
D. debug H.225 media
E. debug H.323 asn1
Correct Answer: AC
Question 9:
What is the 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
Correct Answer: A
Question 10:
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 20000-22000.
Correct Answer: CD
This question is tricky because it\’s asking for two separate solutions. The wording suggests that it\’s not looking for two tasks to become one single solution. In this case, one solution is to change the ports on the SIP profile of the phones to match the ports on the firewall. The other solution is to ask the firewall admin to change his ports to match the ports in UCM.
Question 11:
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
Correct Answer: C
Question 12:
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).
Correct Answer: D
Reference: https://www.cs.columbia.edu/~hgs/rtp/faq.html
Question 13:
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
Correct Answer: A
Question 14:
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 an acknowledgment 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.
Correct Answer: C
Question 15:
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
Correct Answer: A
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