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CCNP 300-815 Certification Exam Questions & Practice Tests, Cisco Collaboration 300-815 CLACCM | SPOTO

Prepare thoroughly for the CCNP 300-815 Certification Exam in Cisco Collaboration (CLACCM 300-815) with SPOTO's exam questions and practice tests. This 90-minute exam, essential for CCNP Collaboration Certification, evaluates your expertise in advanced call control and mobility services. Key topics include signaling and media protocols, CME/SRST gateway technologies, Cisco Unified Board Element, call control and dial planning, Cisco Unified CM Call Control, and mobility solutions. The course 'Implementing Cisco Advanced Call Control and Mobility Services' is designed to support candidates in their exam preparation journey, offering essential insights and strategies. Access our practice tests, exam dumps, and exam simulators to enhance your readiness and excel in demonstrating your proficiency in advanced call control and mobility services during the CCNP 300-815 exam.

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Question #1
Topic 1An 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
B. H
C. H
D. H
E. H
View answer
Correct Answer: BC
Question #2
Topic 1What 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)
View answer
Correct Answer: D
Question #3
An engineer is troubleshooting local ringback on a Cisco SIP gateway The gateway is not ignoring the SIP 180 response with SDP from the service provider, and the far end device is sending the 180 with SDP to play ringback from the IP address specified m the SDP Which configuration change must be made on the gateway to resolve the issue?
A. outer(conf-voi-serv)# dlisable-early-media 180
B. outer(conftg-sip-ua)# disable-early-media 180
C. outer(con(-voi-serv)# no disable-early-media 180
D. outer(config-sip-ua)# no disable-early-media 180
View answer
Correct Answer: B
Question #4
Topic 1What 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
View answer
Correct Answer: A
Question #5
An engineer has temporarily disabled toll fraud prevention for SIP line calls on a Cisco CME12.6x and must enforce security and toll fraud prevention for the SIP line side on Cisco Unified CME. Which configuration must be used to start this process?
A. oice service volp Ip address trusted list
B. oice service volp enablo ip address trust authentication
C. oice service volp enable Ip address trust list
D. oice service volp ip address trusted authenticate
View answer
Correct Answer: D
Question #6
Topic 1Refer to the exhibit. A user reports that when they call a specific phone number, no one answers the call, but when they callfrom a mobile phone, the call is answered. The engineer troubleshooting the issue is expecting the far-end gateway to cutthrough audio on the 183 Session Progress SIP message. Which SIP Profile configuration element is necessary for theCisco 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
View answer
Correct Answer: C
Question #7
Topic 1Which two extended capabilities must be configured on dial peers for fast start-to-early media scenarios (H.323 to SIPinterworking)? (Choose two.)
A. DTMF
B. BFCP
C. VIDEO
D. FAX
E. AUDIO
View answer
Correct Answer: AB
Question #8
Topic 1When an administrator troubleshoots H.323 call setup, which message gives an alert that the called party is being notifiedabout the call?
A. ALERTING
B. PROCEEDING
C. CONNECT
D. RINGING
View answer
Correct Answer: C
Question #9
Topic 1End 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 canprovide 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
View answer
Correct Answer: C
Question #10
Topic 1Refer to the exhibit. In an active SIP call between phone user A and phone user B, phone A initiates a call transfer to phoneuserC. 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
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
View answer
Correct Answer: AC
Question #11
Topic 1Refer to the exhibit. While troubleshooting call failures on the Cisco Unified Border Element, an administrator notices thatmessages are being sent to the service provide, but there is no response. The administrator later learns that this SIPprovider 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
View answer
Correct Answer: A
Question #12
Topic 1Why 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
View answer
Correct Answer: D
Question #13
Topic 1An administrator is troubleshooting a one-way audio issue for a call that uses H.323 protocol in slow-start mode. Theadministrator requests that the IP and port information of the Real-Time Transport Protocol traffic that had the one-way audiocall is provided. The H.225 and H.245 messages for one of the one-way audio calls are gathered and the call flow has notinvoked any media resources. Where is the RTP IP and port information for both sides found?
A. H
B. H
C. H
D. H
View answer
Correct Answer: B
Question #14
The SIP session refresh timer allows the RTP session to stay active during an active call. The Cisco UCM sends either SIP-INVITE or SIP-UPDATE messages in a regular interval of time throughout the active duration of the call. During a troubleshooting session, the engineer finds that the Cisco UCM is sending SIP-UPDATE as the SIP session refresher, and the engineer would like to use SIP-INVITE as the session refresher. What configuration should be made in the Cisco UCM to achieve this?
A. nable SIP ReMXX Options on the SIP profile
B. nable Send send-receive SDP in mid-call INVITE on the SIP profile
C. hange Session Refresh Method on the SIP profile to INVITE
D. ncrease Retry INVITE to 20 seconds on the SIP profile
View answer
Correct Answer: C
Question #15
Topic 1Refer to the exhibit. Users report that when they dial to Cisco Unity Connection from an external network, they cannot enterany 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
C. The rtpmap:0 value for the negotiated codec is marking DTMF as inactive
D. No DTMF is negotiated
View answer
Correct Answer: D
Question #16
Topic 1A support engineer is troubleshooting a voice network. When conducting a search for call setup details related to callingsearch space issues, which trace files should be investigated?
A. CallManager traces
B. CTI Manager traces
C. Cisco IP Manager Assistant
D. Call logs
View answer
Correct Answer: A
Question #17
Refer to the exhibit. A Cisco Unified Border Element continues to send 180/183 with the required: 100rel header to Cisco UCM. and the call eventually disconnects How is the issue resolved?
A. nable 'SIP ReI1XX Options* and -Early Offer Support' on the SIP Profile Configuration Page in Cisco UCM
B. nable *Early Offer support for voice and video calls' on the SIP Profile Configuration Page in Cisco UCM
C. isable 'SIP Rel1XX Options* and 'Early Offer Support* on the SIP Profile Configuration Page m Cisco UCM
D. isable 'Send send-receive SDP in mid-call INVITE* on the SIP Profile Configuration Page in Cisco UCM
View answer
Correct Answer: B
Question #18
Topic 1Cisco SIP IP telephony is implemented on two floors of your company. Afterward, users report intermittent voice issues incalls established between floors. All calls are established, and sometimes they work well, but sometimes there is one-wayaudio or no audio. It is determined that there is a firewall between the floors, and the administrator reports that it is allowingSIP 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
View answer
Correct Answer: AC
Question #19
Topic 1Which 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
View answer
Correct Answer: C
Question #20
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
View answer
Correct Answer: AC

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