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Master CCNP 300-815 Exams with Exam Questions & Study Materials, Cisco Collaboration 300-815 CLACCM | SPOTO

Prepare comprehensively for the CCNP 300-815 Certification Exam in Cisco Collaboration (CLACCM 300-815) with SPOTO's exam questions and study materials. This 90-minute exam, crucial for CCNP Collaboration Certification, evaluates your understanding of advanced call control and mobility services. Key areas 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 a vital resource for exam preparation, offering in-depth coverage of exam topics 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
Where is the dtmf-relay command configured on Cisco Unified Border Element?
A. in the voice-class VoIP configuration
B. in the VoIP dial peer
C. in global SIP configuration
D. in the VoIP or POTS dial peers
View answer
Correct Answer: AC

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Question #2
When you troubleshoot H.323 call setup, which message informs you that the called party is being notified about the call?
A. ALERTING
B. PROCEEDING
C. CONNECT
D. RINGING
View answer
Correct Answer: C
Question #3
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
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 #4
A user in location X dials an extension at location Y. The call travels through a QoS-enabled WAN network, but the user experiences choppy or clipped audio. What is the cause of this issue?
A. missing Call Admission Control
B. codec mismatch
C. ptime mismatch
D. phone class of service issue
View answer
Correct Answer: A
Question #5
Which two statements are correct with respect to the Client Matter Code setting in the route pattern configuration? (Choose two.)
A. The Client Matter Code feature does not support overlap sending because the Cisco Unified CM cannot determine when to prompt the user for the code
B. If you check the Allow Overlap Sending check box, the Require Client Matter Code check box becomes disabled
C. If you check the Allow Overlap Sending check box, you can also check the Require Client Matter Code check box
D. The Client Matter Code feature does support overlap sending because the Cisco Unified Communications Manager can determine when to prompt the user for the code
E. The Client Matter Code has the option to configure Authorization Level such as in the Forced Authorization Code
View answer
Correct Answer: B
Question #6
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).
A. H
B. H
C. H
D. H
View answer
Correct Answer: B
Question #7
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
View answer
Correct Answer: A
Question #8
An engineer must route all SIP calls in the form of @example.com to the SIP trunk gateway corporate local. Which two SIP route patterns can be used to accomplish this task? (Choose two.)
A. example
B. *@example
C. gateway
D. example
E. *
View answer
Correct Answer: B
Question #9
You see the voice register pool 1 command in your Cisco Unified Communications Manager Express configuration. Which configuration is occurring in this section?
A. configuration for a single SIP phone
B. configuration items common for all SIP phones
C. configuration for a pool of SIP phones (similar to device pool on Cisco Unified Communications Manager)D
View answer
Correct Answer: A
Question #10
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
View answer
Correct Answer: C
Question #11
Calls incoming from the provider are not working through newly set up Cisco Unified Border Element. Provider engineers get the 404 Not Found SIP message. Incoming calls are coming from the provider with called number “222333444” and Cisco Unified Communications Manager is expecting the called number to be delivered as “444333222”. The administrator already verified that the IP address of the Cisco Unified CM is set up correctly and there are no dial peers configured other than those shown in the exhibit. Wh
A. Change the destination-pattern on the outgoing dial peer to match “444333222”
B. Set up translation-profile on the incoming dial peer to match incoming traffic
C. Create specific matching for “222333444” on the incoming dial peer
D. Fix the voice translation-rule to match specifically number “222333444” and change it to “444333222”
View answer
Correct Answer: B
Question #12
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 Unified Communications Manager 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
A company has an SRST gateway running an IOS XE image. The company plans to enable the IPv6 addressing companywide. To enable the IPv6 in a unified SRST gateway to support SIP phones, what are two supported supplementary features for an IPv6 fallback scenario? (Choose two.)
A. three-way conference
B. secure SIP lines
C. T
D. transcoding
E. SIP trunk
View answer
Correct Answer: C
Question #14
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
View answer
Correct Answer: AC
Question #15
Which action is correct with respect to toll fraud prevention configuration in the Cisco Unified Communications Manager Express?
A. Configure Direct Inward Dial for Incoming ISDN Calls with overlap dialing
B. Configure IP Address Trusted Authentication for Incoming VoIP Calls
C. Configure the command no ip address trusted authenticate under “voice service voip”
D. Enable Secondary Dial tone on Analog and Digital FXO Ports
View answer
Correct Answer: D
Question #16
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
View answer
Correct Answer: AB
Question #17
Which top-level IOS command is needed to begin the configuration of a Cisco Unified Communications Manager Express gateway to enable phones to be registered via SIP?
A. allow-connections sip to sip
B. voice service voip
C. voice register global
D. voice register dn
View answer
Correct Answer: C
Question #18
Users report that outbound PSTN calls from phones registered to Cisco Unified Communications Manager are not completing. The local service provider in North America has a requirement to receive calls in 10-digit format. The Cisco Unified CM sends the calls to the Cisco Unified Border Element router in a globalized E.164 format. There is an outbound dial peer on Cisco Unified Border Element configured to send the calls to the provider. The dial peer has a voice translation profile applied in the correct dire
A. rule 1 /^/+\([^1]
B. rule 1/^\+1\([2-9]
C. rule 1 /^\([2-9]
D. rule 1 /^\+1\([2-9]
View answer
Correct Answer: C
Question #19
An administrator is troubleshooting why users are not hearing audio when dialing long distance numbers across their Cisco Unified Border Element. The customer’s carrier has a requirement that dialing long distance requires an access code to be entered. Looking at the exhibit, what two actions can be taken to correct signaling? (Choose two.)
A. Enanle PRACK
B. Enable Early Offer on the Cisco Unified Border Element
C. Enable the supplementary-service media-renegotiate command
D. Enable Media Flow Around
E. Enable Mid-Call Signaling Consumption
View answer
Correct Answer: C
Question #20
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.)
A. Go to the SIP profile assigned to these IP phones in Cisco Unified CM 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 Unified CM and change the range of media ports to 20000-22000
E. Go to System Parameters in Cisco Unified Communications Manager and change the range of media ports to 20000-22000
View answer
Correct Answer: B

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