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CCNP Collaboration Dumps 300-815 Exam for Full Questions - Exam Study Guide
NEW QUESTION # 54
Which configuration must an administrator perform to display Translation Pattern operations in Cisco Unified Communications Manager SDL traces?
- A. Check the Translation Patterns Analysis check box in Micro Traces on the Cisco Unified CM Serviceability page.
- B. Enable the Detailed Call Analysis option under Enterprise Parameters for Unified CM.
- C. By default, the Translation Patterns operations are printed in SDL traces, so no additional configuration is necessary.
- D. Set up the Digit Analysis Complexity in Service Parameters for Cisco Unified CM to TranslationAndAlternatePatternAnalysis.
Answer: B
Explanation:
Section: Call Control and Dial Planning
Explanation/Reference: https://community.cisco.com/t5/collaboration-voice-and-video/taking-sip-call-trace-on-cisco-unified- cm-using-rtmt/ta-p/3161200
NEW QUESTION # 55
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. c= line of SDP content
- B. Contact: header of the 200 OK response
- C. Allow: header if the 200 OK response
- D. o= line of SDP content
Answer: A
NEW QUESTION # 56
Refer to the exhibit.
A company needs to ensure that all calls are normalized to E164 format. Which configuration will ensure that the resulting digit string + 14085554001 is created and will be routed to the E.164 routing schema?
- A. Called Party Transformation Mask of + 1408555[35)XXX
- B. Calling Party Transformation Mask of +14085554XXX
- C. Called Party Transformation Mask of + 14085554XXX
- D. Calling Party Transformation Mask of +1408555XXXX
Answer: C
NEW QUESTION # 57
An engineer must route all SIP calls in the form of <user>@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.com
- B. example.com
- C. gateway.corporate.local
- D. *.*
- E. [email protected]
Answer: A,D
NEW QUESTION # 58
Which description of RTP timestamps or sequence numbers is true?
- A. The timestamp is used to place the incoming audio and video packets in the correct timing order (playout delay compensation).
- B. Timestamps increase by the time "carrying" by a packet.
- C. The sequence number is used to detect losses.
- D. Sequence numbers increase by four for each RTP packet transmitted.
Answer: A
NEW QUESTION # 59
In Cisco Unified Communications Manager globalized call routing is implemented and must confirm that it is correctly implemented without making a call. Which tool do you use for verification?
- A. SDI trace
- B. Dialed Number Analyzer
- C. SDL trace
- D. Real-Time Monitoring Tool
Answer: B
Explanation:
Section: Call Control and Dial Planning
NEW QUESTION # 60
Refer to the exhibits.
Regions have been configured for all major branches based on the available circuit bandwidth. Some calls from Region A endpoints to Region B endpoints are failing to connect. How is this issue resolved?
- A. Update all regions to 8 kbps maximum audio bitrate.
- B. Increase the number of available media termination points.
- C. Update the calling search space for affected endpoints to none.
- D. Add a media resource to transcode between available capabilities.
Answer: D
NEW QUESTION # 61
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?
- A. debug H.225 media
- B. debug H.323 asn 1
- C. debug H.323 messages
- D. debug H.246 asn 1
- E. debug H.224 asn1
Answer: E
NEW QUESTION # 62 
Refer to the exhibit. An engineer configures Cisco Unified Border Element to connect the enterprise VoIP network with a SIP telephony provider. Calls are not working in either direction. What must be configured in the dial peer 1 to fix the issue?
- A. codec g729
- B. answer-address 555 ........
- C. incoming called number 555.......
- D. session-protocol sipv2
Answer: C
Explanation:
Section: Call Control and Dial Planning
NEW QUESTION # 63
Refer to the exhibit.
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 direction but an incorrect voice translation rule applied, which is shown in the exhibit. Which rule modified DNIS in the format that the provider is expecting?
- A. rule 1 /^/+\([^1].*\)/ /011\1/
- B. rule 1/^\+1\([2-9]..[2-9]......$\)/ /\1/
- C. rule 1 /^\+1\([2-9]..[2-9]......$\)/ /\0/
- D. rule 1 /^\([2-9]..[2-9]......$\)/ /\1/
Answer: B
NEW QUESTION # 64
Users are reporting that several inter-site calls are failing, and the message "not enough bandwidth" is showing on the display. Voice traffic between locations goes through corporate WAN. and Call Admission Control is enabled to limit the number of calls between sites. How is the issue solved without increasing bandwidth utilization on the WAN links?
- A. Disable Call Admission Control and let the calls use the amount of bandwidth they require.
- B. Configure Call Queuing so that the user waits until there is bandwidth available
- C. Configure AAR to reroute calls that are denied by Call Admission Control through the PSTN.
- D. Reroute all calls through the PSTN and avoid using WAN.
Answer: C
NEW QUESTION # 65
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 rtpmap:0 value for the negotiated codec is marking DTMF as inactive.
- B. There is SIP Delayed Offer. DTMF is supported only in Early Offer.
- C. No DTMF is negotiated.
- D. The negotiated RTP port is outside of the range described by RFC, so inband DTMFs do not work.
Answer: C
NEW QUESTION # 66
Refer to the exhibit.
An administrator is troubleshooting a problem in which some outbound calls from an internal network to the Internet telephony service provider are not getting connected, but some others connect successfully. The firewall team found that some call attempts on port 5060 came from an unrecognized IP that has not been defined in the firewall rule. What should the administrator configure in the Cisco Unified Border Element to fix this issue?
- A. access list allowing the firewall IP
- B. use of port 5061 for SIP secure
- C. bind signaling and media to the loopback interface
- D. ip prefix-list to filter the unwanted IP address
Answer: C
NEW QUESTION # 67
What are two configureation features of the Client matter code setting in the route pattern configuration? (Choose two.)
- A. Selecting the Allow Overlap Sending setting allows a user to select the Require Client Matter Code setting.
- B. The Client Matter Code feature provides the option to configure Authorization Level susch as in the Forced Authorization Code.
- C. Selecting the Allow Overlap Sending setting disables the Require Client Matter Code setting.
- D. The client Matter Code feature supports overlap sending since the Cisco UCM can determine when to prompt the user for the code.
Answer: B,C
NEW QUESTION # 68
If all patterns below are configured in Cisco Unified Communications Manager which would be used when dialing the pattern "123"?
- A. 12X (urgent priority set)
- B. 1XX (urgent Priority Set)
- C. 12!
- D. 12[2-5]
Answer: A
Explanation:
Section: Call Control and Dial Planning
NEW QUESTION # 69
What is first preference condition matched in a SIP-enabled incoming dial peer?
- A. incoming called-number
- B. incoming uri
- C. answer-address
- D. target carrier-id
Answer: B
Explanation:
Section: Signaling and Media Protocols
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
NEW QUESTION # 70
An engineer must configure a Cisco UCM hunt list so that calls to users in a line group are routed to the first idle user and then the next. Which distribution algorithm must be configured to accomplish this task?
- A. circular
- B. top down
- C. broadcast
- D. longest idle time
Answer: B
NEW QUESTION # 71
Where on Cisco Unified Communications Manager do you configure the standard local route group for a group of devices?
- A. System > Location Info
- B. System > Device Pool
- C. Call Routing > Emergency Location > Emergency Location (ELIN) Groups
- D. Call Routing > Route/Hunt > Local Route Group Names
Answer: D
NEW QUESTION # 72
Which call pickup feature allows users to pick up incoming calls in a group that is associated with their own group?
- A. BLF Call Pickup
- B. Group Call Pickup
- C. Directed Call Pickup
- D. Other Group Pickup
Answer: D
NEW QUESTION # 73
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To succeed in the Cisco 300-815 exam, candidates must have practical experience in implementing and managing Cisco Unified Communications Manager, Cisco Unity Connection, and Cisco Unity Express. They must also have a solid understanding of voice and video protocols, such as Session Initiation Protocol (SIP), Real-Time Transport Protocol (RTP), and Media Gateway Control Protocol (MGCP). With this certification, candidates can demonstrate their expertise in advanced call control and mobility services, which can lead to better job opportunities and higher salaries.
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