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Cisco 300-815 exam is designed to test the knowledge and skills of IT professionals in implementing advanced call control and mobility services using Cisco technologies. Implementing Cisco Advanced Call Control and Mobility Services certification exam is part of the Cisco Certified DevNet Professional program, which focuses on developing skills in automation, collaboration, security, and infrastructure. 300-815 exam is aimed at network engineers, network architects, and solutions architects who work with Cisco technologies in enterprise environments.
NEW QUESTION # 55
Refer to the exhibit.
DN 1003 was the last to ring during the most recent call. Which hunting method ensures that DN 1005 is presented with the next call when the hunt pilot is dialed?
- A. sequential
- B. call-blast
- C. peer
- D. parallel
Answer: C
NEW QUESTION # 56
Refer to the exhibit.
How many maximum hops can an ILS updarte traverse?
- A. 0
- B. 1
- C. 2
- D. 3
Answer: C
NEW QUESTION # 57
Refer to the exhibit.
A user dials 84969010 and observes that the call is not routed immediately. The administrator notices that after matching the fixed-length translation pattern, the call hits the \+! pattern and waits for interdigit timeout What should be configured to ensure that the call routes out immediately?
- A. Route Next Hop By Calling Party Number on the translation pattern
- B. Allow Device Override on the route pattern
- C. Do Not Wait For Interdigit Timeout On Subsequent Hops on the route pattern
- D. Do Not Wait For Interdigit Timeout On Subsequent Hops on the translation pattern
Answer: D
NEW QUESTION # 58
An administrator is configuring a cluster for ILS and wants to limit the amount of entities that Cisco Unified Communications Manager can write to the database for data that is learned through ILS. Which service parameter is used to adjust this limit?
- A. ILS Max Number of Learned Objects in Database
- B. ILS Active Learned Object Upper Limit
- C. Imported Dial Plan Replication Database Object Lower Limit
- D. Global Data Service Parameter Limit
Answer: A
NEW QUESTION # 59
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. Enable Send send-receive SDP in mid-call INVITE on the SIP profile.
- B. Enable SIP ReMXX Options on the SIP profile.
- C. Increase Retry INVITE to 20 seconds on the SIP profile.
- D. Change Session Refresh Method on the SIP profile to INVITE.
Answer: D
NEW QUESTION # 60
A single site reports that when they dial select numbers, the call connects, but they do not get audio. The administrator finds that the calls are not routing out of the normal gateway but out of another site's gateway due to a TEHO configuration. What is the next step to diagnose and solve the issue?
- A. Verify that IP routing is correct between the gateway and the IP phone.
- B. Verify that the route pattern is not blocking calls to the destination number.
- C. Verify that the route pattern has the correct calling-party transformation mask
- D. Verify that the dial peer of the gateway has the correct destination pattern configured.
Answer: D
NEW QUESTION # 61
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. incoming called number 555.......
- C. answer-address 555 ........
- D. session-protocol sipv2
Answer: B
NEW QUESTION # 62
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 dn
- D. voice register global
Answer: D
Explanation:
Reference:
https://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified- communications-manager-express/99946-cme-sip-guide.html
NEW QUESTION # 63
Refer to the exhibit.
An administrator is troubleshooting a situation where a call placed from a phone registered to Cisco Unified Communications Manager does not complete. The administrator wants to use the Dialed Number Analyzer on Cisco Unified CM to check which translation pattern the call is matching. However, when logging in to Cisco Unified Serviceability there is no option for Dialed Number Analyzer under the tool menu. Which two steps must be performed to resolve this issue? (Choose two.)
- A. Restart the subscriber
- B. Activate the Cisco Extended Functions service.
- C. Activate the Cisco Dialed Number Analyzer service.
- D. Activate the Cisco Dialed Number Analyzer Server service.
- E. Activate the Cisco CallManager service.
Answer: C,D
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. Reroute all calls through the PSTN and avoid using WAN.
- B. Configure Call Queuing so that the user waits until there is bandwidth available
- C. Disable Call Admission Control and let the calls use the amount of bandwidth they require.
- D. Configure AAR to reroute calls that are denied by Call Admission Control through the PSTN.
Answer: D
NEW QUESTION # 65
What is first preference condition matched in a SIP-enabled incoming dial peer?
- A. target carrier-id
- B. incoming uri
- C. incoming called-number
- D. answer-address
Answer: B
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 # 66
An administrator is trying to apply configuration changes on Cisco CME. When the users registered on Cisco CME to dial a local number to a PSTN call, the Cisco CME sends an incorrect number of digits. What translation rule fixes the issue and sends the correct number of digits?
- A. voice translation-rule 1
rule 1 /^4...V /2404\0/ type any subscriber plan any isdn - B. voice translation-rule 1 rule 1 // // type any subscriber plan any isdn
- C. voice translation-rule 1
rule 1 /^4...$/2404\0/ type any national plan any Isdn - D. voice translation-rule 1 rule 1 /^4...S/ /9132404 0/ type any subscriber plan any Isdn
Answer: A
NEW QUESTION # 67
Which two descriptions of the Standard Local Route Group deployment are true? (Choose two.)
- A. can be assigned directly to the route pattern
- B. can be associated only under the route list
- C. chooses the route group that is configured under the device pool of the called-party device
- D. can be associated under the route group
- E. chooses the route group that is configured under the device pool of the calling-party device
Answer: B,C
Explanation:
Section: Call Control and Dial Planning
NEW QUESTION # 68
How does an engineer globalize routing for ingress calls coming from the PSTN to internal DNs?
- A. At Cisco Unified CM, put the calling number in E.164 format and the called number in PSTN format.
- B. At Cisco Unified Communications Manager, put the calling number in E.164 format and the called number in E.164 format.
- C. At the PSTN gateway, put the calling number in E.164 format and the called number in localized (DN) format.
- D. At the PSTN gateway, put the calling number in PSTN format and the called number in DN format.
Answer: A
NEW QUESTION # 69
An engineer must configure a secure SIP trunk with a remote provider, with a specific requirement to use port
5065 for inbound and otubound traffic. Which two items must be configured to complete this configuration? (Choose two.)
- A. Destination Port in SIP Trunk Security Profile configuration
- B. Destination Port in SIP Information section of the SIP Trunk configuration
- C. Incoming Port in SIP Information section of the SIP Trunk configuration.
- D. Incoming Port in SIP Trunk Security Profile configuration
- E. Incoming Port in Security Information of the SIP Profile configuration.
Answer: B,D
NEW QUESTION # 70
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. 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 20000-22000.
- B. Ask the firewall administrator to change the range of UDP ports to 16384-32767.
- C. Go to System Parameters in Cisco Unified Communications Manager and change the range of media ports to 20000-22000.
- D. Go to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 16384-32767
- E. Ask the firewall administrator to change the ports to TCP.
Answer: B,D
Explanation:
Section: Signaling and Media Protocols
Explanation/Reference: https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/port/9_1_1/ CUCM_BK_T2CA6EDE_00_tcp-port-usage-guide-91/CUCM_BK_T2CA6EDE_00_tcp-port-usage-guide-
91_chapter_01.html
NEW QUESTION # 71
Where on Cisco Unified Communications Manager do you configure the standard local route group for a group of devices?
- A. Call Routing > Emergency Location > Emergency Location (ELIN) Groups
- B. System > Location Info
- C. System > Device Pool
- D. Call Routing > Route/Hunt > Local Route Group Names
Answer: D
Explanation:
Reference:
https://www.uccollabing.com/configuring-standard-local-route-group-cucm/
NEW QUESTION # 72
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.245 Open Logical Channel
- B. H.225 Connect
- C. H.245 Terminal Capability Set
- D. H.245 Open Logical Channel Ack
Answer: A
NEW QUESTION # 73 
Refer to the exhibit. 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. Enable Mid-Call Signaling Consumption.
- B. Enable the supplementary-service media-renegotiate command.
- C. Enanle PRACK.
- D. Enable Early Offer on the Cisco Unified Border Element.
- E. Enable Media Flow Around
Answer: C,D
Explanation:
Section: Cisco Unified Border Element
NEW QUESTION # 74
Refer to the exhibit. A standard local route group is configured for long-distance calls. Calls from building A succeed, but calls from building B fail. On the system. Each building has is own device pool. The DNA tool is used to test the configuration. How is this issue resolved?
- A. Modify the route pattern to add a prefix of 91
- B. Add a sip trunk inside route group Standard Local Route Group.
- C. Add a local route group on the device pool configuration.
- D. Change the partition of the route pattern
Answer: C
NEW QUESTION # 75
Which description of RTP timestamps or sequence numbers is true?
- A. Sequence numbers increase by four for each RTP packet transmitted.
- B. The timestamp is used to place the incoming audio and video packets in the correct timing order (playout
- C. Timestamps increase by the time "carrying" by a packet.
- D. The sequence number is used to detect losses.
Answer: B
Explanation:
delay compensation).
NEW QUESTION # 76
Which description of RTP timestamps or sequence numbers is true?
- A. Sequence numbers increase by four for each RTP packet transmitted.
- B. Timestamps increase by the time "carrying" by a packet.
- C. The sequence number is used to detect losses.
- D. The timestamp is used to place the incoming audio and video packets in the correct timing order (playout delay compensation).
Answer: D
NEW QUESTION # 77
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Cisco 300-815 exam is intended for IT professionals who have experience in implementing and managing Cisco Unified Communications solutions. 300-815 exam covers a range of topics, including call routing, call queuing, call control, dial plans, and mobility services. 300-815 exam format includes multiple-choice questions, simulations, and performance-based questions. Candidates must complete the exam within 120 minutes and score at least 825 out of 1000 to pass.
300-815 exam questions for practice in 2023 Updated 120 Questions: https://www.dumpexams.com/300-815-real-answers.html
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