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Free Cisco 300-815 Exam 2023 Practice Materials Collection
Cisco 300-815 certification exam is designed to test the knowledge and skills of IT professionals in implementing advanced call control and mobility services on Cisco platforms. It is one of the most sought-after certifications in the field of networking and communication, as it validates the ability of candidates to manage and troubleshoot complex communication networks.
What Topics Does 300-815 Assess?
Earning a passing score in such a Cisco exam will require the candidate to have a thorough understanding of all the test parts that are organized in the following exam domain:
- Mobility
The final domain deals with Unified Communications Manager Mobility by Cisco and makes up 10% of the exam content. Therefore, the entrant should know how to configure unified mobility, extension mobility, and device mobility. Moreover, it is also essential for the candidate to be familiar with successfully troubleshooting the aforementioned mobility technologies.
- The element for Cisco Unified Border
The third section of the exam includes 15% of all questions regarding the Cisco Unified Border Element technology. Thus, the entrant should be capable of configuring various Cisco Unified Border Element dial plan parts such as DTMF, voice translations rules & profiles, the list of codec preferences, dial peers, headers, and the manipulation of SDP with diverse SIP profiles. This domain also deals with troubleshooting all the aforementioned Cisco Unified Border Element dial plan components.
- Dial planning & call control
The fourth portion of this exam covers 25% of the content and consists of questions regarding the Cisco Unified Communications Manager. The candidate should be capable of configuring various globalized call routing components of the Cisco Unified Communications Manager. These components include translation patterns, route patterns, transformation patterns, TEHO, SIP trunking, and standard local routing group. Moreover, the candidates must be able to troubleshoot all these globalized elements for call routing.
NEW QUESTION # 53
When you troubleshoot H.323 call setup, which message informs you that the called party is being notified about the call?
- A. PROCEEDING
- B. RINGING
- C. ALERTING
- D. CONNECT
Answer: D
NEW QUESTION # 54
What is first preference condition matched in a SIP-enabled incoming dial peer?
- A. target carrier-id
- B. incoming called-number
- C. incoming uri
- D. answer-address
Answer: C
NEW QUESTION # 55
Refer to the exhibit.
A call mode through the Cisco Unified Border Element to pilot 2000 is foiling. What is causing the call to foil?
- A. VAD was not disabled on the outgoing dial poor.
- B. No codecs are configured on the dial peers
- C. The destination pattern is incorrect for the dialed number.
- D. The Cisco Unified Border Element is not receiving a response to its OPTION keepahves.
Answer: C
NEW QUESTION # 56
Due to a shortage of physical interfaces on a device the administrator requires that a loopback for RTP is used.
Which command is required when using a loopback interface for RTP?
- A. voice-class sip early-offer forced.
- B. voice-class sip bind media source-interface Loopback0
- C. voice-class sip resources priority mode passthrough
- D. voice-class sip bind control source-interface Loopback0
Answer: B
NEW QUESTION # 57 
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 acknowledgement of provisional responses?
- A. Allow Passthrough of Configured Line Device Caller Information must be enabled.
- B. On the SIP Profile, the configuration parameter SIP Rel1XX Options must be set to Send PRACK for all 1xx Messages.
- C. Early Offer for G Clear Calls must be enabled.
- D. Accept Audio Codec Preferences in Received Offer must be set to On.
Answer: B
Explanation:
Section: Signaling and Media Protocols
Explanation
NEW QUESTION # 58
Which description of RTP timestamps or sequence numbers is true?
- A. Sequence numbers increase by four for each RTP packet transmitted.
- B. The sequence number is used to detect losses.
- C. The timestamp is used to place the incoming audio and video packets in the correct timing order (playout
- D. Timestamps increase by the time "carrying" by a packet.
Answer: C
Explanation:
delay compensation).
NEW QUESTION # 59
Refer to the exhibit.
Users report that when they dial the emergency number 9911 from any internal phone, it takes a long time to connect with the emergency operator. Which action resolves this issue?
- A. Adjust the service parameter T204 timer to the desired value.
- B. Adjust the service parameter T302 timet to the desired value.
- C. Point the emergency pattern directly to the PSTN gateway.
- D. Check the Urgent Priority check box under 9.911 pattern.
Answer: D
NEW QUESTION # 60
An IP Telephony administrator is deploying IP phones The administrator has an existing Cisco UCME router with several SCCP & SIP phones registered. The administrator receives a request for a new SIP phone with MAC address 1111 2222.3333 and directory number 2050 to be added in the Cisco UCME. Which two configurations should be added in CME to support this request? (Choose two )
- A. Option B
- B. Option E
- C. Option D
- D. Option A
- E. Option C
Answer: C,E
NEW QUESTION # 61
When locations-based Call Admission Control denies the call, which two masks can AAR apply when routing the call through the PSTN? (Choose two.)
- A. +E.164 alternate number mask
- B. external phone number mask
- C. enterrise alternate number mask
- D. called party transform mask
- E. AAR destination mask
Answer: B,E
NEW QUESTION # 62
Which two configuration parameters are prerequisites to set Native Call Queuing on Cisco Unified Communications Manager? (Choose two.)
- A. Cisco IP Voice Media Streaming Service must be activated on at least one node in the cluster.
- B. The maximum number of callers allowed in queue must be 10.
- C. The phone button template must have the Queue Status Softkey configured.
- D. Cisco RIS data collector service must be running on the same server as the Cisco CallManager service.
- E. A unicast music on hold audio source must be configured.
Answer: A,D
Explanation:
Reference:
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/12_0_1/systemConfig/ cucm_b_system-configuration-guide-1201/cucm_b_system-configuration-guide- 1201_chapter_01001101.html#CUCM_RF_C960BC9A_00
NEW QUESTION # 63
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. Global Data Service Parameter Limit
- B. Imported Dial Plan Replication Database Object Lower Limit
- C. ILS Active Learned Object Upper Limit
- D. ILS Max Number of Learned Objects in Database
Answer: D
NEW QUESTION # 64 
Refer to the exhibit. Users report that when they dial the emergency number 9911 from any internal phone, it takes a long time to connect with the emergency operator. Which action resolves this issue?
- A. Adjust the service parameter T204 timer to the desired value.
- B. Adjust the service parameter T302 timet to the desired value.
- C. Point the emergency pattern directly to the PSTN gateway.
- D. Check the Urgent Priority check box under 9.911 pattern.
Answer: D
Explanation:
Section: Call Control and Dial Planning
NEW QUESTION # 65
After configuring a Cisco CallManager Express with Cisco Unity Express, inbound calls from the PSTN SIP trunk receive a ring tone for 20 seconds and then a busy signal instead of voicemail. Which configuration fixes this problem?
- A. Router(config)# voice service voip
Router(conf-voi-serv)#allow-connections voice-mail mod - B. Router(config)#dial-peer voice 2 voip
Router(config-dial-peer)#no vad - C. Router(config)# voice service voip
Router(conf-voi-serv)#allow-connections h323 to h323 - D. Router(config)# voice service voip
Router(conf-voi-serv)#no supplementary-service sip moved-temporarily
Answer: C
Explanation:
Section: Call Control and Dial Planning
Explanation/Reference: https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cusrst/admin/sccp_sip_srst/configuration/ guide/SCCP_and_SIP_SRST_Admin_Guide/srst_call_handling.html
NEW QUESTION # 66
A network engineer designs a new dial plan and wants to block a certain range of numbers (8135100 through
8135105). What is the most specific route pattern that can be configured to block only the numbers in this range?
- A. 813510[12345]
- B. 813510[^0-5]
- C. 81XXXXX
- D. 813510[012345]
Answer: D
Explanation:
Section: Call Control and Dial Planning
NEW QUESTION # 67
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. Call Routing > Route/Hunt > Local Route Group Names
- C. System > Location Info
- D. System > Device Pool
Answer: B
Explanation:
Section: Call Control and Dial Planning
Explanation/Reference: https://www.uccollabing.com/configuring-standard-local-route-group-cucm/
NEW QUESTION # 68
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 acknowledgement of provisional responses?
- A. Allow Passthrough of Configured Line Device Caller Information must be enabled.
- B. On the SIP Profile, the configuration parameter SIP Rel1XX Options must be set to Send PRACK for all 1xx Messages.
- C. Early Offer for G Clear Calls must be enabled.
- D. Accept Audio Codec Preferences in Received Offer must be set to On.
Answer: B
NEW QUESTION # 69
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. Configure AAR to reroute calls that are denied by Call Admission Control through the PSTN.
- 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. Reroute all calls through the PSTN and avoid using WAN.
Answer: A
NEW QUESTION # 70
Refer to the exhibit.
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. Which action must the administrator take to fix the issue?
- A. Create specific matching for "222333444" on the incoming dial peer.
- B. Fix the voice translation-rule to match specifically number "222333444" and change it to "444333222".
- C. Change the destination-pattern on the outgoing dial peer to match "444333222".
- D. Set up translation-profile on the incoming dial peer to match incoming traffic.
Answer: D
NEW QUESTION # 71
Refer to the exhibit.
For long-distance calls, users must prefix their dialed number with "91." The translation pattern was created to strip the 91 as the PSTN expects a 10- digit number. The PSTN also requires the calling number to be set to 9195551234. However, the service provider has said calls with a different calling number are being received. How is this issue resolved?
- A. Enable Force Authorization Code on the route pattern.
- B. Enable Use Calling Party's External Phone Number Mask on the translation pattern.
- C. Disable Use Calling Party's External Phone Number Mask on the route pattern.
- D. Change the partition of the translation pattern from none to pstn_pt.
Answer: C
NEW QUESTION # 72
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. There is SIP Delayed Offer. DTMF is supported only in Early Offer.
- B. No DTMF is negotiated.
- C. The negotiated RTP port is outside of the range described by RFC, so inband DTMFs do not work.
- D. The rtpmap:0 value for the negotiated codec is marking DTMF as inactive.
Answer: B
NEW QUESTION # 73
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. Real-Time Monitoring Tool
- B. Dialed Number Analyzer
- C. SDL trace
- D. SDI trace
Answer: B
Explanation:
Section: Call Control and Dial Planning
NEW QUESTION # 74
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