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Cisco 300-815 exam is a challenging exam that covers a wide range of topics related to advanced call control and mobility services. Some of the topics covered in 300-815 exam include Cisco Unified Communications Manager (CUCM) call routing and signaling, Cisco Unified Border Element (CUBE), Quality of Service (QoS), Session Initiation Protocol (SIP) and H.323 protocols, and Cisco Unified Communications Mobile and Remote Access (MRA) solutions.
Cisco 300-815 exam consists of multiple-choice questions, drag-and-drop questions, and simlet questions. 300-815 exam duration is 90 minutes, and the passing score is 825 out of 1000. 300-815 exam fee is $300, and candidates can register for the exam through the Pearson VUE website.
NEW QUESTION # 71
An engineer is configuring a call park feature in Cisco Unified Communications Manager Express. Which command does the engineer use to ensure that the call is reverted to the user after 60 seconds?
- A. R2(config-ephone-dn)#park-slot timeout 30 limit 2 recall alternate 3002
- B. R2(config-ephone-dn)#park reservation-group 60
- C. R2(config-ephone-dn)#park reservation-group 1
- D. R2(config-ephone-dn)#park-slot timeout 60 limit 2 recall alternate 3002
Answer: D
NEW QUESTION # 72
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. On the SIP Profile, the configuration parameter SIP Rel1XX Options must be set to Send PRACK for all 1xx Messages.
- B. Accept Audio Codec Preferences in Received Offer must be set to On.
- C. Early Offer for G Clear Calls must be enabled.
- D. Allow Passthrough of Configured Line Device Caller Information must be enabled.
Answer: A
NEW QUESTION # 73
Which two configuration parameters are prerequisites to set Native Call Queuing on Cisco Unified Communications Manager? (Choose two.)
- A. Cisco RIS data collector service must be running on the same server as the Cisco CallManager service.
- B. The phone button template must have the Queue Status Softkey configured.
- C. Cisco IP Voice Media Streaming Service must be activated on at least one node in the cluster.
- D. The maximum number of callers allowed in queue must be 10.
- E. A unicast music on hold audio source must be configured.
Answer: A,C
NEW QUESTION # 74
A user reports when they press the services key they do not receive a user ID and password prompt to assign the phone extension. Which action resolves the issue?
- A. Subscribe the phone to the Cisco Extension Mobility service.
- B. Create the end user and associate it to the device profile.
- C. Create the default device profiles for all phone models that are used.
- D. Assign the extension as a mobile extension.
Answer: A
NEW QUESTION # 75
Signal number reach call phone that not answered are leaving voicemails on the cell phone rather the corporate mailbox. Which two options will resolve this issue? (Choose two.)
- A. Decrease the T301 timer Decrease the Answer Too Late timer
- B. Check the Enable Unified Mobility features checkbox
- C. Check the Enable Extend and Connect checkbox
- D. Decrease the T302 timer
Answer: A,B
NEW QUESTION # 76
Refer to the exhibit.
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_B sends a SIP-REFER message to the Cisco Unified CM with Phone_C information in the Refer-To section.
- B. 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.
- 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. Phone_A sends a SIP-REFER message to the Cisco Unified Communications Manager with Phone_C information in the Refer-To section.
- E. 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.
Answer: D,E
NEW QUESTION # 77
Which two types of authentication are supported for the configuration of Intercluster Lookup Service? (Choose two.)
- A. passwords
- B. username and secret key
- C. FQDN of the servers defined in DNS
- D. TokenID
- E. TLS certificates
Answer: A,E
NEW QUESTION # 78
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. SDI trace
- D. SDL trace
Answer: B
Explanation:
Section: Call Control and Dial Planning
NEW QUESTION # 79
Refer to the exhibit.
The Cisco Unified Border Element receives an INVITE matching inbound dial peer 5002. The outbound dial peer supports only iLBC. and a Local Transcoding Interface is allocated. Based on the configuration and SDP from the INVITE message, which codec is chosen by Cisco Unified Border Element for the inbound call leg?
- A. G.711 U-law
- B. G.711 A-law
- C. G.729br8
- D. G.729r8
Answer: D
NEW QUESTION # 80
Which two types of distribution algorithm are within a line group? (Choose two.)
- A. top down
- B. bottom up
- C. random
- D. highest preference
- E. circular
Answer: A,E
Explanation:
Section: Call Control and Dial Planning
Explanation/Reference: https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/9_0_1/ccmcfg/ CUCM_BK_CDF59AFB_00_admin-guide-90/CUCM_BK_CDF59AFB_00_admin-guide_chapter_0100011.html
NEW QUESTION # 81
When a third-party SIP Phone System is dialed inbound across a Cisco Unified Border Element, DTMF is failing. The third-party vendor accepts only out-of-band DTMF. Which configuration should be added to the outgoing dial peer to resolve this issue?
- A. dtmf-relay rtp-nte
- B. dtmf-relay sip-kpml
- C. dtmf-relay cisco-rtp
- D. dtmf-relay h245-signal
Answer: B
NEW QUESTION # 82
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 E
- B. Option D
- C. Option B
- D. Option A
- E. Option C
Answer: B,E
NEW QUESTION # 83
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
Explanation:
Section: Call Control and Dial Planning
NEW QUESTION # 84
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. The rtpmap:0 value for the negotiated codec is marking DTMF as inactive.
- 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 # 85
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. external phone number mask
- B. called party transform mask
- C. +E.164 alternate number mask
- D. enterrise alternate number mask
- E. AAR destination mask
Answer: A,E
NEW QUESTION # 86
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 System Parameters in Cisco Unified Communications Manager and change the range of media ports to 20000-22000.
- B. Go to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 16384-32767
- C. Go to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 20000-22000.
- D. Ask the firewall administrator to change the ports to TCP.
- E. Ask the firewall administrator to change the range of UDP ports to 16384-32767.
Answer: B,E
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 # 87
An administrator discovers that employees are making unauthorized long-distance and international calls from logged-off Extension Mobility phones when the authorized users are away from their desks Which two configurations should the administrator configure in the Cisco UCM to avoid this issue? (Choose two.)
- A. Remove the long-distance & international pattern's partitions from the calling search space of the physical phone.
- B. Add the long-distance & international pattern's partitions to the calling search space of the physical phone's directory number.
- C. Add the long-distance & international pattern's partitions to the calling search space of the physical phone.
- D. Remove the long-distance & international pattern's partitions from the calling search space of the device profile.
- E. Add the long-distance & international pattern's partitions to the calling search space of the device profile
Answer: A,E
NEW QUESTION # 88
Which configuration element of a hunt group allows for changing Calling Party Transformations settings?
- A. hunt pilot
- B. hunt list
- C. line group
- D. route group
Answer: A
Explanation:
Section: Cisco Unified CM Call Control Features
Explanation/Reference:
Reference: https://community.cisco.com/t5/ip-telephony-and-phones/call-alerting-on-hunt-group-as-shared-line/ td-p/2658015
NEW QUESTION # 89
Refer to the exhibit.
ILS has been configured between two hubs using this configuration. The hubs appear to register successfully, but ILS is not functioning as expected. Which configuration step is missing?
- A. Use TLS Certificates must be selected.
- B. Trust certificates for ILS have not been installed on the clusters
- C. A password has never been set for ILS.
- D. The Cluster IDs have not been set to unique values
Answer: D
NEW QUESTION # 90
CollabCorp is a global company with two clusters, emea.collab corp and apac.collab.corp. URI dialing is implemented and working in each cluster. The company configured routing between clusters to make inter-cluster calls via URI. but this is not working as expected. Which two configuration elements should be checked to resolve this issue? (Choose two.)
- A. SIP trunk
- B. directory URI partition
- C. calling search space and partition
- D. SIP route pattern
- E. intercluster trunk
Answer: A,D
NEW QUESTION # 91
A new deployment is using MVA for a specific user on the sales team, but the user is having issues when dialing DTMF. Which DTMF method must be configured in resolve the issue?
- A. channel
- B. in-band
- C. gateway
- D. out-of-band
Answer: D
NEW QUESTION # 92
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