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300-815 exam torrent Cisco study guide
Cisco 300-815 exam is a certification test that evaluates the knowledge and skills of IT professionals in implementing advanced call control and mobility services using Cisco technologies. 300-815 exam is designed for network administrators, engineers, and consultants who are responsible for designing, deploying, and managing enterprise-level collaboration solutions using Cisco Unified Communications Manager (CUCM) and its associated applications.
Cisco 300-815 exam is designed for IT professionals who want to prove their advanced knowledge and skills in implementing Cisco's call control and mobility services. Implementing Cisco Advanced Call Control and Mobility Services certification exam focuses on the implementation and configuration of various Cisco technologies such as Cisco Unified Communications Manager (CUCM), Cisco Unified IM and Presence, Cisco Unity Connection, and Cisco Expressway. Passing 300-815 exam will validate the candidate's ability to implement complex collaboration solutions that help businesses improve their productivity and communication.
NEW QUESTION # 22
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 /^\([2-9]..[2-9]......$\)/ /\1/
- C. rule 1 /^\+1\([2-9]..[2-9]......$\)/ /\0/
- D. rule 1/^\+1\([2-9]..[2-9]......$\)/ /\1/
Answer: D
NEW QUESTION # 23
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. Set up the Digit Analysis Complexity in Service Parameters for Cisco Unified CM to TranslationAndAlternatePatternAnalysis.
- C. By default, the Translation Patterns operations are printed in SDL traces, so no additional configuration is necessary.
- D. Enable the Detailed Call Analysis option under Enterprise Parameters for Unified CM.
Answer: B
Explanation:
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/hcs/12_5/HCS_Solution/Troubleshooting/ chcs_b_troubleshooting-guide/chcs_m_troubleshooting-voice-application- components.html?dtid=osscdc000283
NEW QUESTION # 24
Refer to the exhibit.
Refer to the exhibit. A company deploys an additional instance of Cisco UCME named CUCME 2 in a branch office. Phones in the main office are register of Unified CME named CUCME 1. Phones in the branch office are registered with their extension to CUCME 2. The engineer needs to configure CUCME the branch office using an 8-kbps codec. Which configuration should be applied to CUCME 1?
- A.
- B.
- C.
- D.
Answer: A
NEW QUESTION # 25
Which call pickup feature allows users to pick up incoming calls in a group that is associated with their own group?
- A. Group Call Pickup
- B. Other Group Pickup
- C. BLF Call Pickup
- D. Directed Call Pickup
Answer: B
NEW QUESTION # 26
An organization has decided to implement hunt groups to help with the distribution of calls between different members. The administrator must configure hunt groups on Cisco UCM. In the configuration, at which step does an administrator have the option to configure a distribution algorithm (top down, circular, longest idle time, broadcast)?
- A. route list
- B. line group
- C. hunt list
- D. hunt group
Answer: C
NEW QUESTION # 27
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 # 28
The same digits sent in OOB, as well as in-band on the Cisco Unified Border Element, are being interpreted as duplicate digits by the receiving end. Which configuration must an engineer implement to resolve this issue?
- A. DTMF Relay SIP-Notify
- B. DTMF Relay Digit-Drop
- C. DTMF Relay SIP-KPML
- D. DTMF Relay RTP-nte
Answer: B
NEW QUESTION # 29
Refer to the exhibit.
Refer to the exhibit An engineer deploys a Cisco Unified Border Element for SIP trunk interconnection to an ITSP The engineer performs test calls but there is no audio when the called party picks up the phone. What must the engineer do to resolve the issue?
- A. Enable Check media source packets.
- B. Disable SIP early-media for 180 responses with SDP
- C. Disable SIP User Agent bind status (media).
- D. Decrease the SIP max-forwards value to 16.
Answer: D
NEW QUESTION # 30
Refer to the exhibit.
A collaboration engineer is troubleshooting an issue where external callers cannot leave voicemail messages. Also, internal users report hearing the reorder tone (fast busy) when they attempt to retrieve voicemail messages from their Cisco IP phones. Which action resolves the issue?
- A. Verify that the correct port numbers are used for the SIP trunk.
- B. Ensure that the SIP Trunk Security Profile is configured to use UDP for transport.
- C. Start the Cisco Call Manager service at the destination.
- D. Ensure that Cisco UCM can resolve the destination address via DNS.
Answer: D
NEW QUESTION # 31
Which IOS command creates a SIP-enabled dial peer?
- A. dial peer voice 20 sip
- B. dial-peer voice 20 voip
- C. voice dial-peer 20 sip
- D. dial-peer voice 20 pots
Answer: B
Explanation:
https://www.ciscopress.com/articles/article.asp?p=664148&seqNum=6
NEW QUESTION # 32
An administrator is configuring Cisco UCM and the system to send *.webex.com traffic to a Cisco UCM Session Management Edition cluster. The administrator wants to limit which endpoints can reach *.webex.com. Which two items must the administrator configure for the SIP route pattern?
(Choose two.)
- A. connected party transformation
- B. calling party transformation
- C. called party URI transformation
- D. destination SIP trunk of the SIP route pattern
- E. partition of the SIP route pattern
Answer: D,E
Explanation:
https://www.cisco.com/c/en/us/td/docs/telepresence/tcs/7_1/admin/administration/tcs_7_0/cucm.h tml
NEW QUESTION # 33
Refer to the exhibit.
An administrator received reports that users do not hear a ringback when they dial specific external numbers. For example, the Cisco Unified Border Element receives a delayed offer invite from Cisco UCM and responds with the 183 session progress in the exhibit. Assume that the PSTN is responsible for generating ringback in these problematic scenarios. Which action resolves the issue?
- A. On Cisco Unified Border Element: voice service voip sip
Re1xx require 100rel - B. On Cisco Unified Border Element: voice service voip sip Re1xx disable 100rel
- C. Enable early media on the appropriate SIP trunk on Cisco UCM.
- D. Edit the configuration for SIP ReMXX Options in the appropriate SIP profile on Cisco UCM to Send PRACK if 1xx Contains SDP.
Answer: D
NEW QUESTION # 34
Refer to the exhibit. All calls from site A to site B are failing, and the issue has been identified as a media negotiation problem. Which configuration change resolves this issue?
- A. Increase the bandwidth allowance between the RTP_Reg and SJ_Reg regions to 64 kbps.
- B. Enable Early Offer on the SIP trunk.
- C. Disable G.722 on all devices at both sites.
- D. Create a new audio codec preference list with G.711 U-law 64k as the highest priority and apply it to RTP_Reg and SJ_Reg.
Answer: D
NEW QUESTION # 35
A customer reports audio quality issues between video endpoints in the HQ location in California and one of the branches in Texas. Which two actions must RTCP take to troubleshoot this issue?
(Choose two.)
- A. Compress the headers of RTP traffic to lower the bandwidth consumption over the WAN, which allows more calls with less bandwidth consumed.
- B. Allow for VAD to be used for calls using the G7.29 codec, which reduces the usage of the WAN bandwidth and saves around 30% of bandwidth per call
- C. Configure the rtcp keepalive command in Cisco Unified Border Element to generate reports, which can be reviewed using the debug voip rtcp packet command.
- D. Gather statistics on a media connection and information such as packets sent, lost packets, jitter, feedback, and round-trip delay. This information can help isolate the type of audio quality issues and the direction of the affected traffic.
- E. Encrypt the media to stop rogue devices from replying and putting that traffic back on the WAN, which avoids any extra bandwidth and ensures the quality of the calls.
Answer: C,D
NEW QUESTION # 36
A user requests a feature to send an active call to the mobile phone number on the physical phone. As an administrator, what should be configured in the Cisco UCM to accomplish this?
- A. A Remote Destination Profile having the same extension as the physical phone's Directory Number, add the mobile phone number to the RDP as a Remote Destination. Add the softkey
"Join" to the physical phone's softkey template. - B. A Remote Destination Profile having the same extension as the Mobile phone's number adds the physical phone's Directory Number to the RDP as a Remote Destination. Add the softkey
"Join" to the physical phone's softkey template. - C. A Remote Destination Profile having the same extension as the physical phone's Directory Number, add the mobile phone number to the RDP as a Remote Destination. Add the softkey
"Mobility" to the physical phone's softkey template. - D. A Remote Destination Profile having the same extension as the Mobile phone's number adds the physical phone's Directory Number to the RDP as a Remote Destination. Add the softkey
"Mobility" to the physical phone's softkey template.
Answer: C
NEW QUESTION # 37
An engineer is configuring a Cisco Collaboration system for SIP endpoints and must enable Survivable Remote Site Telephony for these endpoints. Which code completes this configuration on the SRST gateway?
- A.
- B.
- C.
- D.
Answer: D
NEW QUESTION # 38
What is a function of the metadata carried in SIP sessions between the recording client and the recording server?
- A. It forks RTP media to the recorder.
- B. It provides advanced capabilities, such as speech analytics.
- C. It identifies the participant change due to transfers during the call.
- D. It sets up a new SIP session
Answer: C
NEW QUESTION # 39
Calls to a particular extension are not routing to voicemail. The user reaches the voicemail system from the handset by pressing the Messages button.
Which configuration parameter causes this problem?
- A. The voicemail pilot number is missing from the telephony service configuration on Cisco UCME
- B. The voicemail pilot number is missing from the call handling on Cisco Unity Express
- C. The voicemail pilot number for call forwarding is missing from the ephone
- D. The voicemail pilot number for call forwarding is missing from the ephone-dn
Answer: D
NEW QUESTION # 40
When you troubleshoot H.323 call setup, which message informs you that the called party is being notified about the call?
- A. CONNECT
- B. ALERTING
- C. PROCEEDING
- D. RINGING
Answer: B
NEW QUESTION # 41
Refer to the exhibit. How many maximum hops can an ILS updarte traverse?
- A. 0
- B. 1
- C. 2
- D. 3
Answer: A
NEW QUESTION # 42
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 negotiated RTP port is outside of the range described by RFC, so inband DTMFs do not work.
- C. The rtpmap:0 value for the negotiated codec is marking DTMF as inactive.
- D. No DTMF is negotiated.
Answer: D
NEW QUESTION # 43
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