[Sep 14, 2022] Get New 300-815 Practice Test Questions Answers [Q18-Q43]

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[Sep 14, 2022] Get New 300-815 Practice Test Questions Answers

300-815 Dumps and Exam Test Engine

NEW QUESTION 18
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. Create the default device profiles for all phone models that are used.
  • B. Assign the extension as a mobile extension.
  • C. Subscribe the phone to the Cisco Extension Mobility service.
  • D. Create the end user and associate it to the device profile.

Answer: C

 

NEW QUESTION 19
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. Activate the Cisco CallManager service.
  • B. Activate the Cisco Dialed Number Analyzer service.
  • C. Restart the subscriber
  • D. Activate the Cisco Dialed Number Analyzer Server service.
  • E. Activate the Cisco Extended Functions service.

Answer: B,D

 

NEW QUESTION 20
An engineer must implement call restriction to toll-free numbers using a class of restriction in a branch Cisco UCME. In which two places is the corlist incoming or cor Incoming command configured? (Choose two.)

  • A. "dial-peer cor custom" configuration mode
  • B. "voice register pool" configuration mode
  • C. "voice register global" configuration mode
  • D. "ephone-dn' configuration mode
  • E. "telephony-service" configuration mode

Answer: B,D

 

NEW QUESTION 21
A support engineer is troubleshooting a voice network. When conducting a search for call setup details related to calling search space issues, which trace files should be investigated?

  • A. Cisco IP Manager Assistant
  • B. CTI Manager traces
  • C. CallManager traces
  • D. Call logs

Answer: C

 

NEW QUESTION 22
A user in location X dials an extension at location
Y. The call travels through a QoS-enabled WAN network, but the user experiences choppy or clipped audio. What is the cause of this issue?

  • A. ptime mismatch
  • B. missing Call Admission Control
  • C. phone class of service issue
  • D. codec mismatch

Answer: D

Explanation:
Section: Cisco Unified Border Element

 

NEW QUESTION 23
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.323 messages
  • B. debug H.246 asn 1
  • C. debug H.225 asn1
  • D. debug H.225 media
  • E. debug H.323 asn 1

Answer: C

 

NEW QUESTION 24
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. Set up translation-profile on the incoming dial peer to match incoming traffic.
  • B. Create specific matching for "222333444" on the incoming dial peer.
  • C. Fix the voice translation-rule to match specifically number "222333444" and change it to "444333222".
  • D. Change the destination-pattern on the outgoing dial peer to match "444333222".

Answer: A

 

NEW QUESTION 25
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 > Device Pool
  • C. System > Location Info
  • 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 26
How does an engineer globalize routing for ingress calls coming from the PSTN to internal DNs?

  • A. At Cisco Unified Communications Manager, put the calling number in E.164 format and the called number in E.164 format.
  • B. At Cisco Unified CM, put the calling number in E.164 format and the called number in PSTN 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: B

 

NEW QUESTION 27
When configuring hunt groups, where do you add the individual directory numbers that will be part of the group?

  • A. line group
  • B. hunt pilot
  • C. route group
  • D. hunt list

Answer: A

Explanation:
Section: Cisco Unified CM Call Control Features
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_010101.html

 

NEW QUESTION 28
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. Change Session Refresh Method on the SIP profile to INVITE.
  • C. Enable SIP ReMXX Options on the SIP profile.
  • D. Increase Retry INVITE to 20 seconds on the SIP profile.

Answer: B

 

NEW QUESTION 29
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. parallel
  • B. peer
  • C. call-blast
  • D. sequential

Answer: B

 

NEW QUESTION 30
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. Imported Dial Plan Replication Database Object Lower Limit
  • C. Global Data Service Parameter Limit
  • D. ILS Active Learned Object Upper Limit

Answer: A

Explanation:
Section: Cisco Unified CM Call Control Features
Explanation/Reference: https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/12_5_1SU1/systemConfig/ cucm_b_system-configuration-guide-1251su1/cucm_b_system-configuration-guide-
1251su1_restructured_chapter_0100011.html#CUCM_TK_I7C708C2_00

 

NEW QUESTION 31
Refer to the exhibit.

Outbound calls to the service provider cause intermittent errors due to a codec mismatch. The internal network sends early offer SDP that contains only G.711 A-law. The service provider reports that some destinations support only G.711 A-law while others support only iLBC. The service provider also allows only 20 active calls at a time Which configuration allows successful media negotiation for all calls using outbound dial peers 5002 and 5003?

  • A. Option C
  • B. Option B
  • C. Option A
  • D. Option D

Answer: D

 

NEW QUESTION 32
Drag and drop the commands from the bottom to the blanks in the code to implement a translation rule to allow only 11 digits to be received over a SIP trunk to a SIP provider. The Cisco UCM is currently sending calls to the Cisco Unified Border Element in E.
164 format. Not all options are used.

Answer:

Explanation:

 

NEW QUESTION 33
A company has users that are logged in to hunt groups. However, there is a requirement for hunt group configurations to provide an option to turn on audible ringtones when calls to a line group arrive at a phone that is logged out and on a break. This ringtone alerts a logged-out user that there is an incoming call to a hunt list to which the line is a member, but the call does not ring at the phone of that line group member because of the logged-out status. Which action meets this requirement?

  • A. Set the service parameter Party Entrance Tone to True."
  • B. Configure the service parameter hunt group logoff notification and specify the name of the ringtone file.
  • C. Configure the HLog softkey on the phone so that while a user is logged off, it plays an audible tone when a call is missed.
  • D. Set the service parameter Enterprise Feature Access number for hunt group logout and set up an access number

Answer: B

 

NEW QUESTION 34
Refer to the exhibit.

Within the North American Numbering Plan, gateways located in Ottawa, Canada and marked as "YOW" are assigned to the Calling Party Transformation CSS NANP_CgPTP, which contains partition NANP_calling_xforms. What is the calling-party number and the numbering type if the calling user +1613-555-1234 dials the number?

  • A. calling number 011-1-613-555-1234 and numbering type "subscriber"
  • B. calling number 613-555-1234 and numbering type "national"
  • C. calling number 011613-555-1234 and numbering type "international"
  • D. calling number 613-555-1234 and numbering type "subscriber"

Answer: B

 

NEW QUESTION 35
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.323 messages
  • B. debug H.246 asn 1
  • C. debug H.224 asn1
  • D. debug H.225 media
  • E. debug H.323 asn 1

Answer: C

 

NEW QUESTION 36
Refer to the exhibit.

A Cisco Unified Border Element continues to send 180/183 with the required: 100rel header to Cisco UCM. and the call eventually disconnects How is the issue resolved?

  • A. Enable 'SIP ReI1XX Options* and -Early Offer Support" on the SIP Profile Configuration Page in Cisco UCM.
  • B. Disable "Send send-receive SDP in mid-call INVITE* on the SIP Profile Configuration Page in Cisco UCM.
  • C. Enable *Early Offer support for voice and video calls" on the SIP Profile Configuration Page in Cisco UCM.
  • D. Disable "SIP Rel1XX Options* and 'Early Offer Support* on the SIP Profile Configuration Page m Cisco UCM.

Answer: C

 

NEW QUESTION 37
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 Ack
  • B. H.245 Terminal Capability Set
  • C. H.245 Open Logical Channel
  • D. H.225 Connect

Answer: C

 

NEW QUESTION 38
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. Ask the firewall administrator to change the ports to TCP.
  • B. Go to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 20000-22000.
  • C. Ask the firewall administrator to change the range of UDP ports to 16384-32767.
  • D. Go to System Parameters in Cisco Unified Communications Manager and change the range of media ports to 20000-22000.
  • E. Go to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 16384-32767

Answer: C,E

 

NEW QUESTION 39
Which configuration must an administrator perform to display Translation Pattern operations in Cisco Unified Communications Manager SDL traces?

  • A. Set up the Digit Analysis Complexity in Service Parameters for Cisco Unified CM to TranslationAndAlternatePatternAnalysis.
  • B. By default, the Translation Patterns operations are printed in SDL traces, so no additional configuration is necessary.
  • C. Check the Translation Patterns Analysis check box in Micro Traces on the Cisco Unified CM Serviceability page.
  • D. Enable the Detailed Call Analysis option under Enterprise Parameters for Unified CM.

Answer: D

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 40
What is the relationship between partition, time schedule, and time period in Time-of-Day routing in Cisco Unified Communications Manager?

  • A. A partition can have multiple time schedules assigned. A time schedule contains only one time period.
  • B. A partition can have multiple time schedules assigned. A time schedule contains one or more time periods.
  • C. A partition can have one time schedule assigned. A time schedule contains only one time period.
  • D. A partition can have one time schedule assigned. A time schedule contains one or more time periods.

Answer: B

 

NEW QUESTION 41
An engineer is troubleshooting local ringback on a Cisco SIP gateway The gateway is not ignoring the SIP 180 response with SDP from the service provider, and the far end device is sending the 180 with SDP to play ringback from the IP address specified m the SDP Which configuration change must be made on the gateway to resolve the issue?

  • A. Router(conf-voi-serv)# dlisable-early-media 180
  • B. Router(conftg-sip-ua)# disable-early-media 180
  • C. Router(config-sip-ua)# no disable-early-media 180
  • D. Router(con(-voi-serv)# no disable-early-media 180

Answer: B

 

NEW QUESTION 42
Which IOS command creates a SIP-enabled dial peer?

  • A. dial-peer voice 20 pots
  • B. voice dial-peer 20 sip
  • C. dial peer voice 20 sip
  • D. dial-peer voice 20 voip

Answer: D

 

NEW QUESTION 43
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Cisco 300-815 Exam Certification Details:

Passing ScoreVariable (750-850 / 1000 Approx.)
Sample QuestionsCisco 300-815 Sample Questions
Exam NameImplementing Cisco Advanced Call Control and Mobility Services
Exam Price$300 USD

 

2022 New Exam-Killer 300-815 PDF Recently Updated Questions: https://www.exam-killer.com/300-815-valid-questions.html

Cisco 300-815 DUMPS WITH REAL EXAM QUESTIONS: https://drive.google.com/open?id=1BpG52O_8p6QkvabS7IrTOR2qiPCexDA0