Cisco 300-815 Online Practice Questions and Exam Preparation

Cisco 300-815 Online Questions & Answers

• Question 111:

  Refer to the exhibit.

  

  This message is sent to the device being placed on hold for the Music On Hold audio setup. The held party reports receiving dead air rather than music when the call was put on hold. The software Music On Hold server on Cisco UCM is used in this scenario. Assume that the audio leg between the Music On Hold server and the held device uses G.711, and the relevant region relationship is configured for 64 kbps. What is the cause of the issue?

  A. The bandwidth configured for this region relationship is too low and must be increased to 96 kbps or higher.
  B. The device that is placed on hold does not support G.711, and a transcoder could not be allocated for the call.
  C. Cisco UCM is sending a=inactive to the held device.
  D. The Music On Hold server does not support G.711 and a transcoder could not be allocated for the call.
  C. Cisco UCM is sending a=inactive to the held device.

• Question 112:

  Refer to the exhibit.

  

  Building A Building B

  Results Summary Results Summary Calling Party Information Calling Party Information Dialed Digits = 9195552388 Dialed Digits = 9195552388 Match Result = RouteThisPattern Match Result = RouteThisPattern Matched Pattern Information Matched Pattern Information Called Party Number = 9195552388 Called Party Number = 9195552388 Time Zone = Etc/GMT Time Zone = Etc/GMT End Device = PSTN_RL End Device = PSTN_RL Call Classification = OffNet Call Classification = OffNet InterDigit Timeout = NO InterDigit Timeout = NO Device Override = Disabled Device Override = Disabled Outside Dial Tone = NO Outside Dial Tone = NO

  Call Flow Call Flow Route Pattern: Pattern = Route Pattern: Pattern = [2-9]XX[2-9]XXXXXX [2-9]XX[2-9]XXXXXX Route List: Route List Name = Route List: Route List Name = PSTN_RL PSTN_RL RouteGroup:RouteGroupName = RouteGroup:RouteGroupName = Standard Local Route Group Standard Local Route Group (RTP_trunks) PreTransform Calling Party PreTransform Calling Party Number = 2305 Number = 2304 PreTransform Called Party PreTransform Called Party Number = 9195552388 Number = 9195552388 Calling Party Transformations Calling Party Transformations Called Party Transformations Called Party Transformations Device :Type = SIPTrunk Device :Type = SIPTrunk

  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. Change the partition of the route pattern
  B. Add a sip trunk inside route group Standard Local Route Group.
  C. Modify the route pattern to add a prefix of 91
  D. Add a local route group on the device pool configuration.
  B. Add a sip trunk inside route group Standard Local Route Group.

• Question 113:

  Which built-in bridge configuration option can be set on the individual IP phones to ensure that the cluster- wide service parameter is used?

  A. automatic
  B. default
  C. on
  D. off
  B. default

• Question 114:

  Refer to the exhibit.

  

  Received UPDATE sip:192.168.100.101:5060;transport=udp SIP/20.0 Via: SIP/2.0/UDP 192.168.200.101:5060;branch= From: "Amy" ;tag= To: "Bob" ;tag= Call-ID: [email protected] Max-Forwards: 70 Timestamp: 123456789 Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Cseq: 101 UPDATE Contact: Min-SE: 2000 P-Asserted-Identity: "Joe" Content-Length: 0

  An engineer is troubleshooting an issue where inbound Calls are failing after they transferred. The provider reports that update is not supported, and this is causing the calls to fail. Which command should resolve this issue?

  A. no midcall-signaling passthru
  B. no update-callerId
  C. no contact-passig
  D. rel1xx require "100rel"
  B. no update-callerId

• Question 115:

  Refer to the exhibit.

  dial-peer voice 120 voip description ******Dial Peer Outgoing to offshore**** translation-profile outgoing OUT max-bandwidth 240 destination-pattern 32T progress_ind alert enable 8 progress_ind progress enable 8 progress_ind connect enable 8 session protocol sipv2 voice-class codec 1 voice-class sip bind control source-interface GigabitEthernet0/0.200 dtmf-relay rtp-nte no vad

  An engineer configured Call Admission Control on Cisco Unified Border Element to reject SIP calls when the aggregate bandwidth that is required for the calls exceeds the threshold. However, users complain of dropped calls when they place calls on hold during peak periods.

  Which solution resolves this challenge?

  A. Bind media to the loopback interface
  B. Bind media to interface GigabitEthernet0/0.200
  C. Enable the midcall-exceed feature on the dial peer
  D. Enable the error-code-override feature on the dial peer
  C. Enable the midcall-exceed feature on the dial peer

• Question 116:

  An administrator manages an environment with mobile users who commonly move between locations and take IP phones to remote sites. When mobile users are at one of the sites, users report issues with call quality, such as the audio cutting in and out. In addition, the affected area has a low bandwidth connection compared to other sites, so the administrator has put these devices in a region with an inter-region restriction of 8 kbps per call. Which feature must the administrator configure to mitigate the issue?

  A. Location Bandwidth Manager
  B. Extension Mobility
  C. Device Mobility
  D. Extension Mobility Cross Cluster
  A. Location Bandwidth Manager

• Question 117:

  A Cisco Unified Border Element receives SIP INVITEs but does not match any incoming dial peer. Which dial-peer command is required to match calls based on the called number?

  A. destination-pattern
  B. incoming called-number
  C. answer-address
  D. session-target
  B. incoming called-number

  ---

  The incoming called-number command is used to match inbound VoIP calls based on the dialed number.

• Question 118:

  A collaboration engineer is configuring call pickup for a Cisco UCM user that currently has a working BLF speed dial button for the teammate's extension. The user wants to see calls ringing on the teammate's extension and to answer those calls by pressing the BLF speed dial button. Which configuration step meets this requirement?

  A. Set the call pickup group to for both extensions.
  B. Select the call pickup checkbox on the BLF speed dial settings.
  C. Assign both extensions to the same partition.
  D. Assign the extensions to different call pickup groups.
  B. Select the call pickup checkbox on the BLF speed dial settings.

• Question 119:

  An administrator is configuring Meet-me conferencing in a Cisco UCM deployment and has created the Meet- me number and ensured that it is in a partition accessible by all devices. Which two additional steps must the administrator perform? (Choose two.) Update the softkey template on all phones to ensure that they contain the Meet-me softkey.

  A. Ensure that conferencing-initiating devices are using a media resource group list that contains at least one Cisco UCM conference bridge.
  B. Enable Meet-me conferencing In enterprise parameters.
  C. Ensure that all devices have G.729 enabled.
  D. Disable Early Media on the SIP profile of all devices that will use Meet-me conferencing.
  A. Ensure that conferencing-initiating devices are using a media resource group list that contains at least one Cisco UCM conference bridge.
  B. Enable Meet-me conferencing In enterprise parameters.

• Question 120:

  An engineer configures several calling party transformation patterns in Cisco UCM so that calling numbers are shown in the +E.164 format. The route patterns match seven-digit internal numbers and prefix the numbers with 1408. The SIP trunk matches 1408XXXXXXX and prefixes the number with +. When users make outbound calls, the calling number still shows as seven digits. What is the cause of this problem?

  A. The calling transformation on the SIP trunk is applied first, and then the calling transformation on the route patterns is applied.
  B. Calling transformations cannot be configured on both route patterns and a SIP trunk.
  C. The calling transformation on the route patterns is ignored because there is already a calling transformation on the SIP trunk.
  D. The engineer must also apply a calling transformation at the route group details level.
  C. The calling transformation on the route patterns is ignored because there is already a calling transformation on the SIP trunk.