300-815 Study Guide Brilliant 300-815 Exam Dumps PDF [Q35-Q57]

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300-815 Study Guide Brilliant 300-815 Exam Dumps PDF

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Overview of the Exam Outline

The content of 300-815 exam contains six domains, and questions taken from them will pop up during the test. However, they won’t appear evenly, and some sections take up a larger portion of the syllabus than others. Take a look at them below:

  • Cisco Unified Border Element

    This section takes about 15% of the content. It covers the basic skills in handling the procedures involved setting up as well as troubleshooting the CU-BE dial plan components, including DTMF, voice translation rules and profiles, codec preference list, plus signaling and media bindings.

  • Signaling and Media Protocols

    This module covers about 20% of the test. It is required that you know how to troubleshoot the various elements of a SIP dialogue, including session timers, mid-call signaling, and more. You also need to equip yourself with a clear idea of troubleshooting certain protocol elements and media establishment.

  • Cisco Unified CM Call Control Features

    This module takes about 20% of the syllabus. The subdomains include troubleshooting of the Call Admission Control and configuring ILS, URI synchronization, GDPR, hunt groups, call queuing, time of day routing, and supplementary roles.

  • Mobility

    About 10% of the exam questions come from this topic. To tackle these items, you must understand the concepts of setting up CU-CM mobility, including unified mobility, extension mobility, and device mobility, and troubleshooting them.

  • Call Control and Dial Planning

    This domain has weight of about 25%. You are to have a general idea of troubleshooting and setting up the globalized call routing elements in CU-CM, such as the patterns for route, translation, transformation, and others.

To be eligible for the exam, candidates must have a valid Cisco CCNP Collaboration or CCIE Collaboration certification. They must also have a minimum of three years of experience in designing, deploying, and managing Cisco collaboration and communication solutions.

 

Q35. Which section under the Real-Time Monitoring Tool allows for reviewing the call flow and signaling for a SIP call in real time?

 
 
 
 

Q36. A customer is using a SIP trunk to route calls to ITSP to decrease the possibility of downtime, the customer invested in a failover device How does the customer ensure reachability to ITSP, so that if one device on ITSP fails, the calls will be routed to another device?

 
 
 
 

Q37.

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?

 
 
 
 

Q38. Which call pickup feature allows users to pick up incoming calls in a group that is associated with their own group?

 
 
 
 

Q39. 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?

 
 
 
 

Q40. 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?

 
 
 
 

Q41. Which configuration must an administrator perform to display Translation Pattern operations in Cisco Unified Communications Manager SDL traces?

 
 
 
 

Q42. Refer to the exhibit.

An engineer is troubleshooting a call-establishment problem between Cisco Unified Border Element and Cisco UCM. Which command set corrects the issue?

 
 
 
 

Q43. 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?

 
 
 
 

Q44. 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?

 
 
 
 

Q45. 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?

 
 
 
 

Q46. Configure Call Queuing in Cisco Unified Communications Manager. Where do you set the maximum number of callers in the queue?

 
 
 
 

Q47. 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.)

 
 
 
 
 

Q48. An engineer has temporarily disabled toll fraud prevention for SIP line calls on a Cisco CME12.6x and must enforce security and toll fraud prevention for the SIP line side on Cisco Unified CME. Which configuration must be used to start this process?

 
 
 
 

Q49. What is first preference condition matched in a SIP-enabled incoming dial peer?

 
 
 
 

Q50. 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?

 
 
 
 

Q51. 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?

 
 
 
 

Q52. For s SIP to SIP call flow, when does Cisco Unified Border Element require transcoding resources for DTMF?

 
 
 
 

Q53. 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?

 
 
 
 

Q54. 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.)

 
 
 
 
 

Q55. Refer to the exhibit.

An administrator is troubleshooting a problem in which some outbound calls from an internal network to the Internet telephony service provider are not getting connected, but some others connect successfully. The firewall team found that some call attempts on port 5060 came from an unrecognized IP that has not been defined in the firewall rule. What should the administrator configure in the Cisco Unified Border Element to fix this issue?

 
 
 
 

Q56. 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?

 
 
 
 

Q57. Which description of RTP timestamps or sequence numbers is true?

 
 
 
 

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