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Cisco 300-815 Dumps Questions [2022] Pass for 300-815 Exam

Updated Cisco Study Guide 300-815 Dumps Questions

Cisco 300-815 Exam Syllabus Topics:

Topic Details
Topic 1
  • Troubleshoot These Globalized Call Routing Elements In Cisco Unified Communications Manager
Topic 2
  • Configure These Globalized Call Routing Elements In Cisco Unified Communications Manager
Topic 3
  • Mid-Call Signaling (Hold
  • Resume, Call Transfer, Conferencing)
  • Configure SIP SRST Gateway
Topic 4
  • Configure These Advanced Cisco Unified CME Features
  • Voice Translation Rules And Profiles
Topic 5
  • Troubleshoot Cisco Unified Communications Manager Mobility
  • Troubleshoot These H.323 Protocol Elements
Topic 6
  • Configure ILS, URI Synchronization, And GDPR
  • CME
  • SRST Gateway Technologies
Topic 7
  • Header And SDP Manipulation With SIP Profiles
  • Voice Translation Rules And Profiles
Topic 8
  • Configure Cisco Unified Communications Manager Express For SIP Phone Registration
  • Cisco Unified Border Element
Topic 9
  • Configure Supplementary Functions
  • Troubleshoot These Elements Of A SIP Conversation
Topic 10
  • Troubleshoot These Cisco Unified Border Element Dial Plan Elements
  • Implement Toll Fraud Prevention
Topic 11
  • Configure These Cisco Unified Border Element Dial Plan Elements
  • Configure Cisco Unified CME Dial Plans

As for the exam content that you should master to be able to clear the certification test, it is essential to know that Cisco 300-815 covers 6 topics in total. Each of them has a different weight, but it doesn’t mean that you can study for some of them in a slipshod manner. It is vital to pay attention to all the sections equally. All in all, this certification exam comes with the following subject areas:

  • Signaling & Media Protocols (20%)

    To deal with this first section, you should have some knowledge of troubleshooting processes. Thus, you have to understand how to troubleshoot the elements of SIP conversation, including UPDATE, session timers, mid-call signaling (conferencing, call transfer, hold/resume), PRACK, and early media. Also, the potential candidates should have the skills in troubleshooting media establishment and H.323 protocol elements, including DTMF as well as call set up and tear down.

  • Cisco Unified CM Call Control Features (20%)

    Within this domain, the students will need to be able to configure hunt groups, time of day routing, call queuing, as well as GDPR, URI synchronization, and ILS. Other subtopics measured within this exam part will evaluate your skills in troubleshooting Call Admission Control (exclude RSVP) and configuring the additional functions, such as call pick-up, meet-me, and call park.

  • Call Control & Dial Planning (25%)

    This area has the largest weight out of the whole exam syllabus. It covers a wide topic about the configuration and troubleshooting of the call routing elements that are globalized in the Cisco Unified Communications Manager. These elements include SIP trunking, TEHO, standard local route group, transformation patterns, SIP route patterns, route patterns, and translation patterns.

 

QUESTION 63
A network engineer designs a new dial plan and wants to block a certain range of numbers (8135100 through
8135105). What is the most specific route pattern that can be configured to block only the numbers in this range?

 
 
 
 

QUESTION 64
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?

 
 
 
 

QUESTION 65
What is a component of Cisco Unified Mobility?

 
 
 
 

QUESTION 66
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.)

 
 
 
 
 

QUESTION 67
You see the voice register pool 1 command in your Cisco Unified Communications Manager Express configuration. Which configuration is occurring in this section?

 
 
 
 

QUESTION 68
Which section under the Real-Time Monitoring Tool allows for reviewing the call flow and signaling for a SIP call in real time?

 
 
 
 

QUESTION 69
An engineer must configure a secure SIP trunk with a remote provider, with a specific requirement to use port 5065 for inbound and otubound traffic. Which two items must be configured to complete this configuration? (Choose two.)

 
 
 
 
 

QUESTION 70
For s SIP to SIP call flow, when does Cisco Unified Border Element require transcoding resources for DTMF?

 
 
 
 

QUESTION 71
The Cisco Unified Communications Manager Dialed Number Analyzer allows analysis of calls from which two devices? (Choose two.)

 
 
 
 
 

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?

 
 
 
 

QUESTION 73
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.)

 
 
 
 
 

QUESTION 74
What is first preference condition matched in a SIP-enabled incoming dial peer?

 
 
 
 

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.)

 
 
 
 

QUESTION 76
Which section under the Real-Time Monitoring Tool allows for reviewing the call flow and signaling for a SIP call in real time?

 
 
 
 

QUESTION 77
Refer to the exhibit.

Which change to the translation rule is needed to strip only the leading 9 from the digit string 9123548?

 
 
 
 

QUESTION 78
If all patterns below are configured in Cisco Unified Communications Manager which would be used when dialing the pattern “123”?

 
 
 
 

QUESTION 79
Which configuration must an administrator perform to display Translation Pattern operations in Cisco Unified Communications Manager SDL traces?

 
 
 
 

QUESTION 80
Refer to the exhibit.

An administrator is troubleshooting why users are not hearing audio when dialing long distance numbers across their Cisco Unified Border Element. The customer’s carrier has a requirement that dialing long distance requires an access code to be entered. Looking at the exhibit, what two actions can be taken to correct signaling? (Choose two.)

 
 
 
 
 

QUESTION 81
How does an engineer globalize routing for ingress calls coming from the PSTN to internal DNs?

 
 
 
 

QUESTION 82
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?

 
 
 
 

QUESTION 83
A network engineer designs a new dial plan and wants to block a certain range of numbers (8135100 through
8135105). What is the most specific route pattern that can be configured to block only the numbers in this range?

 
 
 
 

QUESTION 84
End users at a new site report being unable to hear the remote party when calling or being called by users at headquarters. Calls to and from the PSTN work as expected. To investigate the SIP signaling to troubleshoot the problem, which field can provide a hint for troubleshooting?

 
 
 
 

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