300-815 Free Exam Study Guide! (Updated 120 Questions) [Q67-Q83]

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300-815 Free Exam Study Guide! (Updated 120 Questions)

300-815 Dumps for CCNP Collaboration Certified Exam Questions & Answer


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With Cisco’s Training and Certification, you may enhance your career with a new learning portfolio that offers both developers and network engineers chances. Cisco’s training and certification programmed provides a straight route to your technological professional objectives. IT technology led to the evolution of Cisco’s training and certification programmers to educate instructors, engineers and software developers for success in the industry’s most significant jobs. Explore the power of the Cisco Learning Network’s dynamic culture to start your certification and lifetime learning objectives. Obtain valuable IT training resources for all Cisco certifications. Access IT certification research tools, CCNA practice tests, IT salary and IT job findings. The purpose of this exam guide is to assist you assess if you can successfully finish Cisco IP network troubleshooting and support 300-815 exam test. This guide provides information on the test target audience, required preparation and documentation, as well as a complete list of examination aims, to help you get a degree. We strongly advised to study the 300-815 dumps and 300-815 practice exams for increased chances of passing the exam.


Certification Path

After earning the CCNP Collaboration designation, you may want to further develop your tech skills. This way, the most sensible move for you is to pursue another Cisco certificate of the expert level, which is the CCIE Collaboration validation. Having such a certification will equip you with the relevant skills to manage high-level collaboration solutions within an enterprise.


About the Exam

To be eligible, you have to be 18 years or older. If your age group falls between 13-17, you can take part in the test provided your parents or legal guardians support you. However, you're not eligible if you're below 13 years of age.

The exam can be taken in English and requires one to pay a registration fee of $300. The questions are expected to be provided in different forms, including MSQs, fill-in-the-blanks, testlets, and others. The time limit for this test is 1,5 hours.

300-815 is administered by Pearson VUE and can be taken at its testing centers. You can also opt for online format, and any location is fine for taking the test as long as it's quiet and private. In both cases, you will be under close supervision throughout the exam duration. The online format involves high-level supervision through the OnVUE software, so you should have it along with a webcam. Remember to provide an ID that is government-issued. If you decide to go for 300-815 at the center, you should have two valid and self-signed identification documents.

 

NEW QUESTION 67
For s SIP to SIP call flow, when does Cisco Unified Border Element require transcoding resources for DTMF?

  • A. interworking between an OOB method and RFC2833 for flow-through calls
  • B. interworking between an OOB method and RFC2833 for flow-around calls
  • C. interworking between h245-alpha numeric and sip-kpml
  • D. interworking between h245-signal and rtp-nte

Answer: B

Explanation:
Reference:
https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-border- element/200412-DTMF-Relay-and-Interworking-on-CUBE.html#anc35

 

NEW QUESTION 68
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. Imported Dial Plan Replication Database Object Lower Limit
  • B. Global Data Service Parameter Limit
  • C. ILS Max Number of Learned Objects in Database
  • D. ILS Active Learned Object Upper Limit

Answer: C

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 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.)

  • A. Incoming Port in SIP Trunk Security Profile configuration
  • B. Incoming Port in SIP Information section of the SIP Trunk configuration.
  • C. Incoming Port in Security Information of the SIP Profile configuration.
  • D. Destination Port in SIP Information section of the SIP Trunk configuration
  • E. Destination Port in SIP Trunk Security Profile configuration

Answer: A,D

Explanation:
Section: Call Control and Dial Planning

 

NEW QUESTION 70
Refer to the exhibit.

While troubleshooting call failures on the Cisco Unified Border Element, an administrator notices that messages are being sent to the service provider, but there is no response The administrator later learns that this SIP provider does not support PRACK. Which header should be removed from the SIP message to resolve this issue?

  • A. Contact <sip:[email protected]:5060>
  • B. Content-Type: application/sdp
  • C. Content-Disposition: session:handling=required
  • D. Require 100rel

Answer: D

 

NEW QUESTION 71
An engineer is configuring Cisco UCM lo forward parked calls back to the user who parked the call if it is not retrieved after a specified time interval. Which action must be taken to accomplish this task?

  • A. Configure device pools.
  • B. Configure enterprise softkeys.
  • C. Configure class of control.
  • D. Configure service parameters

Answer: D

 

NEW QUESTION 72
After configuring a Cisco CallManager Express with Cisco Unity Express, inbound calls from the PSTN SIP trunk receive a ring tone for 20 seconds and then a busy signal instead of voicemail. Which configuration fixes this problem?

  • A. Router(config)#dial-peer voice 2 voip
    Router(config-dial-peer)#no vad
  • B. Router(config)# voice service voip
    Router(conf-voi-serv)#no supplementary-service sip moved-temporarily
  • C. Router(config)# voice service voip
    Router(conf-voi-serv)#allow-connections h323 to h323
  • D. Router(config)# voice service voip
    Router(conf-voi-serv)#allow-connections voice-mail mod

Answer: C

Explanation:
Reference:
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cusrst/admin/sccp_sip_srst/configuration/ guide/SCCP_and_SIP_SRST_Admin_Guide/srst_call_handling.html

 

NEW QUESTION 73
Refer to the exhibit.

An engineer configures Cisco Unified Border Element to connect the enterprise VoIP network with a SIP telephony provider. Calls are not working in either direction. What must be configured in the dial peer 1 to fix the issue?

  • A. session-protocol sipv2
  • B. codec g729
  • C. incoming called number 555.......
  • D. answer-address 555 ........

Answer: A

 

NEW QUESTION 74
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 one or more time periods.
  • B. A partition can have one time schedule assigned. A time schedule contains only one time period.
  • C. A partition can have multiple time schedules 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: A

 

NEW QUESTION 75
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. Subscribe the phone to the Cisco Extension Mobility service.
  • C. Create the end user and associate it to the device profile.
  • D. Assign the extension as a mobile extension.

Answer: B

 

NEW QUESTION 76
Which two extended capabilities must be configured on dial peers for fast start-to-early media scenarios (H.323 to SIP interworking)? (Choose two.)

  • A. DTMF
  • B. AUDIO
  • C. FAX
  • D. VIDEO
  • E. BFCP

Answer: A,E

 

NEW QUESTION 77
Refer to the exhibit.

How many maximum hops can an ILS update traverse?

  • A. 0
  • B. 1
  • C. 2
  • D. 3

Answer: D

 

NEW QUESTION 78
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?

  • A. voice-class sip bind control source-interface Loopback0
  • B. voice-class sip resources priority mode passthrough
  • C. voice-class sip early-offer forced.
  • D. voice-class sip bind media source-interface Loopback0

Answer: D

 

NEW QUESTION 79
What is a component of Cisco Unified Mobility?

  • A. Mobile Connect
  • B. Smart Client Support
  • C. Unified IVR
  • D. Single Number Connect

Answer: A

 

NEW QUESTION 80
A user reports that when they attempt to log out from the Cisco Extension Mobility service by pressing the Services button, they cannot log out. What is the most likely cause of this issue?

  • A. The user device profile is not subscribed to the Cisco Extension Mobility service.
  • B. The Cisco Extension Mobility service has not been configured on the phone.
  • C. The user device profile has not been assigned to the user.
  • D. There might be a significant delay between the button being pressed and the Cisco Extension Mobility service recognizing it. It would be best to check network latency.

Answer: A

 

NEW QUESTION 81
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.)

  • A. Incoming Port in SIP Trunk Security Profile configuration
  • B. Incoming Port in SIP Information section of the SIP Trunk configuration.
  • C. Incoming Port in Security Information of the SIP Profile configuration.
  • D. Destination Port in SIP Information section of the SIP Trunk configuration
  • E. Destination Port in SIP Trunk Security Profile configuration

Answer: A,D

 

NEW QUESTION 82

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.)

  • A. As soon as Phone_A presses the Transfer button for the first time, Phone_B hears the MOH and the MOH audio is chosen from Phone_B User Hold MOH Audio Source settings.
  • B. Phone_A sends a SIP-REFER message to the Cisco Unified Communications Manager with Phone_C information in the Refer-To section.
  • C. As soon as Phone_A presses the Transfer button for the first time, Phone_B hears the MOH and the MOH audio is chosen from Phone_A User Hold MOH Audio Source settings.
  • D. As soon as Phone_A presses the Transfer button for the first time, Phone_B hears the music on hold and the MOH audio is chosen from Phone_A Network Hold MOH Audio Source settings.
  • E. Phone_B sends a SIP-REFER message to the Cisco Unified CM with Phone_C information in the Refer-To section.

Answer: A,B

Explanation:
Section: Signaling and Media Protocols

 

NEW QUESTION 83
......

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