[UPDATED Feb-2023] Best Value Available Preparation Guide for 300-815 Exam
1 Full 300-815 Practice Test and 120 Unique Questions, Get it Now!
Target Audience and Prerequisites
Every Cisco certification path can be taken by the individuals with any level of technical background, and this option is not an exception. This means that you can go for the Cisco 300-815 test right away if you want to. However, it is important to understand that this is a professional-level exam, so it will be much easier for you if you have some prior experience. The applicants for this test are recommended to possess from 3 to 5 years of experience to get started with the certification process. They are advised to have a general understanding of networking, video, and voice technologies, know how to describe various codecs as well as transform analogue voice into digital streams, be familiar with the Cisco IOS XE command line, possess the relevant skills in configuring and modifying the requirements within Cisco Unified CM, and so on.
The most important task for you to perform before going for the Cisco 300-815 exam is to pass the Cisco 350-801 test first. This is the core exam, which will help you understand the basics of the domain you are planning to enter. And then you can proceed with preparation for the concentration test. Also, please note that if you don’t want to obtain the CCNP Collaboration certificate, you can simply clear Cisco 300-815 and get the specialist-level certification.
Cisco 300-815 Exam Topics:
| Section | Weight | Objectives |
|---|---|---|
| Call Control and Dial Planning | 25% | - Configure these globalized call routing elements in Cisco Unified CommunicationsManager
-Troubleshoot these globalized call routing elements in Cisco Unified CommunicationsManager
|
| Signaling and Media Protocols | 20% | - Troubleshoot these elements of a SIP conversation
-Troubleshoot these H.323 protocol elements
|
| Cisco Unified Border Element | 15% | -Configure these Cisco Unified Border Element dial plan elements
-Troubleshoot these Cisco Unified Border Element dial plan elements
|
How to book the Cisco 300-815: Implementing Cisco Advanced Call Control and Mobility Services Exam
You must take the following procedures to take part in the Cisco IP Networks (300-815) Examination:
- Step 1: Click here for the Pearson VUE website
- Step 2: Sign in or establish an account
- Step 3: enter the examination number 300-135
- Step 4: Follow on-line directions
- Step 5: Taking 300-815 practice exams
- Step 6: Pay for examination with a credit card or examination voucher
NEW QUESTION 36
A single site reports that when they dial select numbers, the call connects, but they do not get audio. The administrator finds that the calls are not routing out of the normal gateway but out of another site's gateway due to a TEHO configuration. What is the next step to diagnose and solve the issue?
- A. Verify that IP routing is correct between the gateway and the IP phone.
- B. Verify that the dial peer of the gateway has the correct destination pattern configured.
- C. Verify that the route pattern has the correct calling-party transformation mask
- D. Verify that the route pattern is not blocking calls to the destination number.
Answer: B
NEW QUESTION 37
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?
- A. Enable transmit security status on the SIP security profile
- B. Enable SIP Option Ping on the SIP profile.
- C. Monitor the link using network management toots, and if it fails, manually change the routing to another working device.
- D. Enable ANAT on the SIP profile.
Answer: B
NEW QUESTION 38
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?
- A. On the SIP Profile, the configuration parameter SIP Rel1XX Options must be set to Send PRACK for all
1xx Messages. - B. Allow Passthrough of Configured Line Device Caller Information must be enabled.
- C. Accept Audio Codec Preferences in Received Offer must be set to On.
- D. Early Offer for G Clear Calls must be enabled.
Answer: A
NEW QUESTION 39
An engineer must route all SIP calls in the form of <user>@example.com to the SIP trunk gateway corporate local. Which two SIP route patterns can be used to accomplish this task? (Choose two.)
- A. *@example.com
- B. *.*
- C. example.com
- D. gateway.corporate.local
- E. [email protected]
Answer: A,B
Explanation:
Section: Call Control and Dial Planning
NEW QUESTION 40
In Cisco Unified Communications Manager globalized call routing is implemented and must confirm that it is correctly implemented without making a call. Which tool do you use for verification?
- A. SDI trace
- B. SDL trace
- C. Real-Time Monitoring Tool
- D. Dialed Number Analyzer
Answer: D
NEW QUESTION 41
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. answer-address 555 ........
- B. session-protocol sipv2
- C. incoming called number 555.......
- D. codec g729
Answer: C
NEW QUESTION 42
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?
- A. o= line of SDP content
- B. Allow: header if the 200 OK response
- C. Contact: header of the 200 OK response
- D. c= line of SDP content
Answer: A
NEW QUESTION 43
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. missing Call Admission Control
- B. ptime mismatch
- C. codec mismatch
- D. phone class of service issue
Answer: A
NEW QUESTION 44
Which configuration must an administrator perform to display Translation Pattern operations in Cisco Unified Communications Manager SDL traces?
- A. By default, the Translation Patterns operations are printed in SDL traces, so no additional configuration is necessary.
- B. Enable the Detailed Call Analysis option under Enterprise Parameters for Unified CM.
- C. Set up the Digit Analysis Complexity in Service Parameters for Cisco Unified CM to TranslationAndAlternatePatternAnalysis.
- D. Check the Translation Patterns Analysis check box in Micro Traces on the Cisco Unified CM Serviceability page.
Answer: A
NEW QUESTION 45
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 range of UDP ports to 16384-32767.
- B. Go to System Parameters in Cisco Unified Communications Manager and change the range of media ports to 20000-22000.
- C. Ask the firewall administrator to change the ports to TCP.
- D. Go to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 16384-32767
- E. Go to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 20000-22000.
Answer: A,D
NEW QUESTION 46
You see the voice register pool 1 command in your Cisco Unified Communications Manager Express configuration. Which configuration is occurring in this section?
- A. configuration for a single SIP phone
- B. configuration for SIP registrar service
- C. configuration for a pool of SIP phones (similar to device pool on Cisco Unified Communications Manager)
- D. configuration items common for all SIP phones
Answer: A
NEW QUESTION 47
Why would RTP traffic that is sent from the originating endpoint fail to be received on the far endpoint?
- A. The far end connection data (c=) in the SDP was overwritten by deep packet inspection in the call signaling path.
- B. Cisco Unified Communications Manager invoked media termination point resources.
- C. A firewall in the media path is blocking TCP ports 16384-32768.
- D. The RTP traffic is arriving beyond the jitter buffer on the receiving end.
Answer: C
Explanation:
Section: Signaling and Media Protocols
NEW QUESTION 48
An administrator configured Cisco Unified Mobility to block access to remote destinations for certain caller IDs. A user reports that a blocked caller was able to reach a remote destination. Which action resolves the issue?
- A. Configure an access list.
- B. Configure Single Number Reach.
- C. Configure a mobility identity.
- D. Configure Mobile Voice Access.
Answer: A
NEW QUESTION 49
An administrator is troubleshooting call failures on an H.323 gateway via the CLI. To see signaling for media and call setup, which two debugs must the administrator turn on? (Choose two).
- A. debug H.323 message
- B. debug H.225 asn1
- C. debug H.323 asn1
- D. debug H.225 media
- E. debug H.245 asn1
Answer: B,E
NEW QUESTION 50
Which action is correct with respect to toll fraud prevention configuration in the Cisco Unified Communications Manager Express?
- A. Configure IP Address Trusted Authentication for Incoming VoIP Calls.
- B. Configure Direct Inward Dial for Incoming ISDN Calls with overlap dialing.
- C. Configure the command no ip address trusted authenticate under "voice service voip".
- D. Enable Secondary Dial tone on Analog and Digital FXO Ports.
Answer: A
Explanation:
Reference:
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/admin/configuration/manual/ cmeadm/cmetoll.html#concept_ECC4F4E7ED0F45C594B703EEF34762F2
NEW QUESTION 51
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. Phone_A sends a SIP-REFER message to the Cisco Unified Communications Manager with Phone_C information in the Refer-To section.
- B. 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.
- 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_B User Hold MOH Audio Source settings.
- D. Phone_B sends a SIP-REFER message to the Cisco Unified CM with Phone_C information in the Refer-To section.
- E. 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.
Answer: A,B
NEW QUESTION 52
Where is the dtmf-relay command configured on Cisco Unified Border Element?
- A. in the VoIP or POTS dial peers
- B. in global SIP configuration
- C. in the VoIP dial peer
- D. in the voice-class VoIP configuration
Answer: C
Explanation:
Section: Cisco Unified Border Element
Explanation/Reference: https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/dtmf- relay.html
NEW QUESTION 53
When locations-based Call Admission Control denies the call, which two masks can AAR apply when routing the call through the PSTN? (Choose two.)
- A. AAR destination mask
- B. +E.164 alternate number mask
- C. external phone number mask
- D. enterrise alternate number mask
- E. called party transform mask
Answer: A,C
Explanation:
Section: Cisco Unified CM Call Control Features
Explanation/Reference: https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/collab10/collab10/dialplan.html
NEW QUESTION 54
An engineer is troubleshooting Cisco Device Mobility and find that the phone has roamed to a building that is assigned to a different device pool but has not changed its device pool accordingly What action resolves the issue?
- A. Set correct Location under Current Device Mobility Settings
- B. Set the correct subnet under Device Mobility Info.
- C. Enable SRST under Current Device Mobility Settings
- D. Set Device CSS under Current Device Mobility Settings.
Answer: B
NEW QUESTION 55
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