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What Topics Does 300-815 Assess?
Earning a passing score in such a Cisco exam will require the candidate to have a thorough understanding of all the test parts that are organized in the following exam domain:
- Cisco Unified CM Call Control Features
To get through the fifth section of the exam that takes up 20%, the applicant should know the concept of Call Admission Control and must be able to troubleshoot it. However, this doesn't include RSVP technology. This section also includes various technologies such as ILS, URI, and GDPR, and the applicant should be able to skillfully configure them. Furthermore, the entrant's knowledge of hunt groups, call queuing, and the time of day routing and their configuration will be tested under such an objective. Finally, the applicant should be capable of configuring various supplementary functions such as call park, meet-me, and call pick-up.
- Dial planning & call control
The fourth portion of this exam covers 25% of the content and consists of questions regarding the Cisco Unified Communications Manager. The candidate should be capable of configuring various globalized call routing components of the Cisco Unified Communications Manager. These components include translation patterns, route patterns, transformation patterns, TEHO, SIP trunking, and standard local routing group. Moreover, the candidates must be able to troubleshoot all these globalized elements for call routing.
- Media protocols & signaling
Signaling and media protocols are the first exam domain that covers a total of 20% of all the questions. The candidate's knowledge and understanding of the elements of SIP conversation will be tested in this section. The entrant is expected to know about the concepts and skillfully describe early media, PRACK, the mid-call signaling like call transfer, session timers, and UPDATE. The applicant should also be capable of troubleshooting various H.323 protocol elements that include DTMF, call set up, and tear down. The comprehension of the media establishment and its troubleshooting will also be beneficial to get through in this portion.
- Mobility
The final domain deals with Unified Communications Manager Mobility by Cisco and makes up 10% of the exam content. Therefore, the entrant should know how to configure unified mobility, extension mobility, and device mobility. Moreover, it is also essential for the candidate to be familiar with successfully troubleshooting the aforementioned mobility technologies.
- The element for Cisco Unified Border
The third section of the exam includes 15% of all questions regarding the Cisco Unified Border Element technology. Thus, the entrant should be capable of configuring various Cisco Unified Border Element dial plan parts such as DTMF, voice translations rules & profiles, the list of codec preferences, dial peers, headers, and the manipulation of SDP with diverse SIP profiles. This domain also deals with troubleshooting all the aforementioned Cisco Unified Border Element dial plan components.
- Gateway technologies for CME/SRST
The second objective takes up 10% of all test questions and deals with the Cisco Unified Communications Manager Express. What is more, the entrant needs to be aware of how to configure the aforementioned technologies for SIP phone registration. In addition to that, the applicant should be capable of configuring Cisco Unified CME dial plans along with implementing the prevention of toll fraud. Furthermore, this domain tests the candidate's capability in configuring various advanced Cisco Unified CME features that include Hunt groups, Call park, and Paging. At last, under such a category, you'll get to know more about the gateway for SIP SRST and its configuration.
NEW QUESTION 70
If all patterns below are configured in Cisco Unified Communications Manager which would be used when dialing the pattern "123"?
- A. 1XX (urgent Priority Set)
- B. 12[2-5]
- C. 12X (urgent priority set)
- D. 12!
Answer: C
Explanation:
Section: Call Control and Dial Planning
NEW QUESTION 71
Why would RTP traffic that is sent from the originating endpoint fail to be received on the far endpoint?
- A. Cisco Unified Communications Manager invoked media termination point resources.
- B. A firewall in the media path is blocking TCP ports 16384-32768.
- C. The RTP traffic is arriving beyond the jitter buffer on the receiving end.
- D. The far end connection data (c=) in the SDP was overwritten by deep packet inspection in the call signaling path.
Answer: B
Explanation:
Section: Signaling and Media Protocols
NEW QUESTION 72
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. BFCP
- B. FAX
- C. AUDIO
- D. DTMF
- E. VIDEO
Answer: A,D
Explanation:
Section: Signaling and Media Protocols
NEW QUESTION 73
Refer to the exhibit.
Calls incoming from the provider are not working through newly set up Cisco Unified Border Element.
Provider engineers get the 404 Not Found SIP message. Incoming calls are coming from the provider with called number "222333444" and Cisco Unified Communications Manager is expecting the called number to be delivered as "444333222". The administrator already verified that the IP address of the Cisco Unified CM is set up correctly and there are no dial peers configured other than those shown in the exhibit. Which action must the administrator take to fix the issue?
- A. Create specific matching for "222333444" on the incoming dial peer.
- B. Set up translation-profile on the incoming dial peer to match incoming traffic.
- C. Change the destination-pattern on the outgoing dial peer to match "444333222".
- D. Fix the voice translation-rule to match specifically number "222333444" and change it to "444333222".
Answer: B
NEW QUESTION 74
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 ports to TCP.
- B. Go to System Parameters in Cisco Unified Communications Manager and change the range of media ports to 20000-22000.
- C. Go to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 16384-32767
- D. Ask the firewall administrator to change the range of UDP 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: C,D
NEW QUESTION 75
Which two types of distribution algorithm are within a line group? (Choose two.)
- A. circular
- B. random
- C. bottom up
- D. highest preference
- E. top down
Answer: A,E
Explanation:
Section: Call Control and Dial Planning
Explanation/Reference: https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/9_0_1/ccmcfg/ CUCM_BK_CDF59AFB_00_admin-guide-90/CUCM_BK_CDF59AFB_00_admin-guide_chapter_0100011.html
NEW QUESTION 76
Which configuration element of a hunt group allows for changing Calling Party Transformations settings?
- A. route group
- B. hunt pilot
- C. line group
- D. hunt list
Answer: B
NEW QUESTION 77
Which section under the Real-Time Monitoring Tool allows for reviewing the call flow and signaling for a SIP call in real time?
- A. Voice/Video > Session Trace Log View > Open From Local Disk
- B. Voice/Video > Session Trace Log View > Real Time Data
- C. System > Tools > Trace and Log Central
- D. Analysis Manager > Inventory > Trace File Repositories
Answer: B
NEW QUESTION 78
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. enterrise alternate number mask
- B. called party transform mask
- C. +E.164 alternate number mask
- D. AAR destination mask
- E. external phone number mask
Answer: D,E
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 79
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-Disposition: session:handling=required
- C. Content-Type: application/sdp
- D. Require 100rel
Answer: D
NEW QUESTION 80
The Cisco Unified Communications Manager Dialed Number Analyzer allows analysis of calls from which two devices? (Choose two.)
- A. CTI ports
- B. IP phones
- C. CTI route points
- D. translation patterns
- E. device pools
Answer: A,B
NEW QUESTION 81
An IP Telephony administrator is deploying IP phones The administrator has an existing Cisco UCME router with several SCCP & SIP phones registered. The administrator receives a request for a new SIP phone with MAC address 1111 2222.3333 and directory number 2050 to be added in the Cisco UCME. Which two configurations should be added in CME to support this request? (Choose two )
- A. Option D
- B. Option B
- C. Option A
- D. Option E
- E. Option C
Answer: A,E
NEW QUESTION 82
Which action is correct with respect to toll fraud prevention configuration in the Cisco Unified Communications Manager Express?
- A. Configure the command no ip address trusted authenticate under "voice service voip".
- B. Enable Secondary Dial tone on Analog and Digital FXO Ports.
- C. Configure IP Address Trusted Authentication for Incoming VoIP Calls.
- D. Configure Direct Inward Dial for Incoming ISDN Calls with overlap dialing.
Answer: C
NEW QUESTION 83
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.)
- A. Activate the Cisco Dialed Number Analyzer Server service.
- B. Activate the Cisco CallManager service.
- C. Activate the Cisco Extended Functions service.
- D. Restart the subscriber
- E. Activate the Cisco Dialed Number Analyzer service.
Answer: A,E
NEW QUESTION 84
Why would RTP traffic that is sent from the originating endpoint fail to be received on the far endpoint?
- A. Cisco Unified Communications Manager invoked media termination point resources.
- B. A firewall in the media path is blocking TCP ports 16384-32768.
- C. The RTP traffic is arriving beyond the jitter buffer on the receiving end.
- D. The far end connection data (c=) in the SDP was overwritten by deep packet inspection in the call signaling path.
Answer: C
NEW QUESTION 85
Refer to the exhibit.
How many maximum hops can an ILS update traverse?
- A. 0
- B. 1
- C. 2
- D. 3
Answer: A
NEW QUESTION 86
Which two types of authentication are supported for the configuration of Intercluster Lookup Service? (Choose two.)
- A. username and secret key
- B. TLS certificates
- C. TokenID
- D. FQDN of the servers defined in DNS
- E. passwords
Answer: B,E
Explanation:
Reference:
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/11_5_1/ sysConfig/11_5_1_SU1/cucm_b_system-configuration-guide-1151su1/cucm_b_system-configuration-guide- 1151su1_chapter_011001.pdf
NEW QUESTION 87
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. incoming called number 555.......
- C. session-protocol sipv2
- D. codec g729
Answer: C
NEW QUESTION 88
Refer to the exhibit.
An administrator has configured a SIP trunk between two Cisco UCM clusters. For calls that should use the trunk, the calls fail with a fast busy. The administrator checks the Cisco CallManager SDL traces and found that the cluster to which the calling device is registered never sends an INVITE to the destination cluster. The administrator also verifies that all nodes from both clusters are powered on, and the CallManager service is running. How is this issue resolved?
- A. The administrator must associate the route pattern with a calling search space the device can dial.
- B. The administrator needs to disable OPTIONS pings on the SIP trunks for both clusters.
- C. The administrator must allow connectivity so that TCP connections do not fail between the nodes.
- D. The administrator needs to enable OPTIONS pings on the SIP trunks for both clusters.
Answer: C
NEW QUESTION 89
What are two configureation features of the Client matter code setting in the route pattern configuration? (Choose two.)
- A. Selecting the Allow Overlap Sending setting allows a user to select the Require Client Matter Code setting.
- B. Selecting the Allow Overlap Sending setting disables the Require Client Matter Code setting.
- C. The Client Matter Code feature provides the option to configure Authorization Level susch as in the Forced Authorization Code.
- D. The client Matter Code feature supports overlap sending since the Cisco UCM can determine when to prompt the user for the code.
Answer: B,C
NEW QUESTION 90
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