300-815 Free Update With 100% Exam Passing Guarantee [2021]
[Dec-2021] Verified Cisco Exam Dumps with 300-815 Exam Study Guide
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:
- 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.
- 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.
- 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 Cisco 300-815 CLACCM is one of the required exams for earning the CCNP Collaboration designation. This exam validates the candidate's knowledge of call control and mobility services in addition to other related topics that will be covered below.
NEW QUESTION 52
Which two types of authentication are supported for the configuration of Intercluster Lookup Service?
(Choose two.)
- A. passwords
- B. username and secret key
- C. TLS certificates
- D. TokenID
- E. FQDN of the servers defined in DNS
Answer: A,C
NEW QUESTION 53
How does an engineer globalize routing for ingress calls coming from the PSTN to internal DNs?
- A. At Cisco Unified Communications Manager, put the calling number in E.164 format and the called number in E.164 format.
- B. At Cisco Unified CM, put the calling number in E.164 format and the called number in PSTN format.
- C. At the PSTN gateway, put the calling number in PSTN format and the called number in DN format.
- D. At the PSTN gateway, put the calling number in E.164 format and the called number in localized (DN) format.
Answer: A
NEW QUESTION 54
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 C
- B. Option B
- C. Option E
- D. Option A
- E. Option D
Answer: A,E
NEW QUESTION 55
An engineer is troubleshooting local ringback on a Cisco SIP gateway The gateway is not ignoring the SIP 180 response with SDP from the service provider, and the far end device is sending the 180 with SDP to play ringback from the IP address specified m the SDP Which configuration change must be made on the gateway to resolve the issue?
- A. Router(con(-voi-serv)# no disable-early-media 180
- B. Router(conftg-sip-ua)# disable-early-media 180
- C. Router(conf-voi-serv)# dlisable-early-media 180
- D. Router(config-sip-ua)# no disable-early-media 180
Answer: B
NEW QUESTION 56
A company has users that are logged in to hunt groups. However, there is a requirement for hunt group configurations to provide an option to turn on audible ringtones when calls to a line group arrive at a phone that is logged out and on a break. This ringtone alerts a logged-out user that there is an incoming call to a hunt list to which the line is a member, but the call does not ring at the phone of that line group member because of the logged-out status. Which action meets this requirement?
- A. Set the service parameter Enterprise Feature Access number for hunt group logout and set up an access number
- B. Set the service parameter Party Entrance Tone to True."
- C. Configure the service parameter hunt group logoff notification and specify the name of the ringtone file.
- D. Configure the HLog softkey on the phone so that while a user is logged off, it plays an audible tone when a call is missed.
Answer: C
NEW QUESTION 57
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. Allow: header if the 200 OK response
- B. c= line of SDP content
- C. o= line of SDP content
- D. Contact: header of the 200 OK response
Answer: B
NEW QUESTION 58
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?
- A. SIP binding in SIP configuration mode:
voice service voip sip
bind control source-interface GigabitEthernetO/0/0 bind media source-interface GigabitEthernetO/0/0 - B. SIP binding in dial-peer configuration mode:
dial-peer voice 100 volp
voice-class sip bind control source-interface GigabitEthernetO/0/0
voice-class sip bind media source-interface GigabitEthernetO/0/0 - C. SIP binding In SIP configuration mode:
voice service volp
sip
bind control source-Interface GlgabltEthernetO/0/1 bind media source-Interface GlgabltEthernetO/0/1 - D. SIP binding In dial-peer configuration mode:
dial-peer voice 300 voip
voice-class sip bind control source-interface GigabitEthernetO/0/1 voice-class sip bind media source-interface GigabitEthernetO/0/1
Answer: B
NEW QUESTION 59
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 Dialed Number Analyzer service.
- C. Restart the subscriber
- D. Activate the Cisco CallManager service.
- E. Activate the Cisco Extended Functions service.
Answer: A,B
NEW QUESTION 60
What is first preference condition matched in a SIP-enabled incoming dial peer?
- A. answer-address
- B. incoming called-number
- C. target carrier-id
- D. incoming uri
Answer: D
NEW QUESTION 61
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 the dial peer of the gateway has the correct destination pattern configured.
- B. Verify that the route pattern is not blocking calls to the destination number.
- C. Verify that the route pattern has the correct calling-party transformation mask
- D. Verify that IP routing is correct between the gateway and the IP phone.
Answer: A
NEW QUESTION 62
Which description of RTP timestamps or sequence numbers is true?
- A. The sequence number is used to detect losses.
- B. Sequence numbers increase by four for each RTP packet transmitted.
- C. The timestamp is used to place the incoming audio and video packets in the correct timing order (playout delay compensation).
- D. Timestamps increase by the time "carrying" by a packet.
Answer: C
NEW QUESTION 63
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. Go to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 20000-22000.
- B. Ask the firewall administrator to change the range of UDP ports to 16384-32767.
- 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 ports to TCP.
- E. Go to System Parameters in Cisco Unified Communications Manager and change the range of media ports to 20000-22000.
Answer: B,C
Explanation:
Reference:
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/port/9_1_1/ CUCM_BK_T2CA6EDE_00_tcp-port-usage-guide-91/CUCM_BK_T2CA6EDE_00_tcp-port-usage-guide- 91_chapter_01.html
NEW QUESTION 64
The administrator of ABC company is troubleshooting a one-way audio issue for a call that uses H.323 protocol (slow-start mode). The administrator requests that you provide the IP and port information of the Real-Time Transport Protocol traffic that had the one-way audio call.
You gather the H.225 and H.245 messages for one of the one-way audio calls. Where can you find the RTP IP and port information for both sides? (Note: This call flow has not invoked any media resources like MTP or transcoders).
- A. H.245 Open Logical Channel
- B. H.225 Connect
- C. H.245 Open Logical Channel Ack
- D. H.245 Terminal Capability Set
Answer: A
Explanation:
Section: Signaling and Media Protocols
Explanation/Reference: http://ccievoicehopeful.blogspot.com/2012/09/h323-notes.html
NEW QUESTION 65
An administrator is implementing a new dial-plan on Cisco Unified Border Element. The administrator must ensure that incoming dial-peers are matched based on the IP address from where the incoming request originates. Which dial-peer configuration should be applied to accomplish this requirement?
- A. dial-peer voice 1 voip
incoming url to - B. dial-peer voice 1 voip
incoming url via - C. dial-peer voice 1 voip
incoming url request - D. dial-peer voice 1 voip
incoming called-number
Answer: B
NEW QUESTION 66
In Cisco Unified Communications Manager, which tool do you use to check SIP traces?
- A. CCSIP
- B. MTP
- C. OS Administration Page
- D. RTMT
Answer: D
Explanation:
Section: Call Control and Dial Planning
NEW QUESTION 67
Which description of RTP timestamps or sequence numbers is true?
- A. The sequence number is used to detect losses.
- B. Sequence numbers increase by four for each RTP packet transmitted.
- C. The timestamp is used to place the incoming audio and video packets in the correct timing order (playout
- D. Timestamps increase by the time "carrying" by a packet.
Answer: C
Explanation:
delay compensation).
NEW QUESTION 68
Where on Cisco Unified Communications Manager do you configure the standard local route group for a group of devices?
- A. Call Routing > Route/Hunt > Local Route Group Names
- B. Call Routing > Emergency Location > Emergency Location (ELIN) Groups
- C. System > Location Info
- D. System > Device Pool
Answer: A
NEW QUESTION 69
Which services are needed to successfully implement Cisco Extension Mobility in a standalone Cisco Unified Communications Manager server?
- A. Cisco Extended Functions, Cisco Extension Mobility, and Cisco AXL Web Service
- B. Cisco TAPS Service, Cisco TFTP, and Cisco Extension Mobility
- C. Cisco CallManager, Cisco TFTP, and Cisco CallManager SNMP Service
- D. Cisco CallManager, Cisco TFTP, and Cisco Extension Mobility
Answer: D
Explanation:
Reference:
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/10_5_2/ccmfeat/ CUCM_BK_C3A84B33_00_cucm-feature-configuration-guide_1052/CUCM_BK_C3A84B33_00_cucm-feature- configuration-guide_chapter_011101.html#CUCM_TK_A337E035_00
NEW QUESTION 70
Which two types of authentication are supported for the configuration of Intercluster Lookup Service? (Choose two.)
- A. passwords
- B. username and secret key
- C. TLS certificates
- D. TokenID
- E. FQDN of the servers defined in DNS
Answer: A,C
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 71
Refer to the exhibit.
An engineer is troubleshooting an issue where inbound calls to Cisco UCM with early media fail to establish. While investigating the issue, the engineer finds that Cisco UCM is set to require a PRACK. but the Cisco Unified Border Element is not sending it. Which command is causing this issue?
- A. voice-class sip early-media update block
- B. voice-class sip rel1xx disable
- C. voice-class sip conn-reuse
- D. voice-class midcall-signaling block
Answer: B
NEW QUESTION 72
An engineer has two cisco UCM Clusters and wants to integrate them using ILS with TLS certificates. Cluster A (pub and 1 subscriber) will be the hub, and Cluster B (pub and 1 subscriber) will be the spoke. Both Clusters have self-signed certificates. The engineer has exchanged Publisher A and subscriber B Tomcat certificates, but the connection fails. What is the cause of the failure?
- A. The tomcat certificate from Cluster B must be the publisher.
- B. Cluster IDs are not unique.
- C. The password is incorrect.
- D. The engineer needs to exchange the CallManager certificate.
Answer: A
NEW QUESTION 73
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 Information section of the SIP Trunk configuration.
- B. Destination Port in SIP Trunk Security Profile configuration
- C. Incoming Port in SIP Trunk Security Profile configuration
- D. Incoming Port in Security Information of the SIP Profile configuration.
- E. Destination Port in SIP Information section of the SIP Trunk configuration
Answer: C,E
NEW QUESTION 74 
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 Dialed Number Analyzer service.
- C. Restart the subscriber
- D. Activate the Cisco CallManager service.
- E. Activate the Cisco Extended Functions service.
Answer: A,B
Explanation:
Section: Call Control and Dial Planning
NEW QUESTION 75
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?
- A. calling number 613-555-1234 and numbering type "subscriber"
- B. calling number 011-1-613-555-1234 and numbering type "subscriber"
- C. calling number 613-555-1234 and numbering type "national"
- D. calling number 011613-555-1234 and numbering type "international"
Answer: C
NEW QUESTION 76
Which configuration must an administrator perform to display Translation Pattern operations in Cisco Unified Communications Manager SDL traces?
- A. Set up the Digit Analysis Complexity in Service Parameters for Cisco Unified CM to TranslationAndAlternatePatternAnalysis.
- B. Enable the Detailed Call Analysis option under Enterprise Parameters for Unified CM.
- C. By default, the Translation Patterns operations are printed in SDL traces, so no additional configuration is necessary.
- D. Check the Translation Patterns Analysis check box in Micro Traces on the Cisco Unified CM Serviceability page.
Answer: B
Explanation:
Reference:
https://community.cisco.com/t5/collaboration-voice-and-video/taking-sip-call-trace-on-cisco-unified- cm-using-rtmt/ta-p/3161200
NEW QUESTION 77
......
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.
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