The Best 300-815 Exam Study Material and Preparation Test Question Dumps
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Cisco 300-815 Exam Certification Details:
| Duration | 90 minutes |
| Number of Questions | 55-65 |
| Recommended Training | Implementing Cisco Advanced Call Control and Mobility Services (CLACCM) |
| Exam Registration | PEARSON VUE |
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. Early Offer for G Clear Calls must be enabled.
- B. Accept Audio Codec Preferences in Received Offer must be set to On.
- C. Allow Passthrough of Configured Line Device Caller Information must be enabled.
- D. On the SIP Profile, the configuration parameter SIP Rel1XX Options must be set to Send PRACK for all 1xx Messages.
Answer: D
Explanation:
Section: Signaling and Media Protocols
Explanation
NEW QUESTION 39
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. The engineer needs to exchange the CallManager certificate.
- C. Cluster IDs are not unique.
- D. The password is incorrect.
Answer: A
NEW QUESTION 40
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 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 - B. SIP binding In SIP configuration mode:
voice service volp
sip
bind control source-Interface GlgabltEthernetO/0/1 bind media source-Interface GlgabltEthernetO/0/1 - C. SIP binding in SIP configuration mode:
voice service voip sip
bind control source-interface GigabitEthernetO/0/0 bind media source-interface GigabitEthernetO/0/0 - 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: A
NEW QUESTION 41
Which IOS command creates a SIP-enabled dial peer?
- A. dial peer voice 20 sip
- B. voice dial-peer 20 sip
- C. dial-peer voice 20 voip
- D. dial-peer voice 20 pots
Answer: C
Explanation:
Section: Cisco Unified Border Element
Explanation/Reference: https://www.ciscopress.com/articles/article.asp?p=664148&seqNum=6
NEW QUESTION 42
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 the PSTN gateway, put the calling number in E.164 format and the called number in localized (DN) format.
- C. At the PSTN gateway, put the calling number in PSTN format and the called number in DN format.
- D. At Cisco Unified CM, put the calling number in E.164 format and the called number in PSTN format.
Answer: A
NEW QUESTION 43
Refer to the exhibit.
Users report that outbound PSTN calls from phones registered to Cisco Unified Communications Manager are not completing. The local service provider in North America has a requirement to receive calls in 10-digit format. The Cisco Unified CM sends the calls to the Cisco Unified Border Element router in a globalized E.164 format. There is an outbound dial peer on Cisco Unified Border Element configured to send the calls to the provider. The dial peer has a voice translation profile applied in the correct direction but an incorrect voice translation rule applied, which is shown in the exhibit. Which rule modified DNIS in the format that the provider is expecting?
- A. rule 1 /^\([2-9]..[2-9]......$\)/ /\1/
- B. rule 1/^\+1\([2-9]..[2-9]......$\)/ /\1/
- C. rule 1 /^\+1\([2-9]..[2-9]......$\)/ /\0/
- D. rule 1 /^/+\([^1].*\)/ /011\1/
Answer: B
NEW QUESTION 44
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. Assign the extension as a mobile extension.
- B. Create the end user and associate it to the device profile.
- C. Create the default device profiles for all phone models that are used.
- D. Subscribe the phone to the Cisco Extension Mobility service.
Answer: D
NEW QUESTION 45
The SIP session refresh timer allows the RTP session to stay active during an active call. The Cisco UCM sends either SIP-INVITE or SIP-UPDATE messages in a regular interval of time throughout the active duration of the call. During a troubleshooting session, the engineer finds that the Cisco UCM is sending SIP-UPDATE as the SIP session refresher, and the engineer would like to use SIP-INVITE as the session refresher. What configuration should be made in the Cisco UCM to achieve this?
- A. Change Session Refresh Method on the SIP profile to INVITE.
- B. Enable SIP ReMXX Options on the SIP profile.
- C. Increase Retry INVITE to 20 seconds on the SIP profile.
- D. Enable Send send-receive SDP in mid-call INVITE on the SIP profile.
Answer: A
NEW QUESTION 46
What is first preference condition matched in a SIP-enabled incoming dial peer?
- A. incoming uri
- B. target carrier-id
- C. incoming called-number
- D. answer-address
Answer: A
NEW QUESTION 47
A user's phone is already configured for Single Number Reach, and the user wants a feature to move an active call from a mobile phone to a desk phone and vice-vers a. As an administrator, which additional configuration should be made to fulfill the user's request?
- A. Use Dialed Number Analyzer to determine if the user extension can dial the mobile phone.
- B. Check to make sure that the Resume softkey option appears on the desk phone.
- C. Add the mobility key to the softkey template that the desk phone is using.
- D. Confirm that the desk phone is subscribed to Cisco Extension Mobility.
Answer: C
NEW QUESTION 48
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?
- A. 81XXXXX
- B. 813510[12345]
- C. 813510[012345]
- D. 813510[^0-5]
Answer: C
Explanation:
Section: Call Control and Dial Planning
NEW QUESTION 49
Which section under the Real-Time Monitoring Tool allows for reviewing the call flow and signaling for a SIP call in real time?
- A. System > Tools > Trace and Log Central
- B. Voice/Video > Session Trace Log View > Real Time Data
- C. Analysis Manager > Inventory > Trace File Repositories
- D. Voice/Video > Session Trace Log View > Open From Local Disk
Answer: B
NEW QUESTION 50
Which configuration element of a hunt group allows for changing Calling Party Transformations settings?
- A. hunt pilot
- B. line group
- C. hunt list
- D. route group
Answer: A
NEW QUESTION 51
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. codec g729
- B. answer-address 555 ........
- C. incoming called number 555.......
- D. session-protocol sipv2
Answer: D
NEW QUESTION 52
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. Early Offer for G Clear Calls must be enabled.
- B. Accept Audio Codec Preferences in Received Offer must be set to On.
- C. Allow Passthrough of Configured Line Device Caller Information must be enabled.
- D. On the SIP Profile, the configuration parameter SIP Rel1XX Options must be set to Send PRACK for all 1xx Messages.
Answer: D
Explanation:
Section: Signaling and Media Protocols
NEW QUESTION 53
Which two statements are correct with respect to the Client Matter Code setting in the route pattern configuration? (Choose two.)
- A. The Client Matter Code has the option to configure Authorization Level such as in the Forced Authorization Code.
- B. If you check the Allow Overlap Sending check box, you can also check the Require Client Matter Code check box.
- C. If you check the Allow Overlap Sending check box, the Require Client Matter Code check box becomes disabled.
- D. The Client Matter Code feature does not support overlap sending because the Cisco Unified CM cannot determine when to prompt the user for the code.
- E. The Client Matter Code feature does support overlap sending because the Cisco Unified Communications Manager can determine when to prompt the user for the code.
Answer: C,D
Explanation:
Reference:
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/10_0_1/ccmfeat/ CUCM_BK_F3AC1C0F_00_cucm-features-services-guide-100/CUCM_BK_F3AC1C0F_00_cucm-features- services-guide-100_chapter_010000.pdf
NEW QUESTION 54
An engineer must configure call queuing under a Hunt Pilot. After the engineer receives the audio file that will be played to callers during queuing, which two steps should be taken to complete the configuration? (Choose two.)
- A. Upload the audio file in "TFTP File Management" via OS Administration GUI
- B. Assign the uploaded audio file to "Network Hold MOH Source & Announcements" under Hunt Pilot's Queuing section.
- C. Assign the uploaded audio file to the hunting Line Group member's "Network Hold MOH Audio Source".
- D. Assign the uploaded audio file to the hunting Line Group member's "User Hold MOH Audio Source
- E. Upload the audio file in "MOH Audio File Management" via CM Administration GUI
Answer: A,B
NEW QUESTION 55
Which description of RTP timestamps or sequence numbers is true?
- A. The timestamp is used to place the incoming audio and video packets in the correct timing order (playout delay compensation).
- B. Timestamps increase by the time "carrying" by a packet.
- C. The sequence number is used to detect losses.
- D. Sequence numbers increase by four for each RTP packet transmitted.
Answer: A
Explanation:
Reference:
https://www.cs.columbia.edu/~hgs/rtp/faq.html
NEW QUESTION 56
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