Best Quality 300-815 Exam Questions Cisco Test To Gain Brilliante Result!
Preparations of 300-815 Exam 2023 CCNP Collaboration Unlimited 120 Questions
Overview of the Exam Outline
The content of 300-815 exam contains six domains, and questions taken from them will pop up during the test. However, they won't appear evenly, and some sections take up a larger portion of the syllabus than others. Take a look at them below:
- Cisco Unified Border Element
This section takes about 15% of the content. It covers the basic skills in handling the procedures involved setting up as well as troubleshooting the CU-BE dial plan components, including DTMF, voice translation rules and profiles, codec preference list, plus signaling and media bindings.
- Mobility
About 10% of the exam questions come from this topic. To tackle these items, you must understand the concepts of setting up CU-CM mobility, including unified mobility, extension mobility, and device mobility, and troubleshooting them.
- CME/SRST Gateway Technologies
About 10% of the questions come from this domain. It assesses one’s knowledge of how to configure CU-CME for SIP telephone registration and dial plans. Also, you should have a solid understanding of how to enforce toll fraud prevention and configure the advanced CU-CME features and SIP SRST gateway.
NEW QUESTION 47
Refer to the exhibit.
How many maximum hops can an ILS update traverse?
- A. 0
- B. 1
- C. 2
- D. 3
Answer: B
NEW QUESTION 48
Which two types of distribution algorithm are within a line group? (Choose two.)
- A. bottom up
- B. random
- C. highest preference
- D. top down
- E. circular
Answer: D,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 49
Drag and drop the steps from the left into the order to provision mobility users through LDAP on the right. Not all options are used.
Answer:
Explanation:
NEW QUESTION 50
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 System Parameters in Cisco Unified Communications Manager and change the range of media ports to 20000-22000.
- B. Go to the SIP profile assigned to these IP phones in Cisco Unified CM 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 ports to TCP.
- E. Ask the firewall administrator to change the range of UDP ports to 16384-32767.
Answer: C,E
NEW QUESTION 51
What is first preference condition matched in a SIP-enabled incoming dial peer?
- A. answer-address
- B. incoming uri
- C. target carrier-id
- D. incoming called-number
Answer: B
Explanation:
Section: Signaling and Media Protocols
Explanation
Explanation/Reference: https://www.cisco.com/c/en/us/support/docs/voice/ip-telephony-voice-over-ip-voip/211306-In- Depth-Explanation-of-Cisco-IOS-and-IO.html#anc8
NEW QUESTION 52
Which two configuration parameters are prerequisites to set Native Call Queuing on Cisco Unified Communications Manager? (Choose two.)
- A. The phone button template must have the Queue Status Softkey configured.
- B. A unicast music on hold audio source must be configured.
- C. The maximum number of callers allowed in queue must be 10.
- D. Cisco RIS data collector service must be running on the same server as the Cisco CallManager service.
- E. Cisco IP Voice Media Streaming Service must be activated on at least one node in the cluster.
Answer: D,E
NEW QUESTION 53
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)#allow-connections voice-mail mod - 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)#no supplementary-service sip moved-temporarily
Answer: C
Explanation:
Section: Call Control and Dial Planning
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 54
Which two statements are correct with respect to the Client Matter Code setting in the route pattern configuration? (Choose two.)
- A. If you check the Allow Overlap Sending check box, you can also check the Require Client Matter Code check box.
- B. If you check the Allow Overlap Sending check box, the Require Client Matter Code check box becomes disabled.
- C. 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.
- D. The Client Matter Code has the option to configure Authorization Level such as in the Forced Authorization 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: B,C
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 55
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 needs to enable OPTIONS pings on the SIP trunks for both clusters.
- B. The administrator must allow connectivity so that TCP connections do not fail between the nodes.
- C. The administrator must associate the route pattern with a calling search space the device can dial.
- D. The administrator needs to disable OPTIONS pings on the SIP trunks for both clusters.
Answer: B
NEW QUESTION 56
Which description of RTP timestamps or sequence numbers is true?
- A. Timestamps increase by the time "carrying" by a packet.
- B. The sequence number is used to detect losses.
- C. The timestamp is used to place the incoming audio and video packets in the correct timing order (playout delay compensation).
- D. Sequence numbers increase by four for each RTP packet transmitted.
Answer: C
Explanation:
Section: Signaling and Media Protocols
Explanation/Reference: https://www.cs.columbia.edu/~hgs/rtp/faq.html
NEW QUESTION 57
An administrator is trying to apply configuration changes on Cisco CME. When the users registered on Cisco CME to dial a local number to a PSTN call, the Cisco CME sends an incorrect number of digits. What translation rule fixes the issue and sends the correct number of digits?
- A. voice translation-rule 1
rule 1 /^4...V /2404\0/ type any subscriber plan any isdn - B. voice translation-rule 1
rule 1 /^4...$/2404\0/ type any national plan any Isdn - C. voice translation-rule 1 rule 1 /^4...S/ /9132404 0/ type any subscriber plan any Isdn
- D. voice translation-rule 1 rule 1 // // type any subscriber plan any isdn
Answer: A
NEW QUESTION 58
Refer to the exhibit.
Outbound calls to the service provider cause intermittent errors due to a codec mismatch. The internal network sends early offer SDP that contains only G.711 A-law. The service provider reports that some destinations support only G.711 A-law while others support only iLBC. The service provider also allows only 20 active calls at a time Which configuration allows successful media negotiation for all calls using outbound dial peers 5002 and 5003?
- A. Option D
- B. Option A
- C. Option B
- D. Option C
Answer: A
NEW QUESTION 59
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. Add the mobility key to the softkey template that the desk phone is using.
- B. Confirm that the desk phone is subscribed to Cisco Extension Mobility.
- C. Use Dialed Number Analyzer to determine if the user extension can dial the mobile phone.
- D. Check to make sure that the Resume softkey option appears on the desk phone.
Answer: A
NEW QUESTION 60 
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 /^\+1\([2-9]..[2-9]......$\)/ /\0/
- B. rule 1/^\+1\([2-9]..[2-9]......$\)/ /\1/
- C. rule 1 /^\([2-9]..[2-9]......$\)/ /\1/
- D. rule 1 /^/+\([^1].*\)/ /011\1/
Answer: B
Explanation:
Section: Cisco Unified Border Element
Explanation/Reference:
NEW QUESTION 61
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 music on hold and the MOH audio is chosen from Phone_A Network Hold MOH Audio Source settings.
- B. 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.
- C. Phone_B sends a SIP-REFER message to the Cisco Unified CM with Phone_C information in the Refer-To section.
- D. 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.
- E. Phone_A sends a SIP-REFER message to the Cisco Unified Communications Manager with Phone_C information in the Refer-To section.
Answer: A,E
NEW QUESTION 62
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. On the SIP Profile, the configuration parameter SIP Rel1XX Options must be set to Send PRACK for all
1xx Messages. - C. Accept Audio Codec Preferences in Received Offer must be set to On.
- D. Allow Passthrough of Configured Line Device Caller Information must be enabled.
Answer: B
NEW QUESTION 63
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. incoming called number 555.......
- B. codec g729
- C. session-protocol sipv2
- D. answer-address 555 ........
Answer: A
NEW QUESTION 64
Calls to a particular extension are not routing to voicemail. The user reaches the voicemail system from the handset by pressing the Messages button Which configuration parameter causes this problem?
- A. The voicemail pilot number for call forwarding is missing from the ephone
- B. The voicemail pilot number is missing from the call handling on Cisco Unity Express
- C. The voicemail pilot number is missing from the telephony service configuration on Cisco UCME
- D. The voicemail pilot number for call forwarding is missing from the ephone-dn
Answer: D
NEW QUESTION 65
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