300-815 100% Pass Guaranteed Download CCNP Collaboration Exam PDF Q&A
300-815 Practice Test Dumps with 100% Passing Guarantee
Cisco 300-815 certification exam is designed for IT professionals who are responsible for implementing and managing advanced call control and mobility services in a Cisco collaboration solution. 300-815 exam covers topics such as configuring and troubleshooting Cisco Unified Communications Manager, implementing and troubleshooting Cisco Collaboration applications, and deploying and troubleshooting Cisco voice gateways and media resources. Candidates who pass the exam will have the knowledge and skills required to design, deploy, and troubleshoot advanced call control and mobility services in a Cisco collaboration solution.
NEW QUESTION # 29
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. Enable Send send-receive SDP in mid-call INVITE on the SIP profile.
- B. Enable SIP ReMXX Options on the SIP profile.
- C. Change Session Refresh Method on the SIP profile to INVITE.
- D. Increase Retry INVITE to 20 seconds on the SIP profile.
Answer: C
NEW QUESTION # 30
An administrator troubleshoots call failure in a new deployment and finds that the SIP INVITE messages sent to the service provider contain a diversion header with the user's 4-digit directory number. These 4-digit directory numbers range from 1000 to 9999. The service provider is rejecting the calls because it requires that the diversion header contain 10 digits. Which command on the Cisco Unified Border Element resolves this issue for all users?
A)
B)
- A. Option B
- B. Option C
- C. Option A
- D. Option D
Answer: D
NEW QUESTION # 31
When you troubleshoot H.323 call setup, which message informs you that the called party is being notified about the call?
- A. RINGING
- B. PROCEEDING
- C. CONNECT
- D. ALERTING
Answer: D
Explanation:
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cata/186_188/3_0/english/administration/g uide/h323/h32330ad/h3288APH.pdf
NEW QUESTION # 32
Users are reporting that several inter-site calls are failing, and the message "not enough bandwidth" is showing on the display. Voice traffic between locations goes through corporate WAN. and Call Admission Control is enabled to limit the number of calls between sites.
How is the issue solved without increasing bandwidth utilization on the WAN links?
- A. Disable Call Admission Control and let the calls use the amount of bandwidth they require.
- B. Configure Call Queuing so that the user waits until there is bandwidth available
- C. Reroute all calls through the PSTN and avoid using WAN.
- D. Configure AAR to reroute calls that are denied by Call Admission Control through the PSTN.
Answer: D
NEW QUESTION # 33
Exhibit.
Refer to the exhibit. All calls from site A to site B are failing, and the issue has been identified as a media negotiation problem. Which configuration change resolves the issue?
- A. Enable Early Offer on the SIP trunk.
- B. Create a new audio codec preference list with G.711 U-law 64k as the highest priority and apply it to RTP_Reg and SJ_Reg.
- C. Increase the bandwidth allowance between the RTP_Reg and SJ_Reg regions to 64 kbps.
- D. Disable G.722 on all devices at both sites.
Answer: C
NEW QUESTION # 34
Refer to the exhibit.
An administrator is troubleshooting a problem in which some outbound calls from an internal network to the Internet telephony service provider are not getting connected, but some others connect successfully. The firewall team found that some call attempts on port 5060 came from an unrecognized IP that has not been defined in the firewall rule. What should the administrator configure in the Cisco Unified Border Element to fix this issue?
- A. ip prefix-list to filter the unwanted IP address
- B. use of port 5061 for SIP secure
- C. access list allowing the firewall IP
- D. bind signaling and media to the loopback interface
Answer: D
NEW QUESTION # 35
Refer to the exhibit. A collaboration engineer is troubleshooting an issue where external callers cannot leave voicemail messages. Also, internal users report hearing the reorder tone (fast busy) when they attempt to retrieve voicemail messages from their Cisco IP phones. Which action resolves the issue?
- A. Ensure that Cisco UCM can resolve the destination address via DNS.
- B. Verify that the correct port numbers are used for the SIP trunk.
- C. Ensure that the SIP Trunk Security Profile is configured to use UDP for transport.
- D. Start the Cisco Call Manager service at the destination.
Answer: A
NEW QUESTION # 36
Which configuration element of a hunt group allows for changing Calling Party Transformations settings?
- A. hunt pilot
- B. route group
- C. line group
- D. hunt list
Answer: A
Explanation:
https://community.cisco.com/t5/ip-telephony-and-phones/call-alerting-on-hunt-group-as-shared- line/td-p/2658015
NEW QUESTION # 37
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 "national"
- B. calling number 011-1-613-555-1234 and numbering type "subscriber"
- C. calling number 011613-555-1234 and numbering type "international"
- D. calling number 613-555-1234 and numbering type "subscriber"
Answer: A
NEW QUESTION # 38
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_A sends a SIP-REFER message to the Cisco Unified Communications Manager with Phone_C information in the Refer-To section.
- 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,C
NEW QUESTION # 39
Refer to the exhibit.
An engineer is troubleshooting an issue with the caller not hearing a PSTN announcement before the SIP call has completed setup. How must the engineer resolve this issue using the reliable provisional response of the SIP?
- A. voice service voip sip send 180 sdp
- B. sip-ua
disable-early-media 180 - C. voice service voip sip no reMxx
- D. voice service voip sip rehxx require 100rel
Answer: D
NEW QUESTION # 40
An administrator manages an environment with mobile users who commonly move between locations and take IP phones to remote sites. When mobile users are at one of the sites, users report issues with call quality, such as the audio cutting in and out. In addition, the affected area has a low bandwidth connection compared to other sites, so the administrator has put these devices in a region with an inter-region restriction of 8 kbps per call. Which feature must the administrator configure to mitigate the issue?
- A. Device Mobility
- B. Extension Mobility
- C. Location Bandwidth Manager
- D. Extension Mobility Cross Cluster
Answer: C
NEW QUESTION # 41
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 Ack
- B. H.245 Terminal Capability Set
- C. H.225 Connect
- D. H.245 Open Logical Channel
Answer: A
Explanation:
H245 Open Logical Channel contains only RTCP port, H245 Open Logical Channel ACk will have both RTP and RTP Port.
https://community.cisco.com/t5/collaboration-voice-and-video/h323-call-flow/ta-p/3160014
NEW QUESTION # 42
Refer to the exhibit.
Refer to the exhibit. Users report that outgoing calls do not work on the new SIP trunk for outgoing calls. The solution consists of a Cisco UCM Cluster linked to a Cisco Unified Border Element where the SIP trunk is terminated. The provider required 10 digits. The logs show a line going toward the Cisco Unified Border Element. Which code snippet must be added to the configuration to meet the requirement?
- A. request Invite sip-header modify "<sip:1(...)@" "<sip:9135551\1@" under the voice translation profile configuration
- B. request Invite sip-header Diversion modify "<sip:1(...)@" "<sip:9135551\1@" under the SIP profile configuration
- C. request Invite sip-header modify "<sip:1(...)@" "<sip:9135551\1@" under the SIP translation profile configuration
- D. sip-header modify "<sip:1(...)@" "<sip:9135551\1@" under the voice translation profile configuration
Answer: B
NEW QUESTION # 43
When route patterns are defined as precisely as possible on a Cisco UCM, the control and reliability of the calling plan is increased. A route pattern must be added to support calls to this number range: 9135200 to 9135205. The calls are on-net and no translation patterns are configured. Which configuration meets these requirements?
- A. 913520X
- B. 913520[012345]
- C. 913520!
- D. 913520[1-5]
Answer: B
NEW QUESTION # 44
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. 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. Ask the firewall administrator to change the range of UDP ports to 16384-32767.
Answer: D,E
Explanation:
Section: Signaling and Media Protocols
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 # 45
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 // // type any subscriber plan any isdn - B. voice translation-rule 1
rule 1 /^4...$/ /9132404\0/ type any subscriber plan any isdn - C. voice translation-rule 1
rule 1 /^4...$/ /2404\0/ type any subscriber plan any isdn - D. voice translation-rule 1
rule 1 /^4...$/ /2404\0/ type any national plan any isdn
Answer: B
NEW QUESTION # 46
When the services key is pressed Cisco Extension Mobility does not show up. What is the cause of the issue?
- A. Cisco Extension Mobility Service is not running.
- B. The URL configured for Cisco Extension Mobility is not correct.
- C. The phone is not subscribed to Cisco Extension Mobility Service.
- D. Cisco Extension Mobility is not enabled in the Phone Configuration Window (Device > Phone)
Answer: C
NEW QUESTION # 47
Refer to the exhibit.
ILS has been configured between two hubs using this configuration. The hubs appear to register successfully, but ILS is not functioning as expected. Which configuration step is missing?
- A. Trust certificates for ILS have not been installed on the clusters
- B. The Cluster IDs have not been set to unique values
- C. A password has never been set for ILS.
- D. Use TLS Certificates must be selected.
Answer: B
NEW QUESTION # 48
......
300-815 PDF Dumps Are Helpful To produce Your Dreams Correct QA's: https://www.exam4docs.com/300-815-study-questions.html
New 300-815 exam Free Sample Questions to Practice: https://drive.google.com/open?id=11yft4hFclHV2u2Wp5evmBhUW7F09XzcX
