Try 300-815 Exam Valid Dumps with Instant Download Free Updates
300-815 Dumps First Attempt Guaranteed Success
Cisco 300-815 (Implementing Cisco Advanced Call Control and Mobility Services) Certification Exam is an important certification for IT professionals who are involved in implementing and managing advanced call control and mobility services using Cisco technologies. 300-815 exam is designed to test the knowledge and skills of candidates in a wide range of topics related to advanced call control and mobility services, and candidates who pass 300-815 exam will be recognized as experts in this field.
NEW QUESTION # 96
Which IOS command creates a SIP-enabled dial peer?
- A. dial-peer voice 20 voip
- B. dial-peer voice 20 pots
- C. voice dial-peer 20 sip
- D. dial peer voice 20 sip
Answer: A
NEW QUESTION # 97
Refer to the exhibit. A standard local route group is configured for long-distance calls. Calls from building A succeed, but calls from building B fail. On the system. Each building has is own device pool. The DNA tool is used to test the configuration. How is this issue resolved?
- A. Modify the route pattern to add a prefix of 91
- B. Change the partition of the route pattern
- C. Add a sip trunk inside route group Standard Local Route Group.
- D. Add a local route group on the device pool configuration.
Answer: C
NEW QUESTION # 98
Which two types of authentication are supported for the configuration of Intercluster Lookup Service? (Choose two.)
- A. TokenID
- B. TLS certificates
- C. username and secret key
- 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 # 99
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\([2-9]..[2-9]......$\)/ /\1/
- B. rule 1 /^/+\([^1].*\)/ /011\1/
- C. rule 1 /^\([2-9]..[2-9]......$\)/ /\1/
- D. rule 1 /^\+1\([2-9]..[2-9]......$\)/ /\0/
Answer: A
NEW QUESTION # 100
Which two descriptions of the Standard Local Route Group deployment are true? (Choose two.)
- A. can be assigned directly to the route pattern
- B. chooses the route group that is configured under the device pool of the calling-party device
- C. can be associated under the route group
- D. can be associated only under the route list
- E. chooses the route group that is configured under the device pool of the called-party device
Answer: B,D
NEW QUESTION # 101
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(config-sip-ua)# no disable-early-media 180
- B. Router(config-sip-ua)# disable-early-media 180
- C. Router(con(-voi-serv)# no disable-early-media 180
- D. Router(conf-voi-serv)# dlisable-early-media 180
Answer: B
NEW QUESTION # 102
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 service.
- B. Activate the Cisco Extended Functions service.
- C. Activate the Cisco Dialed Number Analyzer Server service.
- D. Restart the subscriber
- E. Activate the Cisco CallManager service.
Answer: A,C
NEW QUESTION # 103
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. Increase Retry INVITE to 20 seconds on the SIP profile.
- B. Change Session Refresh Method on the SIP profile to INVITE.
- C. Enable SIP ReMXX Options on the SIP profile.
- D. Enable Send send-receive SDP in mid-call INVITE on the SIP profile.
Answer: B
NEW QUESTION # 104
Refer to the exhibit. A Cisco Unified Border Element continues to send 180/183 with the required:
100rel header to Cisco UCM. and the call eventually disconnects
How is the issue resolved?
- A. Enable *Early Offer support for voice and video calls" on the SIP Profile Configuration Page in Cisco UCM.
- B. Disable "Send send-receive SDP in mid-call INVITE* on the SIP Profile Configuration Page in Cisco UCM.
- C. Enable 'SIP ReI1XX Options* and -Early Offer Support" on the SIP Profile Configuration Page in Cisco UCM.
- D. Disable "SIP Rel1XX Options* and 'Early Offer Support* on the SIP Profile Configuration Page in Cisco UCM.
Answer: A
NEW QUESTION # 105
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. Phone_B sends a SIP-REFER message to the Cisco Unified CM with Phone_C information in the Refer-To section.
- B. Phone_A sends a SIP-REFER message to the Cisco Unified Communications Manager with Phone_C information in the Refer-To section.
- C. 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.
- 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_B User Hold MOH Audio Source settings.
- 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: B,C
NEW QUESTION # 106
An administrator is configuring an Intercluster Lookup Service between 10 Cisco UCM clusters. Due to security requirements, certificate-based authentication must be used. Due to the deployment size, the administrator wants to avoid manually exchanging certificates between the clusters. Which two steps must be followed to meet these requirements? (Choose two)
- A. Ensure that the cluster ID for all clusters is the same.
- B. Enable password-based authentication in conjunction with certificate-based authentication.
- C. Install multiserver CA-signed Tomcat certificates on all cluster publishers.
- D. Install multiserver CA-signed CallManager certificates on all cluster publishers.
- E. Set all dusters to standalone on the ILS configuration page.
Answer: B,C
NEW QUESTION # 107
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. Phone_B sends a SIP-REFER message to the Cisco Unified CM with Phone_C information in the Refer-To section.
- B. Phone_A sends a SIP-REFER message to the Cisco Unified Communications Manager with Phone_C information in the Refer-To section.
- C. 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.
- 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_B User Hold MOH Audio Source settings.
- 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: B,C
NEW QUESTION # 108
You see the voice register pool 1 command in your Cisco Unified Communications Manager Express configuration. Which configuration is occurring in this section?
- A. configuration for a pool of SIP phones (similar to device pool on Cisco Unified Communications Manager)
- B. configuration items common for all SIP phones
- C. configuration for SIP registrar service
- D. configuration for a single SIP phone
Answer: D
Explanation:
To enter voice register pool configuration mode and create a pool configuration for a SIP IP phone in Cisco Unified CME or for a set of SIP phones in Cisco Unified SIP SRST, use the voice register pool command in global configuration mode.
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/command/reference/cme_cr/cme_ v1ht.html#wp2339729225
NEW QUESTION # 109
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. Increase Retry INVITE to 20 seconds on the SIP profile.
- B. Change Session Refresh Method on the SIP profile to INVITE.
- C. Enable SIP ReMXX Options on the SIP profile.
- D. Enable Send send-receive SDP in mid-call INVITE on the SIP profile.
Answer: B
NEW QUESTION # 110
Which two types of authentication are supported for the configuration of Intercluster Lookup Service? (Choose two.)
- A. TokenID
- B. TLS certificates
- C. username and secret key
- D. FQDN of the servers defined in DNS
- E. passwords
Answer: B,E
Explanation:
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/11_5_1/sysConfig/11_5_1_S U1/cucm_b_system-configuration-guide-1151su1/cucm_b_system-configuration-guide-
1151su1_chapter_011001.pdf
NEW QUESTION # 111
An engineer must implement call restriction to toll-free numbers using a class of restriction in a branch Cisco UCME. In which two places is the corlist incoming or cor Incoming command configured? (Choose two.)
- A. "dial-peer cor custom" configuration mode
- B. "voice register global" configuration mode
- C. "telephony-service" configuration mode
- D. "voice register pool" configuration mode
- E. "ephone-dn' configuration mode
Answer: D,E
NEW QUESTION # 112 
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. Phone_B sends a SIP-REFER message to the Cisco Unified CM with Phone_C information in the Refer-To section.
- 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. 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.
- 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: B,C
Explanation:
Section: Signaling and Media Protocols
NEW QUESTION # 113
Which action is correct with respect to toll fraud prevention configuration in the Cisco Unified Communications Manager Express?
- A. Configure IP Address Trusted Authentication for Incoming VoIP Calls.
- B. Configure the command no ip address trusted authenticate under "voice service voip".
- C. Enable Secondary Dial tone on Analog and Digital FXO Ports.
- D. Configure Direct Inward Dial for Incoming ISDN Calls with overlap dialing.
Answer: A
Explanation:
Reference:
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/admin/configuration/manual/ cmeadm/cmetoll.html#concept_ECC4F4E7ED0F45C594B703EEF34762F2
NEW QUESTION # 114
Which description of RTP timestamps or sequence numbers is true?
- A. Timestamps increase by the time "carrying" by a packet.
- B. The timestamp is used to place the incoming audio and video packets in the correct timing order (playout
- C. The sequence number is used to detect losses.
- D. Sequence numbers increase by four for each RTP packet transmitted.
Answer: B
Explanation:
delay compensation).
NEW QUESTION # 115
......
100% Guarantee Download 300-815 Exam Dumps PDF Q&A: https://www.passleader.top/Cisco/300-815-exam-braindumps.html
Kickstart your Career with Real Updated Questions: https://drive.google.com/open?id=18W3bNHi_yVAy04ADs8pA43vbArjWvyFC