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Cisco 300-815 exam covers a wide range of topics related to advanced call control and mobility services. Candidates will be tested on their ability to configure and troubleshoot CUCM features such as call routing, multi-site deployment, media resources, and SIP trunks. They will also be evaluated on their knowledge of Cisco Unity Connection features such as voicemail, unified messaging, and AutoAttendant. Additionally, the exam covers Cisco Unified IM&P features such as presence, chat, and group chat.
Earning the Cisco 300-815 certification demonstrates a professional’s expertise in implementing advanced call control and mobility services using Cisco technologies. Implementing Cisco Advanced Call Control and Mobility Services certification is highly valued in the IT industry and can lead to career advancement opportunities and higher salaries. Additionally, the certification is valid for three years, after which candidates must recertify by passing a relevant exam or earning continuing education credits.
Cisco 300-815 exam is designed for professionals who want to validate their knowledge and expertise in implementing Cisco Advanced Call Control and Mobility Services. Implementing Cisco Advanced Call Control and Mobility Services certification exam is part of the Cisco Certified Network Professional (CCNP) Collaboration program and is intended for individuals who are responsible for designing, configuring, and deploying advanced collaboration solutions in enterprise environments.
NEW QUESTION # 145
Which configuration element of a hunt group allows for changing Calling Party Transformations settings?
- A. line group
- B. route group
- C. hunt list
- D. hunt pilot
Answer: D
Explanation:
Reference:
https://community.cisco.com/t5/ip-telephony-and-phones/call-alerting-on-hunt-group-as-shared- line/td-p/2658015
NEW QUESTION # 146
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 Diversion modify "<sip:1(...)@" "<sip:9135551\1@" under the SIP profile configuration
- B. request Invite sip-header modify "<sip:1(...)@" "<sip:9135551\1@" under the SIP translation profile configuration
- C. sip-header modify "<sip:1(...)@" "<sip:9135551\1@" under the voice translation profile configuration
- D. request Invite sip-header modify "<sip:1(...)@" "<sip:9135551\1@" under the voice translation profile configuration
Answer: A
NEW QUESTION # 147
Which two types of distribution algorithm are within a line group? (Choose two.)
- A. random
- B. highest preference
- C. bottom up
- D. top down
- E. circular
Answer: D,E
Explanation:
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/9_0_1/ccmcfg/CUCM_BK_C DF59AFB_00_admin-guide-90/ CUCM_BK_CDF59AFB_00_admin-guide_chapter_0100011.html
NEW QUESTION # 148
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. Cisco RIS data collector service must be running on the same server as the Cisco CallManager service.
- C. A unicast music on hold audio source must be configured.
- D. Cisco IP Voice Media Streaming Service must be activated on at least one node in the cluster.
- E. The maximum number of callers allowed in queue must be 10.
Answer: B,D
Explanation:
Section: Cisco Unified CM Call Control Features
Explanation/Reference: https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/12_0_1/systemConfig/ cucm_b_system-configuration-guide-1201/cucm_b_system-configuration-guide-
1201_chapter_01001101.html#CUCM_RF_C960BC9A_00
NEW QUESTION # 149
A company has an SRST gateway running an IOS XE image. The company plans to enable the IPv6 addressing companywide. To enable the IPv6 in a unified SRST gateway to support SIP phones, what are two supported supplementary features for an IPv6 fallback scenario? (Choose two.)
- A. three-way conference
- B. transcoding
- C. secure SIP lines
- D. T.38 fax relay
- E. SIP trunk
Answer: A,D
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_sip_isr4000.html
NEW QUESTION # 150
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. answer-address 555 ........
- B. incoming called number 555.......
- C. session-protocol sipv2
- D. codec g729
Answer: B
NEW QUESTION # 151
Which description of RTP timestamps or sequence numbers is true?
- A. Sequence numbers increase by four for each RTP packet transmitted.
- 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. Timestamps increase by the time "carrying" by a packet.
Answer: C
NEW QUESTION # 152
When you troubleshoot H.323 call setup, which message informs you that the called party is being notified about the call?
- A. PROCEEDING
- B. ALERTING
- C. RINGING
- D. CONNECT
Answer: B
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 # 153
Refer to the exhibit.
The customer is troubleshooting an issue where users cannot dial the CMS SIP Route Pattern. The CMS URI they are attempting to dial is [email protected]. Which IPv4 pattern must the customer enter to resolve the issue?
- A. cms.domain.local
- B. '@cms.domain.local
- C. [email protected]
- D. ".cms.domain.local
Answer: A
NEW QUESTION # 154
Refer to the exhibit.
DN 1003 was the last to ring during the most recent call. Which hunting method ensures that DN 1005 is presented with the next call when the hunt pilot is dialed?
- A. call-blast
- B. parallel
- C. peer
- D. sequential
Answer: C
NEW QUESTION # 155
Which two types of distribution algorithm are within a line group? (Choose two.)
- A. random
- B. highest preference
- C. bottom up
- D. top down
- E. circular
Answer: D,E
NEW QUESTION # 156
Some users report having issues dialing some external numbers when traveling to other locations within the company. The company has five locations in five cities in one country and has an egress gateway in each location for TEHO. The configuration has no specific entry stating that the roaming users are using the local gateway, but calls are going out. How is a verification of the call routing in such a specific configuration performed to further identify the problem?
- A. TEHO
- B. local route groups
- C. standard local route group
- D. device mobility
Answer: D
NEW QUESTION # 157
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-cigit directory number. These 4-digit directory numbers range from 1000 to 9899. 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. voice class sip-profiles 105
request INVITE sip-header Diversion modify "andlt;sip:1(...)@""andlt;sip:263411\1@" - B. voice class sip-profiles 105
request INVITE sip-header Diversion modify "andlt;sip:(...)@""andlt;sip:263411\1@" - C. voice class sip-profiles 105
request INVITE sip-header Diversion add "andlt;sip:(...)@""andlt;sip:263411\1@" - D. voice class sip-profiles 105
request INVITE sip-header Diversion modify "andlt;sip:1(...)@""andlt;sip:263411\2@"
Answer: B
NEW QUESTION # 158
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-versa. As an administrator, which additional configuration should be made to fulfill the user's request?
- A. Check to make sure that the Resume softkey option appears on the desk phone.
- B. Confirm that the desk phone is subscribed to Cisco Extension Mobility.
- C. Add the mobility key to the softkey template that the desk phone is using.
- D. Use Dialed Number Analyzer to determine if the user extension can dial the mobile phone.
Answer: C
Explanation:
Step 11 Configure a Mobility softkey for the phone user that uses Cisco Unified Mobility.
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_0110111.html
NEW QUESTION # 159
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. answer-address 555 "¦"¦..
- B. session-protocol sipv2
- C. incoming called number 555"¦"¦.
- D. codec g729
Answer: B
NEW QUESTION # 160
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 pots
- D. dial-peer voice 20 voip
Answer: D
Explanation:
https://www.ciscopress.com/articles/article.asp?p=664148&seqNum=6
NEW QUESTION # 161
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. use of port 5061 for SIP secure
- B. access list allowing the firewall IP
- C. ip prefix-list to filter the unwanted IP address
- D. bind signaling and media to the loopback interface
Answer: D
NEW QUESTION # 162
......
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