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Cisco 300-815 certification exam is a challenging exam that requires candidates to have a deep understanding of advanced call control and mobility services within a Cisco collaboration solution. Candidates who successfully pass 300-815 exam will have demonstrated their ability to configure and troubleshoot complex call control and mobility services, which is highly valued in the IT industry. Implementing Cisco Advanced Call Control and Mobility Services certification is an excellent way for IT professionals to validate their skills and knowledge in this area and to enhance their career prospects.

Cisco 300-815 exam covers a wide range of topics, including call routing, dial plans, media resources, call control protocols, mobility services, and Cisco Unified Border Element (CUBE). Candidates are expected to have a deep understanding of Cisco Unified Communications solutions, as well as hands-on experience in implementing and troubleshooting these technologies. Successful completion of 300-815 Exam demonstrates the candidate's ability to design and deploy advanced collaboration solutions that meet the needs of modern enterprises.

Cisco 300-815 certification is ideal for IT professionals who want to demonstrate their skills in implementing and troubleshooting advanced call control and mobility services on Cisco platforms. Implementing Cisco Advanced Call Control and Mobility Services certification is recommended for collaboration and communication engineers, network administrators, and network engineers. Achieving this certification validates your skills in implementing and troubleshooting complex Cisco collaboration solutions, which can help you advance your career in the field of communication and collaboration.

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Cisco Implementing Cisco Advanced Call Control and Mobility Services Sample Questions (Q74-Q79):

NEW QUESTION # 74
What are two functions of the metadata in SIP recording? (Choose two.)

• A. carries the communication session data (audio and video calls) that describes the call to the recording server
• B. identifies the session and media association time
• C. includes the details of the application media stream
• D. contains the cryptographic context of the SRTP passthrough calls
• E. carries the Agent Statistics Report

Answer: D,E

NEW QUESTION # 75
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. Ask the firewall administrator to change the ports to TCP.
• B. Ask the firewall administrator to change the range of UDP ports to 16384-32767.
• C. Go to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 20000-22000.
• 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. Go to System Parameters in Cisco Unified Communications Manager and change the range of media ports to 20000-22000.

Answer: B,C

Explanation:
In this case, one solution is to change the ports on the SIP profile of the phones to match the ports on the firewall. The other solution is to ask the firewall admin to change his ports to match the ports in UCM.
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 # 76
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 /

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