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The Cisco 300-815 certification exam is designed to test the skill level of network professionals in implementing advanced call control and mobility services in Cisco environments. This exam is ideal for individuals who are seeking a career in the field of networking or for those who are already working in the industry and want to advance their skills and knowledge. The certification is also a great way for network professionals to demonstrate their expertise to their employers and peers.

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Cisco Implementing Cisco Advanced Call Control and Mobility Services Sample Questions (Q89-Q94):

NEW QUESTION # 89

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

Explanation:
Section: Call Control and Dial Planning
Explanation/Reference:


NEW QUESTION # 90
Which two types of authentication are supported for the configuration of Intercluster Lookup Service? (Choose two.)

  • A. passwords
  • B. TLS certificates
  • C. TokenID
  • D. FQDN of the servers defined in DNS
  • E. username and secret key

Answer: A,B

Explanation:
Section: Cisco Unified CM Call Control Features
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 # 91
Where is the dtmf-relay command configured on Cisco Unified Border Element?

  • A. in the VoIP dial peer
  • B. in global SIP configuration
  • C. in the VoIP or POTS dial peers
  • D. in the voice-class VoIP configuration

Answer: A


NEW QUESTION # 92
For s SIP to SIP call flow, when does Cisco Unified Border Element require transcoding resources for DTMF?

  • A. interworking between an OOB method and RFC2833 for flow-around calls
  • B. interworking between h245-signal and rtp-nte
  • C. interworking between an OOB method and RFC2833 for flow-through calls
  • D. interworking between h245-alpha numeric and sip-kpml

Answer: A

Explanation:
Reference:
https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-border- element/200412-DTMF-Relay-and-Interworking-on-CUBE.html#anc35


NEW QUESTION # 93
What is first preference condition matched in a SIP-enabled incoming dial peer?

  • A. answer-address
  • B. incoming called-number
  • C. target carrier-id
  • D. incoming uri

Answer: D

Explanation:
Section: Signaling and Media Protocols
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 # 94
......

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