Newest 300-815 Vce File & Leader in Qualification Exams & Free Download Cisco Implementing Cisco Advanced Call Control and Mobility Services

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Cisco 300-815 exam consists of 60-70 multiple-choice and multiple-answer questions that are designed to test your knowledge and skills in implementing and troubleshooting advanced call control and mobility services. 300-815 exam duration is 90 minutes, and candidates must achieve a score of at least 825 out of 1000 points to pass. 300-815 exam can be taken in English and Japanese languages and is available globally through Pearson VUE testing centers.

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The Implementing Cisco Advanced Call Control and Mobility Services certification exam is one of the top-rated career advancement 300-815 certifications in the market. This Implementing Cisco Advanced Call Control and Mobility Services certification exam has been inspiring candidates since its beginning. Over this long period, thousands of Implementing Cisco Advanced Call Control and Mobility Services exam candidates have passed their 300-815 Certification Exam and now they are doing jobs in the world's top brands.

Cisco Implementing Cisco Advanced Call Control and Mobility Services Sample Questions (Q143-Q148):

NEW QUESTION # 143
An administrator has configured two route patterns, 9.911 and 9.[2-9]XXXXXX. When a user dials
9911. Cisco UCM waits for the T302 timer before routing the call. How will the administrator force interdigit timeout and route the call as soon as the user has finished dialing 9911, without waiting for the T302 timer to expire?

  • A. enable Urgent Priority on the 9.911 pattern
  • B. enable Urgent Priority on the 9.[2-9]XXXXXX pattern
  • C. enable Device Override on both route patterns
  • D. decrease the T302 timer in Service Parameters from the default value

Answer: A


NEW QUESTION # 144
Refer to the exhibit. When setting up a new connection to Cisco UCM, the engineer must use out- of-band DTMF. Which configuration meets this requirement?

  • A. dtmf-relay sip-kpml
  • B. dtmf-relay h245-alphanumeric
  • C. dtmf-relay cisco-rtp
  • D. dtmf-relay rtp-nte

Answer: A


NEW QUESTION # 145
End users at a new site report being unable to hear the remote party when calling or being called by users at headquarters. Calls to and from the PSTN work as expected. To investigate the SIP signaling to troubleshoot the problem, which field can provide a hint for troubleshooting?

  • A. c= line of SDP content
  • B. Contact: header of the 200 OK response
  • C. Allow: header if the 200 OK response
  • D. o= line of SDP content

Answer: D

Explanation:
Section: Signaling and Media Protocols


NEW QUESTION # 146
Refer to the exhibit.

A call mode through the Cisco Unified Border Element to pilot 2000 is foiling. What is causing the call to foil?

  • A. No codecs are configured on the dial peers
  • B. The Cisco Unified Border Element is not receiving a response to its OPTION keepahves.
  • C. The destination pattern is incorrect for the dialed number.
  • D. VAD was not disabled on the outgoing dial poor.

Answer: C


NEW QUESTION # 147
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:(...)@""andlt;sip:2634111@"
  • B. voice class sip-profiles 105
    request INVITE sip-header Diversion modify "andlt;sip:1(...)@""andlt;sip:2634111@"
  • C. voice class sip-profiles 105
    request INVITE sip-header Diversion modify "andlt;sip:1(...)@""andlt;sip:2634112@"
  • D. voice class sip-profiles 105
    request INVITE sip-header Diversion add "andlt;sip:(...)@""andlt;sip:2634111@"

Answer: A


NEW QUESTION # 148
......

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