300-075 Exam Details

  • Exam Code
    :300-075
  • Exam Name
    :Implementing Cisco IP Telephony and Video
  • Certification
    :Cisco Certifications
  • Vendor
    :Cisco
  • Total Questions
    :522 Q&As
  • Last Updated
    :Dec 10, 2021

Cisco 300-075 Online Questions & Answers

  • Question 1:

    Which three commands are mandatory to implement SRST for five Cisco IP Phones? (Choose three.)

    A. call-manager-fallback
    B. max-ephones
    C. keepalive
    D. limit-dn
    E. ip source-address

  • Question 2:

    Which two results occur in SRST mode if the pickup command is configured under the call-manager-fallback section of a router? (Choose two.)

    A. Calls ringing an unassigned directory number are forwarded to the audio attendant.
    B. The GPickUp soft key is on all SRST all SRST phones.
    C. Calls coming into a Cisco Unified Communications Manager registered phone can be picked up by a SRST phone.
    D. Calls coming into one directory number can be picked up from another directory number.
    E. The PickUp soft key is on all SRST phones.

  • Question 3:

    Which command displays the detailed configuration of all the Cisco Unified IP phones, voice ports, and dial peers of the Cisco Unified SRST router?

    A. show call-manager-fallback all
    B. show dial-peer voice summary
    C. show ephone summary
    D. show voice port summary

  • Question 4:

    What is the maximum allowed round-trip time between Cisco Unified Communications Manager nodes?

    A. 40 ms
    B. 500 ms
    C. 80 ms
    D. 150 ms

  • Question 5:

    A video endpoint is configured with SIP only. What does a video endpoint use to register with the VCS Control?

    A. IP address
    B. SIP URI
    C. MAC address
    D. system name

  • Question 6:

    Which two devices or applications support call preservation? (Choose two.)

    A. JTAPI applications
    B. CTI applications
    C. a software conference bridge
    D. an annunciator
    E. SIP trunks

  • Question 7:

    Which three devices or applications support call preservation? (Choose three.)

    A. a software conference bridge
    B. Cisco Unified IP Phone 7962G
    C. an annunciator
    D. SIP trunks
    E. JTAPI applications
    F. TAPI applications
    G. CTI applications
    H. third-party H.323 endpoints

  • Question 8:

    Which three globalization dialing functions are enhanced in Cisco Unified Communications Manager 7.x and later? (Choose three.)

    A. MGRL
    B. HO
    C. CER
    D. AAR
    E. SAF
    F. click-to-call

  • Question 9:

    Refer to the exhibit.

    A user in RTP calls a phone in San Jose during congestion with Call Forward No Bandwidth (CFNB) configured to reach cell phone 4085550150. The user in RTP sees the message "Not Enough Bandwidth" on their phone and hears a fast busy tone. Which two conditions can correct this issue? (Choose two.)

    A. The calling phone (RTP) needs to have AAR Group value of AAR under the AAR Settings.
    B. The called phone (San Jose) needs to have AAR Group value of AAR under the AAR Settings.
    C. The calling phone (RTP) needs to have the AAR destination mask of 914085550150 configured under the AAR Settings.
    D. The calling phone (RTP) needs to have the AAR destination mask of 4085550150 configured under the AAR Settings.
    E. The called phone (San Jose) needs to have the AAR destination mask of 914085550150 configured under the AAR Settings.
    F. The called phone (San Jose) needs to have the AAR destination mask of 4085550150 configured under the AAR Settings.

  • Question 10:

    Which solution is needed to enable presence and extension mobility to branch office phones during a WAN failure?

    A. SRST without MGCP fallback
    B. SRST with VoIP dial peers to Cisco Unified Communications Manager Express
    C. SRST with MGCP fallback
    D. Cisco Unified Communications Manager Express in SRST mode

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