Cisco 400-051 Online Practice Questions and Exam Preparation

Cisco 400-051 Online Questions & Answers

• Question 521:

  What does a comma accomplish when it is used in a SIP Dial Rule pattern that is associated with a Cisco 9971 IP Phone that is registered to Cisco Unified Communications Manager?

  A. It inserts a 500-millisecond pause between digits.
  B. It causes the phone to generate a secondary dial tone.
  C. It is a delimiter and has no significant dialing impact.
  D. It indicates a timeout value of 5000 milliseconds.
  E. It is an obsolete parameter and will be ignored.
  B. It causes the phone to generate a secondary dial tone.

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  Comma is accepted in speed dial as delimiter and pause. -Comma used to delineate dial string, FAC, CMC, and post connect digits For post connect digits, commas insert a 2 second delay Commas may be duplicated to create longer delays.

• Question 522:

  What is the recommend non-adjacent channels' cell boundary overlap range for voice and video implementation over wireless LAN?

  A. 25% to 35%
  B. 30% to 40%
  C. 20% to 30%
  D. 15% to 25%
  E. 10% to 20%
  E. 10% to 20%

• Question 523:

  Which protocol does the Cisco Jabber client use, in conjunction with Cisco IM and Presence, to deliver enterprise-class instant messaging services?

  A. SIP
  B. CTI/QBE
  C. XMPP
  D. IRC
  E. ICQ
  C. XMPP

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  Many federated IM networks communicate using an open standard, such as Jabber, that leverages the Extensible Messaging and Presence Protocol (XMPP). Networks using XMPP provide open communications with other XMPP-based networks.

• Question 524:

  Which design restriction applies to Cisco Unified Communications Manager Session Management Edition clustering over the WAN deployment with extended round-trip times in Cisco Unified CM 9.1 and later releases?

  A. SIP and H.323 intercluster trunks are supported.
  B. Only SIP trunk is supported.
  C. SIP trunks and H.323 gateways are supported.
  D. A minimum of 1.544 Mb/s bandwidth is required for all traffic between any two nodes in the cluster.
  E. Only RSVP agents can be configured and registered to the SME cluster as media resources.
  B. Only SIP trunk is supported.

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  Using only SIP trunks in the SME cluster allows you to deploy a "media transparent" cluster where media resources, when required, are inserted by the end or leaf Unified Communications system and never by SME. Using only SIP trunks also allows you to use extended round trip times (RTTs) between SME nodes when clustering over the WAN.

• Question 525:

  Which two analog telephony signaling methods are most vulnerable to glare conditions? (Choose two.)

  A. FXS Loop-start
  B. FXO Ground-start
  C. EandM Wink-start
  D. EandM Delay-dial
  E. EandM Immediate-start
  F. EandM Feature Group D
  A. FXS Loop-start
  E. EandM Immediate-start

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  The loop start signaling method is more common and is typically used by residential phone lines. When a voice port is configured with loop start signaling, the device (telephone) closes the circuit loop that signals the CO voice port to provide dial tone; an incoming call is signaled on the CO by supplying a predefined voltage on the line. The loop start signaling method has one main disadvantage in that it has no method of preventing both sides of the connection from attempting to seize the line at the same time; this condition is referred to as glare. Because of this, loop start signaling is typically not used on high demand circuits.

  With immediate-start, the calling side of the connection seizes the line by going off hook on the E-lead and address information is sent using dual-tone multifrequency (DTMF) digits. Immediate start signaling is vulnerable to glare just like loop-start signaling.

• Question 526:

  Which Cisco IOS multipoint video conferencing profile attempts to reserve DSPs only when it is activated with an actual conference?

  A. homogeneous
  B. guaranteed-audio
  C. rendezvous
  D. heterogeneous
  E. flex mode video
  B. guaranteed-audio

• Question 527:

  Which statement describes how the digit zero is handled in the predefined restriction tables in Cisco Unity Connection?

  A. Zero is listed in the Default Out-Dial Restriction table.
  B. Zero is listed in the Default System Transfer Restriction table.
  C. Zero is listed in the Default Transfer Restriction table.
  D. Zero is listed in the User-Defined and Automatically Added Alternate Extensions Restriction table.
  E. Zero is not listed in any default restriction table configuration.
  E. Zero is not listed in any default restriction table configuration.

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  When user dials "0", by default Unity Connection treats it as an operator call and does not block "0" by any restriction table configuration. Only the operator can modify transfer extension associated with operator call.

• Question 528:

  Which two parameters are requested in an Audit Connection message from a Cisco Unified CM to endpoint on a MGCP gateway? (Choose two)

  A. Call ID
  B. Capabilities
  C. Bearer Information
  D. Connection Parameters
  E. Connection Mode
  F. Connection ID
  C. Bearer Information
  F. Connection ID

• Question 529:

  In a Cisco EnergyWise domain, which two terms describe a Cisco IP phone? (Choose two.)

  A. endpoint
  B. domain member
  C. child domain member
  D. EnergyWise agent
  E. Cisco power distribution unit
  A. endpoint
  C. child domain member

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  References: http://www.cisco.com/en/US/docs/switches/lan/energywise/phase2_5/ios/configuration/guide/one_ent.html

• Question 530:

  Refer to the exhibit.

  

  A user is going through a series of dialing steps on a SIP Type A IP phone to call a SCCP IP phone. Both phones are registered to the same Cisco Unified Communications Manager cluster. Assume that the calling SIP phone is not associated with any SIP dial rules.

  Which statement about how digits are forwarded to the Cisco Unified Communications Manager for further call processing is true?

  A. As each digit is pressed on the SIP IP phone, it is sent to the Cisco Unified Communications Manager in a SIP NOTIFY message as a KPML event.
  B. The SIP IP phone waits for the inter-digit timer expiry and then sends each digit to the Cisco Unified Communications Manager as a separate KPML event in a SIP NOTIFY message.
  C. The SIP IP phone waits for the inter-digit timer expiry or for the Dial soft key to be selected before it sends all digits to the Cisco Unified Communications Manager in a SIP INVITE message.
  D. The SIP IP phone waits for the inter-digit timer expiry, or for the Dial soft key to be selected before it sends the first digit in a SIP INVITE and the subsequent digits in SIP NOTIFY messages.
  E. The SIP IP phone sends all digits to the Cisco Unified Communications Manager in a SIP INVITE message as soon as the fourth digit is pressed.
  C. The SIP IP phone waits for the inter-digit timer expiry or for the Dial soft key to be selected before it sends all digits to the Cisco Unified Communications Manager in a SIP INVITE message.

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  Because Type A SIP phone with no SIP dial rules sends digit in Enbloc style.All digits are sent to CUCM after the user completes the dialing and press the Dial softkey.

  References: https://supportforums.cisco.com/document/87236/working-concept-sccp-sip-phones-and-dial-rules.