Cisco 642-432 Dumps : Cisco Voice Over IP

642-432 real exams

Exam Code: 642-432

Exam Name: Cisco Voice Over IP

Updated: Sep 10, 2026

Q & A: 72 Questions and Answers

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Cisco 642-432 Exam Syllabus Topics:

SectionObjectives
Call Control and Survivability- Redundancy and high availability
  • 1. CallManager redundancy mechanisms
    - SRST (Survivable Remote Site Telephony)
    • 1. SRST configuration and failover behavior
      Voice Gateway and Protocol Integration- VoIP signaling protocols
      • 1. SIP fundamentals
        • 2. H.323 call setup and interoperability
          - PSTN integration
          • 1. Voice gateways configuration concepts
            • 2. Dial peers and call legs
              Quality of Service (QoS) for Voice- Voice traffic prioritization
              • 1. Latency, jitter, and packet loss considerations
                • 2. QoS marking and classification (DSCP)
                  IP Phone Deployment and Features- Advanced telephony features
                  • 1. Call forwarding and hunt groups
                    • 2. Conference and transfer services
                      - Endpoint registration and configuration
                      • 1. Device configuration templates
                        • 2. Cisco IP phone registration process
                          Cisco Unified Communications Architecture- CUCM (Cisco Unified Communications Manager) fundamentals
                          • 1. Device pools, regions, and locations
                            • 2. System components and architecture overview
                              - Call processing design
                              • 1. Call routing logic and dial plan concepts
                                • 2. Digit manipulation and translation patterns

                                  Cisco Voice Over IP Sample Questions:

                                  Question #1

                                  Site A uses three-digit internal numbers and Site B uses four-digit internal numbers.
                                  All calls to the PSTN are routed through Site B.
                                  What dial plan below best represents provision simplicity?

                                  • A. Translate all called numbers at either site to ten digits.
                                  • B. Translate all called numbers leaving Site A to ten digits.
                                  • C. Translate all called numbers within Site A to four digits.
                                  • D. Translate all called numbers within Site B to three digits.
                                  Reveal Solution  Discussion  0

                                  Correct Answer: B  🗳️

                                  Question #2

                                  You have a customer that is interested in determining the number of VoIP calls their Frame Relay WAN links can support. Each of their Frame Relay WAN links has 84 kbps of bandwidth available outside all other applications and overhead. How many G.729 calls using the 8 kbps codec and 20 byte sample size can be supported?

                                  • A. 2
                                  • B. 4
                                  • C. 3
                                  • D. 1
                                  Reveal Solution  Discussion  0

                                  Correct Answer: C  🗳️

                                  Question #3

                                  To hide its identity when initiating calls, Phone B requests that Server B place its calls for it. What kind of device is Server B?

                                  • A. registrar
                                  • B. proxy
                                  • C. user agent client
                                  • D. redirect
                                  • E. user agent server
                                  Reveal Solution  Discussion  0

                                  Correct Answer: B  🗳️

                                  Question #4

                                  What are two basic parameters needed to setup a dial peer connected to the PSTN? (Choose two.)

                                  • A. signaling type
                                  • B. destination pattern
                                  • C. voice port
                                  • D. interface bandwidth
                                  Reveal Solution  Discussion  0

                                  Correct Answer: B,C  🗳️

                                  Question #5

                                  Examine the example output.
                                  hostname GW1 ! interface Ethernet 0/0 ip address 172.16.2.1 255.255.255.0 h323-gateway voip interface h323-gateway voip id GK1-zone1.abc.com abc.com ipaddr 172.16.2.2 h323-gateway voip h323-id GW1 h323-gateway voip bind srcaddr 172.16.2.1 ! dial-peer voice 1 voip destination-pattern 1212. session-target ras ! dial-peer voice 2 pots destination-pattern 2125551212 no register e164 !
                                  end
                                  Choose the command that will restore communication with gatekeeper functionality to this device.

                                  • A. h323-gateway voip h323-id GK1
                                  • B. h323-gateway voip GW1-zone2.abc.com abc.com ipaddr 172.16.2.1
                                  • C. h323-gateway voip bind srcaddr 172.16.2.2
                                  • D. gateway
                                  Reveal Solution  Discussion  0

                                  Correct Answer: D  🗳️

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