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This chapter is from the book

Review Questions

Use these questions to review what you've learned in this chapter. The answers appear in the Answers Appendix.

  1. Which of the following CUCM and IOS gateway features provides failover for MGCP controlled gateways?

    1. MGCP SRST
    2. SRST fallback
    3. MGCP fallback
    4. MGCP in SRST mode
    5. MGCP failover
  2. Which two show the correct numbers of supported phones in Cisco Unified SRST 8.0 for the given platform?

    1. 800: 24
    2. 2801–2851: 25-100
    3. 2901–2951: 400-800
    4. 3825, 3845: 350,730
  3. What can you use to configure the dial plan at the remote-site gateway so that branch users can still reach the headquarters when dialing internal directory numbers during fallback?

    1. This is not possible. Users have to dial headquarters users by their E.164 PSTN number while in fallback mode.
    2. Translation profiles modifying the calling number.
    3. The dialplan-pattern command.
    4. Translation profiles modifying the called number.
    5. Although this is possible, it should be avoided, because it may confuse users.
  4. Which two signaling protocols can be used on a remote gateway with MGCP fallback when in fallback mode?

    1. MGCP
    2. SCCP
    3. SIP
    4. H.323
    5. Megaco
    6. RTP
  5. What IOS commands are used on a remote gateway configured with MGCP fallback to give PSTN access for IP Phones during fallback mode?

    1. This feature is unavailable. Remote phones can only dial each other.
    2. H.323 or SIP dial peers.
    3. Static routes.
    4. Dynamic dial peers.
    5. MGCP dial peers.
  6. What protocol is used between the remote-office Cisco IP Phones and CUCM during fallback to send keepalives to CUCM for SRST?

    1. MGCP
    2. SCCP
    3. H.323
    4. OSPF hello packets
    5. EIGRP keepalives
  7. What method maintains SIP connectivity from remote-office Cisco IP Phones to CUCM during a complete WAN failure?

    1. This is not possible. Remote-office Cisco IP Phones will fail over to the SRST remote-office router.
    2. SIP can be routed through the PSTN with POTS dial peers.
    3. SIP can be routed through the PSTN if the local exchange carrier enables SIP through the PSTN connection.
    4. SIP can be routed through the PSTN with POTS dial peers combined with the local exchange carrier, enabling SIP through the PSTN connection.
  8. What are two requirements to have multiple MOH sources in SRST?

    1. This is not possible.
    2. SRST v8.x must be implemented.
    3. Configure MOH groups.
    4. Only use SIP phones.
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