Review Questions
Use these questions to review what you've learned in this chapter. The answers appear in the Answers Appendix.
-
Which of the following CUCM and IOS gateway features provides failover for MGCP controlled gateways?
- MGCP SRST
- SRST fallback
- MGCP fallback
- MGCP in SRST mode
- MGCP failover
-
Which two show the correct numbers of supported phones in Cisco Unified SRST 8.0 for the given platform?
- 800: 24
- 2801–2851: 25-100
- 2901–2951: 400-800
- 3825, 3845: 350,730
-
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?
- This is not possible. Users have to dial headquarters users by their E.164 PSTN number while in fallback mode.
- Translation profiles modifying the calling number.
- The dialplan-pattern command.
- Translation profiles modifying the called number.
- Although this is possible, it should be avoided, because it may confuse users.
-
Which two signaling protocols can be used on a remote gateway with MGCP fallback when in fallback mode?
- MGCP
- SCCP
- SIP
- H.323
- Megaco
- RTP
-
What IOS commands are used on a remote gateway configured with MGCP fallback to give PSTN access for IP Phones during fallback mode?
- This feature is unavailable. Remote phones can only dial each other.
- H.323 or SIP dial peers.
- Static routes.
- Dynamic dial peers.
- MGCP dial peers.
-
What protocol is used between the remote-office Cisco IP Phones and CUCM during fallback to send keepalives to CUCM for SRST?
- MGCP
- SCCP
- H.323
- OSPF hello packets
- EIGRP keepalives
-
What method maintains SIP connectivity from remote-office Cisco IP Phones to CUCM during a complete WAN failure?
- This is not possible. Remote-office Cisco IP Phones will fail over to the SRST remote-office router.
- SIP can be routed through the PSTN with POTS dial peers.
- SIP can be routed through the PSTN if the local exchange carrier enables SIP through the PSTN connection.
- SIP can be routed through the PSTN with POTS dial peers combined with the local exchange carrier, enabling SIP through the PSTN connection.
-
What are two requirements to have multiple MOH sources in SRST?
- This is not possible.
- SRST v8.x must be implemented.
- Configure MOH groups.
- Only use SIP phones.