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Q101. What is the difference between an MGCP gateway and a SIP gateway?
A. An MGCP gateway that dial peers be configured before PSTN calls can be placed and received. The SIP gateway requires no dial peers.
B. An MGCP gateway can be added in Cisco Unified Communications Manager under the Gateway Type field using the gateway model. The SIP gateway can connect to Cisco Unified Communications Manager only through a SIP trunk.
C. A SIP gateway requires a call agent for PSTN calls to be placed and received. An MGCP gateway does not require a call agent for PSTN calls to be placed and received.
D. An MGCP gateway can register with Cisco Unified Communications Manager. A SIP gateway will show status of "Unknown".
E. The SIP gateway must be configured in Cisco Unified Communications Manager using a valid IP address on the gateway. The MGCP gateway must be configured in Cisco Unified
Communications Manager using the domain name.
Q102. During device failover to the secondary Cisco Unified Communications Manager server, how does the phone recognize that the primary server is back?
A. The secondary server keeps sending keepalive message to the primary server and when it succeeds, it unregisters the phones to force them to register to the primary.
B. When the primary server goes online, it sends out an "ALIVE" message via broadcast so that the phones re-register.
C. The phones never re-register with the primary server until the active (secondary) one goes down.
D. The phone sends keepalive messages to the primary server frequently and when it succeeds, the phone re-registers with it.
Q103. Which two are gatekeeper-controlled trunk options that support gatekeeper call administration control? (Choose two.)
There are two call control systems in this item. The Cisco UCM is controlling the DX650, the Cisco Jabber for Windows Client, and the 9971 Video IP Phone. The Cisco VCS is controlling the SX20, the Cisco TelePresence MCU, and the Cisco Jabber TelePresence for Windows.
Use the exhibits to answer the following questions.
SX20 System Information
What two issues could be causing the Cisco Jabber Video for TelePresence failure shown in the exhibit? (Choose two.)
A. Incorrect username and password
B. Wrong SIP domain configured.
C. User is not associated with the device.
D. IP or DNS name resolution issue.
E. CSF Device is not registered.
F. IP Phone DN not associated with the user.
Q105. Refer to the exhibit.
The HQ site uses area code 650. The BR1 site uses area code 408. The long distance national code for PSTN dialing is 1. To make a long distance national call, an HQ or BR1 user dials access code 9, followed by 1, and then the 10-digit number.
Both sites use MGCP gateways. AAR must use globalized call routing using a single route pattern. Assume that all outgoing PSTN numbers are localized at the egress gateway as shown in the exhibit.
How many route lists and route groups should be configured for AAR at a minimum?
A. a single route list with a local route group for each site
B. two route lists and two route groups for each site
C. a single route list and four route groups for each site
D. None. The AAR CSS can point directly to the route pattern.
Q106. How does the system intelligently shift call processing upon restoration of WAN connectivity?
A. automatically back to the primary Cisco Unified Communications Manager cluster
B. manually back to the primary Cisco Unified Communications Manager cluster
C. automatically back to the secondary Cisco Unified Communications Manager cluster
D. manually back to the secondary Cisco Unified Communications Manager cluster
Q107. Assume that local route groups are configured. When an IP phone moves from one device mobility group to another, which two configuration components are not changed? (Choose two.)
A. IP subnet
B. user settings
C. SRST reference
E. phone button settings
Incorrect Answer: A, C, D Although the phone may have moved from one subnet to another, the physical location and associated services have not changed. Link:
Q108. Refer to the exhibit.
What must be configured on the HQ Cisco Unified Communications Manager to allow HQ users to dial the SAF learned directory number pattern 3XXX?
A. Route pattern 3XXX should be configured and made available to HQ users through the phone CSS.
B. Route pattern 3XXX should be configured and made available to HQ phone users through the phone AAR CSS.
C. The SAF partition assigned to the SAF learned patterns must be available to the HQ phone users through the phone CSS.
D. The SAF partition assigned to the SAF learned patterns must be available to the HQ phone users through the phone AAR CSS.
E. The SAF directory number pattern 3XXX will be made available to all users automatically as soon as the SAF partition is selected.
Incorrect Answer: A, B, D By adopting the SAF network service, the call control discovery feature allows Cisco Unified Communications Manager to advertise itself along with other key attributes. Link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmfeat/fscallcontrol discovery.html
Q109. Which trunk should you use in an H.323 gatekeeper-controlled network?
E. MGCP FXO trunk
F. MGCP T1/E1 trunk
Q110. The following exhibit shows configs for H.323 gateway. You have been asked to implement TEHO from a remote branch office with area code 301 to the HQ office with area code 201 using Cisco Unified Communications Manager. The remote office has an MGCP gateway and the HQ office has an H.323 gateway. Once the call arrives at the HQ, it should break out to the PSTN as a seven-digit local call. Which statement about the route pattern is true?
A. route pattern should be 91201.[2-9]XXXXXX with Discard Digit as Predot and Prefix 9
B. route pattern should be 91201.[2-9]XXXXXX with Discard Digit as Predot
C. route pattern should be 91201.[2-9]XXXXXX
D. route pattern should be 9.1201[2-9]XXXXXX with Discard Digit as Predot
E. route pattern should be 9.1201[2-9]XXXXXX with Discard Digit as Predot and Prefix 9
Incorrect Answer: B, C, D, E
Destination pattern is 91, HQ office area code is 201 .