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Q76. Which two statements about remote survivability are true? (Choose two.)
A. SRST supports more Cisco IP Phones than Cisco Unified Communications Manager Express in SRST mode.
B. Cisco Unified Communications Manager Express in SRST mode supports more Cisco IP Phones than SRST.
C. MGCP fallback is required for ISDN call preservation.
D. MGCP fallback functions with SRST.
Answer: A,D
Q77. Which two actions ensure that the call load from Cisco TelePresence Video Communication Server to a Cisco Unified Communications Manager cluster is shared across Unified CM nodes? (Choose two.)
A. Create a neighbor zone in VCS with the Unified CM nodes listed as location peer addresses.
B. Create a single traversal client zone in VCS with the Unified CM nodes listed as location peer addresses.
C. Create one neighbor zone in VCS for each Unified CM node.
D. Create a VCS DNS zone and configure one DNS SRV record per Unified CM node.
E. In VCS set Unified Communications mode to Mobile and remote access and configure each Unified CM node.
Answer: A,D
Q78. Company X has a Cisco Unified Communications Manager cluster and a VCS Control server with video endpoints registered on both systems. Users find that video endpoints registered on Call manager can call each other and likewise for the endpoints registered on the VCS server. The administrator for Company X realizes he needs a SIP trunk between the two systems for any video endpoint to call any other video endpoint. Which two steps must the administrator take to add the SIP trunk? (Choose two.)
A. Set up a SIP trunk on Cisco UCM with the option Device-Trunk with destination address of the VCS server.
B. Set up a subzone on Cisco UCM with the peer address to the VCS cluster.
C. Set up a neighbor zone on the VCS server with the location of Cisco UCM using the menu option VCS Configuration > Zones > zone.
D. Set up a SIP trunk on the VCS server with the destination address of the Cisco UCM and Transport set to TCP.
E. Set up a traversal subzone on the VCS server to allow endpoints that are registered on Cisco UCM to communicate.
Answer: A,C
Q79. 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.
Answer: B
Q80. Which three items must you configure to enable SAF Call Control Discovery? (Choose three.)
A. the SIP or H.323 trunk
B. hosted DN groups
C. hosted DN patterns
D. route patterns
E. a calling search space
F. translation patterns
Answer: A,B,C
Q81. When you use the Query wizard to configure the trace and log central feature to collect install logs, if you have servers in a cluster in a different time zone, which time is used?
A. TLC adjusts the time change appropriately.
B. TLC uses its local time for all systems.
C. TLC queries for the time zone as part of configuration.
D. TLC produces an error and must be run remotely.
Answer: A
Q82. What component acts as a user agent for both ends of a SIP call while Cisco Unified SIP SRST is providing failover during a WAN outage?
A. B2BUA
B. SIP server
C. SIP proxy
D. SIP SRST router
E. SIP registrar
Answer: A
Q83. Refer to the exhibit.
You have configured transcoder resources in both an IOS router and a Cisco Unified Communications Manager. When you review the configurations in both devices the IP addresses and transcoder names are correct, but the transcoder is failing to register with the Cisco Unified Communications Manager. Which command needs to be edited to allow the transcoder to register properly?
A. The associate profile and dsp farm profile numbers need to match associate ccm 2 command.
B. The associate ccm 2 priority 1 command needs to be changed so the ccm value matches identifier 1 in the sccp ccm 10.1.1.1 command.
C. The sccp ccm group number needs to match the associate ccm 2 command.
D. The maximum sessions command must match the number of codecs configured under the dsp farm profile.
E. The sccp ccm group number must match the voice-card number.
Answer: B
Explanation:
Incorrect Answer: A, C, D, E The value of the IP address should match the IP address in the ip source-address command Link:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmevtr ns.html
Q84. When an H.323 trunk is added for Call Control Discovery, which statement is true?
A. The H.323 trunk is added by selecting Inter-Cluster Trunk (Non-Gatekeeper Controlled) and Device Protocol Inter-Cluster Trunk. The Enable SAF check box should be selected in the trunk configuration.
B. The H.323 trunk is added by selecting Inter-Cluster Trunk (Non-Gatekeeper Controlled) and Device Protocol Inter-Cluster Trunk. The Trunk Service Type should be Call Control Discovery.
C. The H.323 trunk is added by selecting Call Control Discovery Trunk and then selecting
H.323 as the protocol to be used.
D. The H.323 trunk is added by selecting H.323 Trunk, and selecting Inter-Cluster Trunk as the Device Protocol. The destination IP address field is configured as ‘SAF’ to indicate that this trunk is used for SAF.
Answer: A
Explanation:
Reference. Implementing Cisco Unified Communications Manager Part 2 (CIPT2), Chapter3: Implementing Multisite Connections, pg 70-71, Fig 3-14 and Fig 3-15
Q85. When an external call is placed from Ajax, they would like the ANI that is sent to the PSTN to be the main number, not the extension. For domestic calls, they would like 10 digits sent; for international calls, they would like to send the country code 1 and the 10 digits. How can this be accomplished?
A. Add a translation pattern to the dial peers in the gateway that adds the appropriate digits to the outgoing ANI.
B. In the external call route patterns, set the external phone number mask to the main number. Use 10 digits in the domestic route pattern and 1 followed by the main number digits in the international route patterns.
C. Use a calling party transform mask for each route group in the corresponding route list configuration. Set the explicit 10-digit main number for domestic calls and 1 followed by the main number for the international route patterns.
D. In the directory number configurations, set the prefix digits field to the country code and the 10 digits of the main number. This will be truncated to the 10-digit number for domestic calls and sent out in its entirety for international calls.
Answer: C
Explanation:
Incorrect Answer: A, B, D calling party transformation mask value is Valid entries for the NANP include the digits 0 through 9; the wildcard characters X, asterisk (*), and octothorpe (#); and the international escape character +. Link:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucmbe/admin/8_6_1/ccmcfg/b03trpat.ht ml
Q86. Which method can be used to address variable-length dial plans?
A. Overlap sending and receiving.
B. Add a prefix for all calls that are longer than 10-digits long
C. Use nested translation patterns to eliminate inter-digit timeout
D. Use the @macro on the route pattern
E. Use MGCP gateways, which support variable-length dial plans
Answer: A
Explanation:
Incorrect Answer: B, C, D, E If the dial plan contains overlapping patterns, Cisco Unified Communications Manager does not route the call until the interdigit timer expires (even if it is possible to dial a sequence of digits to choose a current match). Check this check box to interrupt interdigit timing when Cisco Unified Communications Manager must route a call immediately. By default, the Urgent Priority check box displays as checked. Unless your dial plan contains overlapping patterns or variable length patterns that contain!, Cisco recommends that you do not uncheck the check box. Link:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmfeat/fsintrcm.htm
Q87. Which three steps configure Cisco Unified Survivable Remote Site Telephony for SIP phones? (Choose three.)
A. Configure a phone NTP reference.
B. Configure an SRST reference.
C. Configure the SIP registrar.
D. Configure voice register global dn.
E. Configure voice register pool.
F. Configure telephony service.
Answer: B,C,E
Q88. Which three options are supplementary services that are affected by MTP? (Choose three.)
A. Call Hold
B. Call Transfer
C. Call Park
D. Call Pickup
E. Speed Dial
F. Call Back
Answer: A,B,C
Q89. What user profile is used to define the settings for a user on login?
A. Device Profile
B. Group Profile
C. Pool Profile
D. Specific Profile
Answer: A
Q90. Which two features require or may require configuring a SIP trunk? (Choose two.)
A. SIP gateway
B. Call Control Discovery between a Cisco Unified Communications Manager and Cisco Unified Communications Manager Express
C. Cisco Device Mobility
D. Cisco Unified Mobility
E. registering a SIP phone
Answer: A,B
Explanation:
Incorrect Answer: C, D, E All protocols require that either a signaling interface (trunk) or a gateway be created to accept and originate calls. Device mobility allows Cisco Unified Communications Manager to determine whether the phone is at its home location or at a roaming location. Cisco Unified Mobility gives users the ability to redirect incoming IP calls from Cisco Unified Communications Manager to different designated phones, such as cellular phones. Link:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmsys/a08sip.html# wpxref77849