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Q61. Which three options describe the main functions of SAF Clients? (Choose three.)
A. registering the router as a client with the SAF network B. providing publishing services to the SAF network
C. subscribing to SAF network services
D. registering Cisco Unified Communications Manager subscribers with the publisher
E. starting Cisco Unified Communications Manager services throughout the cluster
F. integrating with Cisco IM and Presence for additional services
Answer: A,B,C
Q62. Which three devices support the SAF Call Control Discovery protocol? (Choose three.)
A. Cisco Unified Border Element
B. Cisco Unity Connection
C. Cisco IOS Gatekeeper
D. Cisco Catalyst Switch
E. Cisco IOS Gateway
F. Cisco Unified Communications Manager
Answer: A,E,F
Q63. The corporate WAN has been extended to two newly acquired sites and it includes gatekeeper support. Each site has a Cisco CallManager and an H.323 gateway that allows connection to the PSTN. Which connection method is best for these two new customers?
A. H.225 trunk (gatekeeper-controlled)
B. intercluster trunk (non-gatekeeper controlled)
C. SIP trunk
D. intercluster trunk (gatekeeper-controlled)
Answer: D
Q64. Which statement is correct about AAR?
A. The end users see, "Network Congestion Rerouting?" but AAR is otherwise transparent to the end user and works without user intervention.
B. AAR will display "not enough bandwidth" on the IP phone while it reroutes the call.
C. AAR allows calls to be rerouted because of insufficient Cisco Unified Border Element controlled bandwidth to an ITSP.
D. AAR allows calls to be rerouted due to insufficient gatekeeper controlled IP WAN bandwidth.
Answer: A
Explanation:
Incorrect Answer: B, C, D Automated alternate routing (AAR) provides a mechanism to reroute calls through the PSTN or other network by using an alternate number when Cisco Unified Communications Manager blocks a call due to insufficient location bandwidth. With automated alternate routing, the caller does not need to hang up and redial the called party. Link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucmbe/admin/8_6_1/ccmcfg/b03aar.ht ml
Q65. Which option describes the reason that transcoding resources are added in Cisco Unified Communications Manager?
A. to enable transcoding resources in a Cisco Unified Communications Manager server
B. to enable Cisco Unified Communications Manager to select the optimal single codec for end-to-end calls
C. to enable transcoding resources in Cisco IP Phones
D. to provide transcoding resources in Cisco IOS gateways to Cisco Unified Communications Manager
Answer: D
Q66. Refer to the exhibit.
Locations-based CAC has been configured between HQ and the BR site. Assume that the priority queue has been provisioned correctly for three G.729 calls. What happens when
the fourth call is placed from HO to BR?
A. The call will get through via the WAN. But it will experience poor audio quality.
B. The call will fail.
C. The call will be queued until one of the existing calls drop.
D. The call will get through without any issues.
Answer: B
Q67. What command is used to map internal extensions to the corresponding E.164 PSTN number when using Cisco Unified Communications Manager Express in SRST mode?
A. ephone-dn
B. dialplan-pattern
C. number
D. number-e.164
E. ephone-transnumber
Answer: B
Q68. Which two statements about SAF service identifier numbers are true? (Choose two.)
A. They are generated in the format service:sub-service:instance.instance.instance.instance.
B. They are 16-bit decimal identifiers.
C. They are generated in the format data-source:sub-service:instance.matrix.fifty.saf.
D. They are 32-bit decimal identifiers.
E. They are generated in the format data.saf.cucm-publisher.asf@domain.local.
F. They are generated in the format telco.cisco.saf-forwader.db.replicate.data.local.
Answer: A,B
Q69. Which action configures transcoding resources in Cisco Unified Communications Manager to function with branch office Cisco IP Phones?
A. Configure the branch office IP phones with CSS and partitions.
B. Configure the branch office IP phones with MRGs and MRGLs.
C. Configure the branch office IP phones with regions.
D. Configure the branch office IP phones with locations.
Answer: B
Q70. Which two options are valid service parameter settings that are used to set up proper video QoS behavior across the Cisco Unified Communications Manager infrastructure? (Choose two.)
A. DSCP for Video Calls when RSVP Fails
B. Default Intraregion Min Video Call Bit Rate (Includes Audio)
C. Default Interregion Max Video Call Bit Rate (Includes Audio)
D. DSCP for Video Signaling
E. DSCP for Video Signaling when RSVP Fails
Answer: A,C
Q71. Which action configures AAR to route the calls that have been rejected by the gatekeeper CAC through the PSTN?
A. Configure Cisco IP Phones for AAR.
B. Configure AAR to work with SRST.
C. Configure AAR to work with CTI route points.
D. This configuration is not possible using AAR.
Answer: D
Q72. Refer to the exhibit.
Which CSS is used at the HQ Cisco Unified Communications Manager to reroute calls via the PSTN when the SAF network is unavailable?
A. the phone device CSS
B. the phone line CSS
C. the phone line/device combined CSS
D. the SAF CSS configured on the CCD requesting service
E. the phone AAR CSS configured at the phone device
F. No special CSS is required. If SAF patterns are accessible, the PSTN reroute is automatic.
Answer: E
Q73. Refer to the exhibit.
Which statement about the configuration between the Default and BR regions is true?
A. Calls between the two regions can use either 64 kbps or 8 kbps.
B. Calls between the two regions can use only the G.729 codec.
C. Only 64 kbps will be used between the two regions because the link is "lossy".
D. Both codecs can be used depending on the loss statistics of the link. When lossy conditions are high, the G.711 codec will be used.
Answer: B
Q74. Which statement about SIP precondition is most correct?
A. When configuring SIP precondition, the SIP trunk must have access to an RSVP agent.
B. When configuring SIP precondition, the IP phones must have access to an RSVP agent.
C. When configuring SIP precondition, the IP phones and SIP trunk must have access to an RSVP agent.
D. RSVP agents are only required for the IP phones. SIP trunks require RSVP agents only when fall back to local RSVP is configured.
E. SIP trunk will always require RSVP agents regardless of what RSVP type is configured.
Answer: D
Q75. When an incoming PSTN call arrives at an H.323 gateway, how does the called number get normalized to an internal directory number in Cisco Unified Communications Manager?
A. Normalization is done by configuring the significant digits for inbound calls on the H.323 gateway configuration in Cisco Unified Communications Manager.
B. Normalization is done using route patterns.
C. Normalization is done using the gateway incoming calling party prefixes based on number type.
D. Normalization is achieved by local route group that is assigned to the H.323 gateway.
Answer: A