만약 아직도  Cisco 642-457 (Implementing Cisco Unified Communications Manager, Part 2 v8.0 (CIPT2 v8.0)) 시험패스를 위하여 고군분투하고 있다면 바로 우리 Pass4Test를 선택함으로 여러분의 고민을 날려버릴수 있습니다. 우리 Pass4Test에서는 최고의 최신의 덤프자료를 제공 함으로 여러분을 도와  Cisco 642-457 (Implementing Cisco Unified Communications Manager, Part 2 v8.0 (CIPT2 v8.0)) 인증자격증을 쉽게 취득할 수 있게 해드립니다. 만약  Cisco 642-457 (Implementing Cisco Unified Commu nications Manager, Part 2 v8.0 (CIPT2 v8.0)) 인증시험으로 한층 업그레이드된 자신을 만나고 싶다면 우리 Pass4Test선택을 후회하지 않을 것입니다. 우리 Pass4Test과의 만남으로 여러분은 한번에 아주 간편하게  Cisco 642-457 (Implementing Cisco Unified Communications Manager, Part 2 v8.0 (CIPT2 v8.0)) 시험을 패스하실 수 있으며  Cisco 642-457 (Implementing Cisco Unified Communications Manager, Part 2 v8.0 (CIPT2 v8.0)) 자격증으로 완벽한 스펙을 쌓으실 수 있습니다. Pass4Test에서 제공하는 제품들은 품질이 아주 좋으며 또 업뎃속도도 아주 빠릅니다 . 우리가 제공하는  Cisco 642-457 (Implementing Cisco Unified Communications Manager, Part 2 v8.0 (CIPT2 v8.0)) 인증시험관련 덤프를 구매하신다면  Cisco 642-457 (Implementing Cisco Unified Communications Manager, Part 2 v8.0 (CIPT2 v8.0)) 시험은 손쉽게 성공적으로 패스하실 수 있습니다.Pass4Test는 여러분의 꿈을 이루어줄 뿐만 아니라 일년무료 업뎃서비스도 제공해드립니다. Pass4Test에서 제공하는 덤프로 여러분은 100%시험을 패스하실수 있고  Cisco 642-457 (Implementing Cisco Unified Communications Manager, Part 2 v8.0 (CIPT2 v8.0)) 자격증을 취득하실 수 있습니다.지금 바로 사이트에서  Cisco 642-457 (Implementing Cisco Unified Communications Manager, Part 2 v8.0 (CIPT2 v8.0)) 덤프데모 즉 덤프의 일부 문제와 답을 다운 받으셔서 체험하실 수 있습니다.

 

 

NO.1 Refer to the exhibit.
The exhibit shows centralized Cisco Unified Communications Manager configuration components for
TEHO calls to U.S. area code 408 from the U.K. The PSTN access code for the U.K. is 9 and 001 for
international calls to the U.S. To match the US-TEHO pattern \+!, how should the translation pattern be
configured?
A.9001.4085551234 with the Called Party Transformation:
Discard Digits PreDot
Prefix Digits Outgoing Calls: +
B.9.0014085551234 with the Called Party Transformation:
Discard Digits PreDot
Prefix Digits Outgoing Calls: +1
C.900.14085551234 with the Called Party Transformation:
Discard Digits PreDot
Prefix Digits Outgoing Calls: +1
D.900.14085551234 with the Called Party Transformation:
Discard Digits PreDot
Prefix Digits Outgoing Calls: +
E.001.4085551234 with the Called Party Transformation:
Prefix Digits Outgoing Calls: +
Answer: D

NO.2 Refer to the exhibit.
What should the destination IP address be configured as on the HQ and BR1 SIP trunks?
A.The HQ SIP trunk destination IP address should be 10.1.6.10. The BR1 SIP trunk destination IP
address should be 10.1.5.10.
B.The destination IP address is not configurable. Each cluster will learn about the remote trunk IP address
through SAF learned routes.
C.The destination IP address will be learned automatically and configured on the SIP trunks after the
Cisco Unified Communications Managers discover themselves.
D.The HQ SIP trunk destination IP address should be the HQ SAF Forwarder IP address. The BR1 SIP
trunk destination IP address should be the BR1 SAF Forwarder IP address.
Answer: B

NO.3 Refer to the exhibit.
The exhibit shows centralized Cisco Unified Communications Manager configuration components for
TEHO calls to U.S. area code 408 from the U.K. The PSTN access code for the U.K. is 9 and 001 for
international calls to the U.S. What should the TEHO-US route list configuration consist of?
A.First route group should point only to the U.K. gateway. The second route group should point to the U.S.
gateway.
B.First route group should be only the local route group. The second route group should point to the U.S.
gateway.
C.First route group should point only to the U.S. gateway. The second route group should be the local
route group.
D.The TEHO-US route list should contain only the local route group. The globalized configuration means
that the appropriate gateway will be selected automatically.
E.The \+! route pattern should point directly to the U.S. gateway.
Answer: C

NO.4 Which statement about Service Advertisement Framework is true?
A.SAF requires that the EIGRP be configured on all routers, including non-SAF routers.
B.SAF requires that the EIGRP be configured only on SAF routers. Non-SAF routers act as an IP cloud.
C.SAF has no dependency on the underlying routing protocol, as long as it is a dynamic routing protocol.
Static routes are not supported.
D.SAF operates on any dynamic or static IP routing configuration. SAF is totally independent of the
underlying routing protocol.
Answer: D

NO.5 Refer to the exhibit.
Which configuration elements must match for adjacent neighbors to establish a SAF neighbor
relationship?
A. the label name specified in the router eigrp command
B. the autonomous-system number specified in the service-family ipv4 autonomous-system command
C. the sf-interface configuration
D. the topology base configurations
E. the label name specified in the router eigrp command and the autonomous-system number
Answer: B

NO.6 Refer to the exhibit.
The exhibit shows centralized Cisco Unified Communications Manager configuration components for
TEHO calls to U.S. area code 408 from the U.K. The PSTN access code for the U.K. is 9 and 001 for
international calls to the U.S. Assuming the PSTN does not accept globalized numbers with + prefix. What
should the Called Party Transformation Pattern at the U.S. gateway be configured as?
A.\+.! with the following Called Party Transformation:
Discard Digits PreDot
Prefix Digits Outgoing Calls: +
B.\+1.! with the following Called Party Transformation:
Discard Digits PreDot
Prefix Digits Outgoing Calls: None
C.\+408.! with the following Called Party Transformation:
Discard Digits PreDot
Prefix Digits Outgoing Calls: 1
D.\+1408.! with the following Called Party Transformation:
Discard Digits PreDot
Prefix Digits Outgoing Calls: None
E.\+1.408! with the following Called Party Transformation:
Discard Digits PreDot
Prefix Digits Outgoing Calls: None
Answer: D

NO.7 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

NO.8 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: AB

NO.9 Assume that the Cisco IOS SAF Forwarder is configured correctly. Which minimum configurations on
Cisco Unified Communications Manager are needed for the SAF registration to take place?
A.SAF Trunk, SAF Security Profile, SAF Forwarder, and CCD Advertising Service
B.SAF Trunk, SAF Security Profile, SAF Forwarder, and CCD Requesting Service
C.SAF Trunk, SAF Security Profile, SAF Forwarder, CCD Requesting Service, and CCD Advertising
Service
D.SAF Trunk, SAF Security Profile, and SAF Forwarder
E.SAF Trunk, CCD Requesting Service, and CCD Advertising Service
Answer: B

NO.10 Refer to the exhibit.
Which trunks would be most suitable for Connection Y?
A.an H.225 trunk (gatekeeper-controlled)
B.intercluster trunk (gatekeeper-controlled)
C.a SIP trunk on the U.S. cluster and an intercluster trunk on the remote cluster
D.intercluster trunk (nongatekeeper-controlled)
E.No extra connections are required. As long as IP connectivity exists, you need only configure a route
pattern for each site. The Cisco Unified Communications Manager will automatically forward the calls over
the WAN if the destination directory number is not registered locally.
Answer: D

NO.11 When an incoming PSTN call arrives at an MGCP 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 MGCP gateway.
B.Normalization is done using route patterns.
C.Normalization is done using the gateway incoming called party prefixes based on number type.
D.Normalization is done using the gateway incoming calling party prefixes based on number type.
E.Normalization is achieved by local route group that is assigned to the MGCP gateway.
Answer: A

NO.12 A Cisco 3825 needs to be added in Cisco Unified Communications Manager as an H.323 gateway.
What should the gateway type be?
A.H.323 gateway
B.Cisco 3825
C.Cisco 3800 series router. The specific model will be selected after the Cisco 3800 is selected.
D.The gateway can be added either as an H.323 gateway or Cisco 3800 series router.
E.The gateway can be added either as an H.323 gateway or Cisco 3825 series router.
Answer: A

NO.13 Which statement best describes globalized call routing in Cisco Unified Communications Manager?
A.All incoming calling numbers on the phones are displayed as an E.164 with the + prefix.
B.Call routing is based on numbers represented as an E.164 with the + prefix format.
C.All called numbers sent out to the PSTN are in E.164 with the + prefix format.
D.The CSS of all phones contain partitions assigned to route patterns that are in global format.
E.All phone directory numbers are configured as an E.164 with the + prefix.
Answer: B

NO.14 When an incoming PSTN call arrives at an MGCP gateway, how does the calling number get
normalized to a global E.164 number with the + prefix in Cisco Unified Communications Manager?
A.Normalization is done using translation patterns.
B.Normalization is done using route patterns.
C.Normalization is done using the gateway incoming called party prefixes based on number type.
D.Normalization is done using the gateway incoming calling party prefixes based on number type.
E.Normalization is achieved by local route group that is assigned to the MGCP gateway.
Answer: D

NO.15 Which process can localize a global E.164 with + prefix calling numbers for inbound calls to an IP phone
so that users see the calling number in a local format?
A.Calling number localization is done using translation patterns.
B.Calling number localization is done using route patterns.
C.Calling number localization is done by configuring a calling party transformation CSS at the phone.
D.Calling number localization is done by configuring a calling party transformation CSS at the gateway.
E.Calling number localization is done by configuring the phone directory number in a localized format.
Answer: C