Download New Latest (May) Cisco 642-437 Actual Tests 11-20

Ensurepass

 

QUESTION 11

DRAG DROP  (Topic 1)

 

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Answer:

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A.

 

B.

 

C.

 

D.

 

 

Answer:

 

 

QUESTION 12  (Topic 1)

 

When configuring AutoQoS VoIP on a Cisco Catalyst switch how is the configuration performed?

 

A.

The auto qos voip command is applied to each interface.

B.

The auto qos voip command is applied globally in the switch.

C.

Each interface will need either the auto qos voip cisco-phone or auto qos voip trust on each interface depending on the upstream device.

D.

Each interface will need either the auto qos voip trust cisco-phone or auto qos voip trust trust on each interface depending on the upstream device.

 

Answer: C

Explanation: The QoS mechanisms on a Catalyst switch differ from those QoS mechanisms found on a router. For example, while a router uses LLQ as a priority queuing strategy, a Catalyst switch might use weighted round-robin (WRR) as a priority queuing strategy. Fortunately, the AutoQoS feature available on some Catalyst switch models applies voice-specific QoS features globally to a Catalyst switch and also at the port level. To configure AutoQoS on supported Catalyst switch platforms, issue the following command from interface configuration mode:

 

 

 

Switch(config-if)#auto qos voip [trust | cisco-phone] If the trust option is used in the previous command, the Catalyst switch makes queuing decisions based on Layer 2 Class of Service (CoS) markings. However, if the cisco-phone option is used, the Catalyst switch makes queuing decisions based on CoS markings originating from a Cisco IP phone. The switch detects the presence of a Cisco IP phone via the CDP.

http://www.cisco.com/en/US/docs/ios/12_2t/12_2t15/feature/guide/ftautoq1.html

 

 

QUESTION 13  (Topic 1)

 

Refer to the exhibit. An administrator is migrating a PBX telephony system to a VoIP solution using a fixed numbering plan. The extension numbers and PSTN DIDs cannot be changed. Which of the following methods can be used in order to reach the individual extensions at Site B when called via the PSTN?

 

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A.< /p>

The administrator can replace the last three digits of the DID with xxx to cover the individual extensions.

B.

The administrator can replace the last three digits of the DID with xxx and use translation rules to map the individual extensions.

C.

The administrator needs to implement an auto-attendant solution where individual extensions can be dialed.

D.

The administrator needs to map the last four digits in the DID to the extension numbers

 

 

using translation rules.

 

Answer: D

 

 

QUESTION 14  (Topic 1)

 

Which QoS technology provides a strict priority queuing scheme that allows delay-sensitive data such as voice to be dequeued and sent before packets in other queues are dequeued, and also works with WFQ and CBWFQ.


 

A.

header compression

B.

IP RTP Priority and Frame Relay IP RTP Priority

C.

RSVP

D.

low latency queuing

E.

FRF.12

 

Answer: B

 

 

QUESTION 15  (Topic 1)

 

Which codec complexity type will offer the greatest number of voice channels, provided that the complexity type is compatible with the particular codecs that are in use?

 

A.

low complexity

B.

medium complexity

C.

high complexity

D.

flex complexity

 

Answer: A

Explanation: Codec complexity refers to the amount of processing required to perform voice compression. Codec complexity affects the call density–the number of calls reconciled on the DSPs. With higher codec complexity, fewer calls can be handled. Select a higher codec complexity if that is required to support a particular codec or combination of codecs. Select a lower codec complexity to support the greatest number of voice channels, provided that the lower complexity is compatible with the particular codecs in use. http://www.cisco.com/en/US/docs/ios/12_3t/voice/command/reference/vrht_c6_ps5207_TS D_Products_Command_Reference_Chapter.html

 

 

QUESTION 16

DRAG DROP  (Topic 1)

 

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Answer:

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A.

 

B.

 

C.

 

D.

 

 

Answer:

 

 

QUESTION 17  (Topic 1)

 

Which statement best describes dial peers in a voice gateway. (Choose two.)

 

A.

Dial peers are call legs that are used to identify call source and destination endpoints and to define the characteristics that are applied to each call leg in the call connection.

B.

Dial peers are configured with call legs that are essential to implementing dial plans and providing voice services over an IP packet network.

C.

A dial peer is a physical addressable endpoint in a voice gateway.

D.

Dial peers create physical connections called call legs to complete an end-to-end call.

 

Answer: AC

 

 

QUESTION 18  (Topic 1)

 

Which two statements are true regarding SCCP? (Choose two.)

 

A.

SCCP requires each endpoint or gateway event to be communicated to Cisco Unified Communications Manager.

B.

Endpoints can operate autonomously if communication with Cisco Unified Communications Manager is lost.

C.

SCCP may interoperate with H.323 endpoints if it is implemented with Cisco Unified Communications Manager.

D.

Endpoints and gateways maintain the dial plan.

E.

SCCP uses hex messages for communication.

 

Answer: AC

Explanation: The Skinny client (i.e. an Ethernet Phone) uses TCP/IP to transmit and receive calls. Skinny messages are carried above TCP and use port 2000. Cisco IP Phones that use SCCP can coexist in an H.323 environment. When used with CUCM, the SCCP client can interoperate with H.323-compliant terminals. The client communicates with the CUCM using TCP/IP-based communication to establish a call with another H.323- compliant end station. Once the CUCM has established the call, the two H.323 end stations use connectionless UDP/IP-based communication for audio transmissions. The CUCM acts as a proxy by processing all H.323 and SIP transactions. This allows the IP Phone to process the VoIP RTP data stream.

http://www.cisco.com/en/US/docs/voice_ip_comm/cata/186_188/2_15_ms/english/administ ration/guide/sccp/sccpaaph.pdf

 

 

QUESTION 19  (Topic 1)

 

 

Which types of voice ports allow a small office to provide outbound DNIS and inbound DID?

 

A.

FXS and FXO

B.

FXO and E&M

C.

FXS and FXS-DID

D.

FXS and E&M

E.

FXS-DID and FXO

 

Answer: E

Explanation: An FXO trunk is one of the simplest analog trunks available. Because Dialed Number Information Service (DNIS) information can only be sent out to the PSTN, no direct inward dialing (DID) is possible. ANI is supported for inbound calls. Two signaling types exist, loopstart and groundstart, with groundstart being the preferred method. An FXS DID trunk can receive only inbound calls, thus a combination of FXS DID, and FXO ports is required for inbound and outbound calls

http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/srnd/design/guide/gatewy.html#w p1052323

 

 

QUESTION 20  (Topic 1)

 

Which type of voice port supports immediate-start, wink-start, and delay-start followed by pulse or DTMF tones?

 

A.

FXS

B.

FXS-DID

C.

FXO

D.

E&M

 

Answer: D

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