Download New Latest (May) Cisco 642-437 Actual Tests 41-50

Ensurepass

 

QUESTION 41

DRAG DROP  (Topic 1)

 

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Answer:

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A.

 

B.

 

C.

 

D.

 

 

Answer:

 

 

QUESTION 42  (Topic 1)

 

 

Voice packets are arriving at a destination with a variance of between 20 and 50 milliseconds. If the jitter buffer has a capacity of 30 milliseconds, what is the impact on the audio at the receivers IP phone?Voice packets are arriving at a destination with a variance of between 20 and 50 milliseconds. If the jitter buffer has a capacity of 30 milliseconds, what is the impact on the audio at the receivers IP phone?

 

A.

The jitter buffer will replay the previous voice packets to replace those packets that exceed 30 milliseconds to avoid speech gaps.

B.

There will be no impact the audio stream because the audio packets are arriving in the jitter buffer window.

C.

The DSP will automatically increase the jitter buffer size after sampling the range of incoming voice packets to accommodate the wider range in variation of voice packet arrival times to avoid voice gaps.

D.

The IP phone will negotiate in mid-call a lower bandwidth codec to reduce the delay in the arrival of voice packets to avoid voice gaps.

 

Answer: B

 

 

QUESTION 43  (Topic 1)

 

Your PSTN carrier sends digits to your T1 PRI circuit in a digit-by-digit format. How must the T1 PRI circuit be configured to support this capability?

 

A.

The T1 PRI controller supports either en-bloc or digit-by-digit formats natively.

B.

The serial interface that is associated with the T1 controller needs to include the isdn incoming-voice command.

C.

The T1 controller needs to include the isdn overlap-receiving command.

D.

The serial interface that is associated with the T1/E1 controller needs to include the isdn overlap-receiving command.

 

Answer: D

Explanation: Configuring Overlap-receiving on the D-channel changes the way routers behave when receiving ISDN calls. Overlap receiving allows the matching of dial peers as the digits are being received. The router responds to the setup message with a SETUP ACK. This informs the network that it is ready to receive further information messages containing additional call routing elements.

http://www.cisco.com/en/US/tech/tk801/tk133/technologies_tech_note09186a00800b48cb.

shtml

 

 

QUESTION 44

DRAG DROP  (Topic 1)

 

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Answer:

clip_image004

 

A.

 

B.

 

C.

 

D.

 

 

Answer:

 

 

QUESTION 45  (Topic 1)

 

Which three functions are associated with MGCP? (Choose three.)

 

A.

Control is implemented by a series of plain-text commands that are sent over UDP port 2427 between Cisco Unified Communications Manager and the gateway.

B.

A PRI backhaul channel forwards PRI Layer 2 (Q.921) signaling information via a TCP connection from the gateway to the call agent.

C.

MGCP uses a separate channel for backhauling signaling information between the call agent and the gateway.

D.

The gateway maintains a separate dial plan for redundancy in case the call agent fails.

E.

Users query the call agent to determine the location of the call recipient.

F.

A call agent uses control messages to direct its gateways and their operational behavior.

 

Answer: ACF

Explanation: MGCP is a plain-text protocol used by call-control devices to manage IP Telephony gateways. MGCP is a master/slave protocol that allows a call control device to take control of a specific port on a gateway. With this protocol, the Cisco CallManager knows and controls the state of each individual port on the gateway. It allows complete control of the dial plan from Cisco CallManager, and gives CallManager per-port control of connections to the PSTN, legacy PBX, voice mail systems, POTS phones and so forth. This is implemented with the use of a series of plain-text commands sent over User Datagram Protocol (UDP) port 2427 between the Cisco CallManager and the gateway.

 

Another concept relevant to the MGCP implementation with Cisco CallManager is PRI Backhaul. This occurs when Cisco CallManager takes control of the Q.931 signaling data used on an ISDN PRI. The one thing that distinguishes a PRI from other interfaces is the fact that the data that is received from the PSTN on the D-channel and needs to be carried in its raw form back to the Cisco CallManager to be processed. The gateway does not process or change this signaling data, it simply passes it onto the Cisco CallManager through TCP port 2428. The gateway is still responsible for the termination of the Layer 2 data. That means that all the Q.921 data-link layer connection protocols are terminated on the gateway, but everything above that (Q.931 network layer data and beyond) is passed onto the Cisco CallManager. This als
o means that the gateway does not bring up the D- channel unless it can communicate with Cisco CallManager to backhaul the Q.931 messages contained in the D-channel.

 

http://www.cisco.com/en/US/tech/tk1077/technologies_tech_note09186a00801da84e.shtml

 

 

QUESTION 46  (Topic 1)

 

Refer to the exhibit. When an international call to 90114989531212001 is placed from extension 2001, which of the following statements is true?

 

 

 

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A.

The call will fail because no incoming COR list is applied.

B.

The call will succeed because the incoming COR list is a superset of the outgoing COR list.

C.

The call will fail because the incoming COR list is not a superset of the outgoing COR list

D.

The call will succeed because the incoming COR list has the highest priority, by default, when no incoming COR list is applied.

 

Answer: D

Explanation: By default, an incoming call leg has the highest COR priority and the outgoing COR list has the lowest COR priority. This means that if there is no COR configuration for incoming calls on a dial-peer, then you can make a call from this dial-peer (a phone attached to this dial-peer) going out of any other dial-peer, irrespective of the COR configuration on that dial-peer.

http://www.cisco.com/en/US/tech/tk652/tk90/technologies_configuration_example09186a0 08019d649.shtml

 

 

QUESTION 47  (Topic 1)

 

When a Cisco Unified Border Element is deployed to support RSVP-based CAC, which media flow method is required?

 

A.

RSVP-based CAC can be supported with either media flow-through or media flow- around if the Cisco Unified Communications Manager is configured as an RSVP agent.

B.

RSVP-based CAC only supports media flow-around.

C.

The Cisco Unified Border Element does not have to participate in the RSVP message exchange and will pass RSVP messages through unchanged using media flow-around.

D.

RSVP-based CAC requires Cisco Unified Border Element to use media flow-through.

 

Answer: D

 

 

QUESTION 48  (Topic 1)

 

Refer to the exhibit. When 5551234 is being matched with the outgoing dial peer that is shown in the exhibit, which of the following called numbers will be sent to the VoIP network?

 

clip_image006

 

A.

5551234

B.

1234

C.

555

D.

Null

E.

5

F.

51234

 

Answer: D

 

 

QUESTION 49  (Topic 1)

 

Refer to the exhibit.

 

 

 

clip_image007

 

How does a switch port that receives marked traffic from a Cisco IP phone use the mls qos trust cos command?

 

A.

The CoS setting is modified according to the CoS-to-DSCP map.

B.

CoS is used to select the ingress and egress queues.

C.

For non-IP packets, the CoS is set to 7 and DSCP-to-CoS mapping is not applied.

D.

The DSCP-to-CoS map is applied.

 

Answer: A

 

 

QUESTION 50  (Topic 1)

 

Refer to the exhibit. Which type of ephone-dn is configured for the two ephones that are shown?

 

clip_image008

 

A.

single-line-octo

B.

hunt line

C.

shared-line, nonexclusive

D.

two directory numbers with one telephone number

E.

shared-line, overlay

F.

octo-line

 

Answer: E

Explanation: The above exhibit shows the configuration for a simple shared-line overlay set. The primary ephone-dn that is configured for each phone is unique while the remaining ephone-dns 10, 11, and 12 are shared in the overlay set on both phones. The primary ephone-dn in a shared-line overlay set is configured unique to the phone to guarantee that the phone has a line available for outgoing calls, and to ensure that the phone user can obtain dial-tone even when there are no idle lines available in the rest of the shared-line overlay set. Using a unique ephone-dn also provides a unique calling party identity on outbound calls made by the phone so that the called user can see which specific phone is calling.

http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmeco ver.html#wp1099687

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