[Free] 2017(Apr) Ensurepass Passguide Cisco 400-051 Latest Dumps 181-190

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CCIE Collaboration Written Exam

QUESTION 181

Refer to the exhibit. Which number is sent as the caller ID when a user at extension 5001 places a call that matches this translation profile?

 

clip_image001

 

A.

14087775001

B.

+4087775001

C.

4087750001

D.

+14087775001

 

Correct Answer: D

Explanation:

When someone dials 5001, it will match rule 2 because it exactly starts with 5(five) using the ^ sign and ends with [0-9] followed by $. In replace pattern you can see +1408777 & \0 means all set of match pattern. Thus, +14087775001.

 

 

 

 

QUESTION 182

Which two are characteristics of jitter buffers? (Choose two.)

 

A.

Jitter buffers are used to change asynchronous packet arrivals into a synchronous stream by turning variable network delays into constant delays at the destination end systems.

B.

Jitter buffers are used to change asynchronous packet arrivals into a synchronous stream by turning variable network delays into constant delays at the sending systems.

C.

The role of the jitter buffer is to balance the delay and the probability of interrupted playout due to late packets.

D.

The role of the jitter buffer is to queue late packets and reorder out-of-order packets.

E.

Jitter buffers are used to change asynchronous packet arrivals into a synchronous stream by queuing packets into constant delays at the sending systems.

 

Correct Answer: AC

Explanation:

Jitter buffers are used to remove the effects of jitter so that asynchronous packet arrivals are changed to a synchronous stream. The jitter buffer trades off between delay and the probability of interrupted playout because of late packets (discard).

 

Reference:

http://www.appneta.com/blog/jitter-voip/

 

 

QUESTION 183

Refer to the exhibit. Which two statements about calls that match dial-peer voice 7 voip are true? (Choose two.)

 

clip_image002

 

A.

All calls that match dial-peer voice 7 use G.711.

B.

All calls that match dial-peer voice 7 have the Diversion header removed from SIP Invites.

C.

All calls that match dial-peer voice 7 use NOTIFY-based, out-of-band DTMF relay.

D.

All calls that match dial-peer voice 7 are marked with DSCP 32.

E.

All calls that match dial-peer voice 7 are marked with DSCP 34.

 

Correct Answer: BE

Explanation:

Dial peer 7 refers to SIP profile 102, which we can see is configured to have the Diversion header removed from SIP Invites.

Dial peer 7 marks traffic with AF41, which is equivalent to DSCP 34.

 

 

QUESTION 184

Which statement about a virtual SNR DN-configured Cisco Unified Communications Manager Express-enabled Cisco IOS router is true?

 

A.

Virtual SNR DN supports either SCCP or SIP IP phone DNs.

B.

A virtual SNR DN is a DN that is associated with multiple registered IP phones.

C.

Calls in progress can be pulled back from the phone that is associated with the virtual SNR DN.

D.

The SNR feature can only be invoked if the virtual SNR DN is associated with at least one registered IP phone.

E.

A call that arrived before a virtual SNR DN is associated with a registered phone, and still exists after association is made, but cannot be answered from the phone.

 

Correct Answer: E

Explanation:

Virtual SNR DN only supports Cisco Unified SCCP IP phone DNs.

Virtual SNR DN provides no mid-call support.

Mid-calls are either of the following:

 

Calls that arrive before the DN is associated with a registered phone and is still present after the DN is associated with the phone.

Calls that arrive for a registered DN that changes state from registered to virtual and back to registered.

Mid-calls cannot be pulled back, answered, or terminated from the phone associated with the DN.

State of the virtual DN transitions from ringing to hold or remains on hold as a registered DN.

 

Reference:

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/admin/configuration/guide/cmeadm/cmesnr.html

 

 

QUESTION 185

Which three options are valid per-session video conference participants supported on the Cisco Integrated Router Generation 2 with packet voice and video digital signal processor 3? (Choose three.)

 

A.

3

B.

4

C.

6

D.

8

E.

9

F.

12

G.

16

 

Correct Answer: BDG

Explanation:

The integrated video conferencing services use the same DSP resources on PVDM3s that are used for widely deployed ISR G2 voice capabilities. These modules, in conjunction with Cisco IOS Software, perform audio and video mixing, video transcoding for certain resolutions, and other functions for video endpoints. PVDM3 modules support flexible media resources and conference profile management to maximize capacity with predictable end-user experiences. Both homogenous and heterogonous video conferences are supported. A homogenous con
ference refers to one in which participants connect to the ISR G2 with devices that support the same video format attributes (for example, the same codec, resolution, frame rate, and bit rate). A heterogeneous conference refers to one in which participants can connect to a conference bridge with devices that support different video format attributes. Each conference allows 4-, 8-, or 16-party participants.

 

Reference:

http://www.cisco.com/c/en/us/products/collateral/unified-communications/voice-video-conferencing-isr-routers/data_sheet_c78-649427.html

 

 

QUESTION 186

Refer to the exhibit. How many calls, inbound and outbound combined, are supported on the IP phone?

 

clip_image003

 

A.

1

B.

2

C.

8

D.

12

E.

50

 

Correct Answer: E

 

 

 

QUESTION 187

Refer to the exhibit. Assume the B-ACD configuration on a Cisco Unified Communications Manager Express router is operational. How much time does a member of the hunt group have to answer a queue call that is ringing on their extension?

 

clip_image004

 

A.

5 seconds

B.

10 seconds

C.

20 seconds

D.

30 seconds

E.

40 seconds

 

Correct Answer: B

Explanation:

As you can see the timeout 10 sec in ephone-hunt 1 means hunt group membes have to answer the queued call within 10 sec.

 

 

QUESTION 188

Refer to the exhibit. In an effort to troubleshoot a caller ID delivery problem, a customer emailed you the voice port configuration on a Cisco IOS router. Which type of voice port is it?

 

clip_image005

 

A.

FXS

B.

E&M

C.

BRI

D.

FXO

E.

DID

 

Correct Answer: D

Explanation:

Configuring FXS and FXO Voice Ports to Support Caller ID To configure caller-ID on FXS and FXO voice ports, use the following commands beginning in global configuration mode:

 

Command

Purpose

 

Step 1

Router(config)# caller-id enable

 

Enables caller ID. This command applies to FXS voice ports that send caller-ID information and to FXO ports that receive it. By default caller ID is disabled.

 

Note

If the station-id or a caller-id alerting command is configured on the voice port, these automatically enable caller ID, and the caller-id enable command is not necessary.

 

Step 2

Router(config-voiceport)# station-id name name

 

Configures the station name on FXS voice ports connected to user telephone sets. This sets the caller-ID information for on-net calls originated by the FXS port. You can also configure the station name on an FXO port of a router for which incoming Caller ID from the PSTN subscriber line is expected. In this case, if no caller-ID information is included on the incoming PSTN call, the call recipient receives the information configured on the FXO port instead. If the PSTN subscriber line does provide caller-ID information, this information is used and the configured station name is ignored.

The name argument is a character string of 1 to 15 characters identifying the station. Note This command applies only to caller-ID calls, not Automatic Number Identification (ANI) calls. ANI supplies calling number identification only.

 

Step 3

Router(config-voiceport)# station-id number number

 

Configure the station number on FXS voice ports connected to user telephone sets. This sets the caller-ID information for on-net calls originated by the FXS port. You can also configure the station number on an FXO port of a router for which incoming caller ID from the
PSTN subscriber line is expected. In this case, if no caller-ID information is included on the incoming PSTN call, the call recipient receives the information configured on the FXO port instead. If the PSTN subscriber line does provide caller-ID information, this information is used and the configured station name is ignored. If the caller-ID station number is not provided by either the incoming PSTN caller ID or by the station number configuration, the calling number included with the on-net routed call is determined by Cisco IOS software by using a reverse dial-peer search. In this case, the number is obtained by searching for a POTS dial-peer that refers to the voice-port and the destination-pattern number from that dial-peer is used. Number is a string of 1 to 15 characters identifying the station telephone or extension number.

 

Reference:

http://www.cisco.com/c/en/us/td/docs/ios/12_2/voice/configuration/guide/fvvfax_c/vvfclid.html

 

 

QUESTION 189

Refer to the exhibit. Assuming this NFAS-enabled T1 PRI configuration on a Cisco IOS router is fully functional, what will the controller T1 1/1 D-channel status be in the output of the show isdn status command?

 

clip_image007

 

A.

MULTIPLE_FRAME_ESTABLISHED

B.

TEI_ASSIGNED

C.

AWAITING_ESTABLISHMENT

D.

STANDBY

E.

INITIALIZED

 

Correct Answer: B

Explanation:

TEI_ASSIGNED, which indicates that the PRI does not exchange Layer 2 frames with the switch. Use the show controller t1 x command to first check the controller t1 circuit, and verify whether it is clean (that is, it has no errors) before you troubleshoot ISDN Layer 2 problem with the debug isdn q921.

 

 

QUESTION 190

Refer to the exhibit. Which out-of-dialog SIP OPTIONS ping response put dial-peer tag 1111 into its current operational state?

 

clip_image009

 

A.

501 Not Implemented

B.

504 Server Time-out

C.

408 Request Timeout

D.

486 Busy Here

E.

503 Service Unavailable

 

Correct Answer: E

Explanation:

SIP 503 Service Unavailable is commonly seen in a VoIP network when a SIP device (such as a SIP server) is knowingly unable to process a call. Typically when this happens the endpoint that originated the Invite will try the next available host it receives in the SIP Contact header.

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