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QUESTION 21
Refer to the exhibit. A user is going through a series of dialing steps on a SIP Type B IP phone (for example, a Cisco 7975) to call an SCCP IP phone. Both phones are registered to the same Cisco Unified Communications Manager cluster. Assuming the calling SIP phone is associated with a SIP Dial Rule with a pattern value of 2001, which statement about the call setup process of this call is true?

A.Each digit will arrive at Cisco Unified Communications Manager in a SIP NOTIFY message as a KPML
event, and Cisco Unified Communications Manager will extend the call as soon as the collected digits
match the extension of the SCCP IP phone, bypassing class of service configuration on both IP phones.
B.Each digit will arrive at Cisco Unified Communications Manager in a SIP NOTIFY message as a KPML
event. When the collected digits match the extension of the SCCP IP phone, Cisco Unified Communications
Manager
will extend the call only if the class of service configuration on both phones permits this action.
C.As soon as the user selects the Dial softkey, the SIP IP phone will forward all digits to Cisco Unified
Communications Manager in a SIP INVITE message. Cisco Unified Communications Manager will extend
the call as soon as the collected digits match the extension of the SCCP IP phone, bypassing class of service
configuration on both IP phones.
D.As soon as the user selects the Dial softkey, the SIP IP phone will forward all digits to Cisco Unified
Communications Manager in a SIP INVITE message. Cisco Unified Communications Manager will extend
the call only if class of service configuration on both phones permits this action.
E.The SIP IP phone will wait for the interdigit timer to expire, and then send all digits to Cisco Unified
Communications Manager in a SIP INVITE message. Cisco Unified Communications Manager will extend
the call as soon as the collected digits match the extension of the SCCP IP phone, bypassing class of
service configuration on both IP phones.

Answer: D
Explanation:
Cisco Type B SIP Phones offer functionality based SIP INVITE Message. Every key the end user presses triggers an individual SIP message. The first event is communicated with a SIP INVITE, but subsequent messages use SIP NOTIFY messages. The SIP NOTIFY messages send KPML events corresponding to any buttons or soft keys pressed by the user. Cisco Type B SIP IP Phones with SIP dial rules operate in the same manner as Cisco Type A phones with dial rules.

QUESTION 22
What does a comma accomplish when it is used in a SIP Dial Rule pattern that is associated with a Cisco 9971 IP Phone that is registered to Cisco Unified Communications Manager?

A.It inserts a 500-millisecond pause between digits.
B.It causes the phone to generate a secondary dial tone.
C.It is a delimiter and has no significant dialing impact.
D.It indicates a timeout value of 5000 milliseconds.
E.It is an obsolete parameter and will be ignored.

Answer: B
Explanation:
Comma is accepted in speed dial as delimiter and pause. -Comma used to delineate dial string, FAC, CMC, and post connect digits For post connect digits, commas insert a 2 second delay Commas may be duplicated to create longer delays

QUESTION 23
Which Call Admission Control mechanism is supported for the Cisco Extension Mobility Cross Cluster solution?

A.Location CAC
B.RSVP CAC
C.H.323 gatekeeper
D.intercluster Enhanced Location CAC
E.visiting cluster’s LBM hub

Answer: B
Explanation:
Configuring extension mobility cross cluster (EMCC) is nothing you should take lightly. EMCC requires a lot of configuration parameters including the exporting and importing of each neighbor cluster’s X.509v3 digital certificates. EMCC is supported over SIP trunks only. Presence is another feature that’s only supported over SIP trunks. If you want to be able to perform scalable Call Admission Control (CAC) in a distributed multi- cluster call processing model, you will need to point an H.225 or Gatekeeper controlled trunk to an H.323 Gatekeeper for CAC… but if you want to support presence and EMCC between clusters and maintain CAC.

QUESTION 24
Which two Cisco Unified Communications Manager SIP profile configuration parameters for a SIP intercluster trunk are mandatory to enable end-to-end RSVP SIP Preconditions between clusters? (Choose two.)

A.Set the RSVP over SIP parameter to Local RSVP.
B.Set the RSVP over SIP parameter to E2E.
C.Set the SIP Rel1XX Options parameter to Disabled.
D.Set the SIP Rel1XX Options parameter to Send PRACK If 1xx Contains SDP.
E.Set the SIP Rel1XX Options parameter to Send PRACK for All 1xx Messages.
F.Check the Fall Back to Local RSVP check box.

Answer: BD
Explanation:
Each Unified Communications Manager cluster and Unified CME should have the same configuration information. For example, Application ID should be the same on each Unified Communications Manager cluster and Unified CME. RSVP Service parameters should be the same on each Unified Communications Manager cluster.

QUESTION 25
What is the number of directory URIs with which a Cisco Unified Communications Manager directory number can be associated?

A.1
B.up to 2
C.up to 3
D.up to 4
E.up to 5

Answer: E
Explanation:
Cisco Unified Communications Manager supports dialing using directory URIs for call addressing. Directory URIs look like email addresses and follow the username@host format where the host portion is an IPv4 address or a fully qualified domain name. A directory URI is a uniform resource identifier, a string of characters that can be used to identify a directory number. If that directory number is assigned to a phone, Cisco Unified Communications Manager can route calls to that phone using the directory URI. URI dialing is available for SIP and SCCP endpoints that support directory URIs.

QUESTION 26
Which Cisco Unified Communications Manager partition will be associated with a directory URI that is configured for an end user with a primary extension?

A.null
B.none
C.directory URI
D.default
E.any partition that the Cisco Unified Communications Manager administrator desires

Answer: C
Explanation:
Cisco Unified Communications Manager supports dialing using directory URIs for call addressing. Directory URIs look like email addresses and follow the username@host format where the host portion is an IPv4 address or a fully qualified domain name. A directory URI is a uniform resource identifier, a string of characters that can be used to identify a directory number. If that directory number is assigned to a phone, Cisco Unified Communications Manager can route calls to that phone using the directory URI. URI dialing is available for SIP and SCCP endpoints that support directory URIs.

QUESTION 27
Which Call Control Discovery function allows the local Cisco Unified Communications Manager to listen for advertisements from remote call-control entities that use the SAF network?

A.CCD advertising service
B.CCD requesting service
C.SAF forwarder
D.SAF enabled trunks
E.CCD registration service

Answer: B
Explanation:
SAF and CCD will allow large distributed multi-cluster deployments to have the directory number (DN) ranges of each call routing element advertised dynamically over SAF. Cisco routers act as SAF Forwarders (SAFF), while the call routing elements (e.g. CUCM) act as clients that register with the routers to advertise their DN ranges and listen to the advertisements of other routers.

QUESTION 28
Which codec complexity mode, when deployed on Cisco IOS routers with DSPs using the C5510 chipset, supports the most G.711 calls per DSP?

A.Low
B.Medium
C.High
D.Secure
E.Flex

Answer: E
Explanation:
The flex parameter allows the complexity to automatically adjust to either medium or high complexity depending on the needs of a call. For example, if a call uses the G.711 codec, the C5510 chipset automatically adjusts to the medium-complexity mode. However, if the call uses G.729, the C5510 chipset uses the high complexity mode

QUESTION 29
When DSP oversubscription occurs on a Cisco IOS router using DSP modules that are based on the C5510 chipset, what will happen when an analog phone connected to a FXS port goes off-hook?

A.A fast busy tone will be played.
B.A slow busy tone will be played.
C.A network busy tone will be played.
D.A dial tone will be played, but digits will not be processed.
E.No tone will be played.

Answer: E
Explanation:
When DSP oversubscription occurs for both analog ports and digital ports, except PRI and BRI. FXO signaling and application controlled endpoints are not supported. This feature does not apply to insufficient DSP credits due to mid-call codec changes (while a call is already established).

QUESTION 30
Refer to the exhibit. How many simultaneous outbound calls are possible with this Cisco Unified Communications Manager Express configuration on these two phones?

A.6
B.7
C.8
D.9
E.11

Answer: C
Explanation:
Ephone is configured as octo line so maximum call number is 8 and it will be devided between lines.


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QUESTION 11
Which two responses are examples of client error responses in SIP protocol? (Choose two.)

A.    302 Moved Temporarily
B.    404 Not Found
C.    503 Service Unavailable
D.    502 Bad Gateway
E.    604 Does Not Exist Anywhere
F.    408 Request Timeout

Answer: BF
Explanation:
Client Error (400 to 499)–Request contains bad syntax or cannot be fulfilled at this server. This class of 400 to 499 contains only error messages.

QUESTION 12
Which H.245 information is exchanged within H.225 messages in H.323 Fast Connect?

A.    Terminal Capability Set
B.    Open Logical Channel
C.    Master-Slave Determination
D.    Call Setup
E.    Call Progress

Answer: B
Explanation:
With the standard H.245 negotiation, the two endpoints need three round- trips before they agree on the parameters of the audio/video channels (1. master/slave voting, 2. terminal capability set exchange, and finally, 3. opening the logical channels). In certain situations and especially with high-latency network links, this can last too long and users will notice the delay.

QUESTION 13
Which two compression formats for high-definition video have technical content that is identical to H.264? (Choose two.)

A.    MPEG-4 Part 10
B.    MPEG-4 Part 14
C.    MPEG-2 Part 7
D.    AVC
E.    VC3
F.    VP8

Answer: AD
Explanation:
MPEG-4 Part 10, also known as MPEG-4 AVC (Advanced Video Coding), is actually defined in an identical pair of standards maintained by different organizations, together known as the Joint Video Team (JVT). While MPEG-4 Part 10 is a ISO/IEC standard, it was developed in cooperation with the ITU, an organization heavily involved in broadcast television standards. Since the ITU designation for the standard is H.264, you may see MPEG-4 Part 10 video referred to as either AVC or H.264. Both are valid, and refer to the same standard.

QUESTION 14
Refer to the exhibit. A user is going through a series of dialing steps on an SCCP IP phone (extension 1001) to call another SCCP IP phone (extension 2003). Both phones are registered to the same Cisco Unified Communications Manager cluster. Which user inputs are sent from the calling IP phone to the Cisco Unified Communications Manager, in forms of SCCP messages, after the user pressed the Dial softkey? Note that the commas in answer choices below are logical separators, not part of the actual user input or SCCP messages.

A.    A separate SCCP message is sent to Cisco Unified Communications Manager for each of the following
user inputs: 2, 0, 0, 3.
B.    A separate SCCP message is sent to Cisco Unified Communications Manager for each of the following
user inputs: 2, 0, 1, <<, 0, 3.
C.    A single SCCP message is sent to Cisco Unified Communications Manager to report that digits 2003
have been dialed.
D.    A single SCCP message is sent to Cisco Unified Communications Manager to report that digits 201<<03
have been dialed.
E.    A separate SCCP message is sent to Cisco Unified Communications Manager for each of the following
user inputs: 2, 0, 1, <<, 2, 0, 0, 3.

Answer: C
Explanation:
After the user delete phone stop the digit by digit dialing and send it as a whole setup.

QUESTION 15
How are DTMF digits transported in RFC 2833?

A.    In the RTP stream with the named telephone events payload format.
B.    In the RTP stream with the regular audio payload format.
C.    In SIP NOTIFY messages.
D.    In SIP INFO messages.
E.    In SIP SUBSCRIBE messages.

Answer: A
Explanation:
DTMF digits and named telephone events are carried as part of the audio stream, and MUST use the same sequence number and time-stamp base as the regular audio channel to simplify the generation of audio waveforms at a gateway. The default clock frequency is 8,000 Hz, but the clock frequency can be redefined when assigning the dynamic payload type.

QUESTION 16
Refer to the exhibit. Which DTMF relay method is advertised when the originating SIP gateway sends an INVITE message with a Call-Info header shown?
 

A.    RFC 2833
B.    SIP INFO
C.    SIP NOTIFY
D.    SIP KPML
E.    In-band audio

Answer: C
Explanation:
You can develop user-specific applications that reside on your network entity and have the ability to subscribe for event services supported by the IMG. If the network entity wants the ability to detect an entered DTMF digit (only telephone event of “###” are currently supported) from the TDM-side of a call to the IP side of a call, the entity can subscribe to the IMG for these events and receive SIP NOTIFY events containing the digit event.

QUESTION 17
What is the maximum length of any numeric geographic area address in ITU recommendation E.164?

A.    15
B.    18
C.    21
D.    22
E.    25

Answer: A
Explanation:
E.164 defines a general format for international telephone numbers. Plan- conforming numbers are limited to a maximum of 15 digits. The presentation of numbers is usually prefixed with the character + (plus sign), indicating that the number includes the international country calling code (country code), and must typically be prefixed when dialing with the appropriate international call prefix, which is a trunk code to reach an international circuit from within the country of call origination.

QUESTION 18
According to ITU-T E.164 recommendations, which two fields in the National Significant Number code may be further subdivided? (Choose two.)

A.    Country Code
B.    National Destination Code
C.    Subscriber Number
D.    Regional Significant Number
E.    Local User Code
F.    National Numbering Plan

Answer: BC
Explanation:
A telephone number can have a maximum of 15 digits .The first part of the telephone number is the country code (one to three digits) .The second part is the national destination code (NDC). The last part is the subscriber number (SN). The NDC and SN together are collectively called the national (significant) number

QUESTION 19
Refer to the exhibit. A user is going through a series of dialing steps on a SIP Type B IP phone (for example, a Cisco 7975) to call an SCCP IP phone. Both phones are registered to the same Cisco Unified Communications Manager cluster. Assuming that the calling SIP phone is not associated with any SIP dial rules, which statement about how digits are forwarded to Cisco Unified Communications Manager for further call processing is true?

A.    Each digit is sent to Cisco Unified Communications Manager in a SIP NOTIFY message KPML event, at the
time that the user enters the digit on the keypad.
B.    The SIP IP phone will wait for the interdigit timer to expire, or for the Dial softkey to be selected before
sending each digit to Cisco Unified Communications Manager as a separate KPML event in a SIP NOTIFY
message.
C.    The SIP IP phone will wait for the interdigit timer to expire, or for the Dial softkey to be selected before
sending all digits to Cisco Unified Communications Manager in a SIP INVITE message.
D.    The SIP IP phone will wait for the interdigit timer to expire or for the Dial softkey to be selected before sending
the first digit in a SIP INVITE and the subsequent digits in SIP INFORMATION messages.
E.    The SIP IP phone will send all digits to Cisco Unified Communications Manager in a SIP INVITE message
as soon as the fourth digit is pressed.

Answer: A
Explanation:
KPML procedures use a SIP SUBSCRIBE message to register for DTMF digits. The digits themselves are delivered in NOTIFY messages containing an XML encoded body. And it is Out of Band DTMF

QUESTION 20
Refer to the exhibit. A user is going through a series of dialing steps on a SIP Type B IP phone (for example, a Cisco 7975) to call an SCCP IP phone. Both phones are registered to the same Cisco Unified Communications Manager cluster. Assuming the calling SIP phone is associated with a SIP dial rule with a pattern value of 2001, which statement about how digits are forwarded to Cisco Unified Communications Manager for further call processing is true?
 

A.    As each digit is pressed on the SIP IP phone, it is sent to Cisco Unified Communications Manager in
a SIP NOTIFY message as a KPML event.
B.    The SIP IP phone will wait for the interdigit timer to expire, and then send each digit to Cisco Unified
Communications Manager as a separate KPML event in a SIP NOTIFY message.
C.    The SIP IP phone will wait for the interdigit timer to expire, or for the Dial softkey to be selected before
sending all digits to Cisco Unified Communications Manager in a SIP INVITE message.
D.    The SIP IP phone will wait for the interdigit timer to expire, or for the Dial softkey to be selected before
sending the first digit in a SIP INVITE and the subsequent digits in SIP INFORMATION messages.
E.    The SIP IP phone will wait for the interdigit timer to expire, and then send all digits to Cisco Unified
Communications Manager in a SIP INVITE message.

Answer: E
Explanation:
Cisco Type B SIP Phones offer functionality based SIP INVITE Message. Every key the end user presses triggers an individual SIP message. The first event is communicated with a SIP INVITE, but subsequent messages use SIP NOTIFY messages. The SIP NOTIFY messages send KPML events corresponding to any buttons or soft keys pressed by the user. Cisco Type B SIP IP Phones with SIP dial rules operate in the same manner as Cisco Type A phones with dial rules.


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QUESTION 1
Which two SCCP call signaling messages are sent by an IP phone to Cisco Unified Communications Manager? (Choose two.)

A.    SoftKeyEvent
B.    OpenReceiveChannelAck
C.    StartMediaTransmission
D.    SelectSoftKeys
E.    CloseReceiveChannel
F.    StopTone

Answer: AB
Explanation:
This message indicates which soft key was pressed. Upon receipt of this mesage,CallManager invokesthe action associated with thepressed soft key. For example, if Hold was the pressed soft key,CallManager places the active call on user hold.In some trace files you might see a soft key number without the corresponding description. The following list defines each soft key number.

QUESTION 2
Which device is the initiator of a StationInit message in a Cisco Unified Communications Manager SDI trace?

A.    Cisco Unified Communications Manager
B.    MGCP gateway
C.    Cisco Music on Hold server
D.    SCCP IP phone
E.    SIP Proxy Server

Answer: D
Explanation:
StationInit means that an inbound Transmission Control Protocol (TCP) message
from a Skinny station reached CallManager. A Skinny station is any endpoint that uses the Skinny protocol to communicate with CallManager

QUESTION 3
Refer to the exhibit. You received this debug output to troubleshoot a Cisco IOS MGCP gateway problem at a customer site.
Which statement about this endpoint on the Cisco MGCP gateway is true?

A.    This endpoint is on a T1 Controller 0/1/0.
B.    This endpoint is on an E1 Controller 0/1/0.
C.    This endpoint is on a T1 Controller 0/1/1.
D.    This endpoint is on an E1 Controller 0/1/2.
E.    This endpoint is on an T1 Controller 0/1/2.

Answer: A

QUESTION 4
Refer to the exhibit. You received this debug output to troubleshoot a Cisco IOS MGCP gateway media-related problem at a customer site. What is the purpose of this message?

A.    The MGCP gateway is responding to an RQNT message from Cisco Unified Communications Manager
to poll the media capabilities on its endpoints.
B.    The MGCP gateway is responding to an AUEP message from Cisco Unified Communications Manager
to poll the media capabilities on its endpoints.
C.    The MGCP gateway is responding to an AUCX message from Cisco Unified Communications Manager
to poll the active calls on its endpoints.
D.    The MGCP gateway is responding to an MDCX message from Cisco Unified Communications Manager
during a call setup.
E.    The MGCP gateway is responding to a DLCX message from Cisco Unified Communications Manager
during a call setup.

Answer: E

QUESTION 5
To which SIP response class do the SIP response codes 300 to 399 belong?

A.    Provisional
B.    Client Failure
C.    Server Failure
D.    Successful
E.    Redirection

Answer: E
Explanation:
Redirection — further action needs to be taken in order to complete the request.
That is what this class implies.

QUESTION 6
Which SIP request method enables reliability of SIP 1xx response types?

A.    ACK
B.    PRACK
C.    OPTIONS
D.    CANCEL
E.    REGISTER

Answer: B
Explanation:
In order to achieve reliability for provisional responses, we do nearly the same thing.
Reliable provisional responses are retransmitted by the TU with an exponential backoff.
Those retransmissions cease when a PRACK message is received.
The PRACK request plays the same role as ACK, but for provisional responses.
There is an important difference, however. PRACK is a normal SIP message, like BYE.

QUESTION 7
Which SIP response is considered a final response?

A.    183 Session in Progress
B.    199 Early Dialog Terminated
C.    200 OK
D.    180 Ringing
E.    100 Trying

Answer: C
Explanation:
Indicates the request was successful.Wheather other options state the request is still in progress or request is intiated.

QUESTION 8
Which two SDP content headers can be found in a SIP INVITE message? (Choose two.)

A.    Expires
B.    Contact
C.    Connection Info
D.    Media Attributes
E.    Allow
F.    CSeq

Answer: CD
Explanation:
Connection info is optional field in sdp wheather Media attributes decide the codec and media type for that call.

QUESTION 9
Refer to the exhibit. If this SIP call is initiated using early offer, which SIP message will UA#2 use to communicate its media capability to UA#1?

A.    INVITE
B.    180 Ringing
C.    200 OK
D.    ACK
E.    RTP Media

Answer: C
Explanation:
In Early offer, SIP Send SDP in the invite , the other node will send the SDP in the 200 message.

QUESTION 10
Refer to the exhibit. If this SIP call is initiated using delayed offer, which SIP message will UA#1 use to communicate its media capability to UA#2?

A.    INVITE
B.    180 Ringing
C.    200 OK
D.    ACK
E.    RTP Media

Answer: D
Explanation:
In the Delayed Offer process, the calling does not send its offer in the SIP INVITE Message. The callee sends the offer within the SDP fields of its answer (SIP 200 OK). The calling answers within the ACK message.


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QUESTION 396
Refer to the exhibit. An engineer is trying to provision CUCME with three 8841 phones.
However all phone fail to register.
Which two changes in the configuration would allow the phones to register? (Choose two)
 

A.    The registrar server command must be added under the voice register global configuration
B.    The IP address trusted authenticate command must be added under voice service voip
C.    The source-address command must be added under the voice register global configuration
D.    The local SIP proxy address must be configuration under the sip-ua configuration
E.    The registrar server command must be added under the sip section of voice service voip

Answer: CE

QUESTION 397
A collaboration engineer has been asked to implement secure real-time protocol between a Cisco Unified CM and SIP gateway. Which option is a consideration for this implementation?

A.    Only T.38 and Cisco fax protocol are supported
B.    SIP require the all the time be sent in GMT
C.    Call hold RE-INVITE is not supported
D.    SRTP is supported only in cisco IOS 15.x and higher

Answer: B

QUESTION 398
Refer to the exhibit. A collaboration engineer configures Cisco Unified CM location using G.711 and iLBC for each site. The bandwidth for each link is shown. Which two options represent the maximum concurrent number of calls supported from Grand Junction to Casper for each Codec?
(Choose two)
 

A.    20 G.711 calls
B.    18 G.711 calls
C.    36 iLBC calls
D.    42 iLBC calls
E.    11 G.711 calls
F.    51 iLBC calls

Answer: CE

QUESTION 399
A collaboration engineer is troubleshooting an MOH problem on a Cisco IOS SIP gateway. While searching through a debug ccsip message output, which three parameters in the SIP messages can be used to determine if the call was placed on hold? (Choose three)

A.    OPTIONS WITH 301 CALLHOLD
B.    INVITE WITH a=INACTIVE
C.    INVITE WITH a=SENDONLY
D.    OPTION WITH c=INACTIVE
E.    c=IN IP4 0.0.0.0
F.    BYE WITH A = CALLHOLD

Answer: BCE

QUESTION 400
Refer to the exhibit. A cisco collaboration engineer discovers that an instance of IOS media termination point (MTP) could not maintain stable registration with CUCM. Call manager traces is showing in the exhibit. What is the reason for the flapping registration?
 

A.    The CCM version on IOS configuration does not match the CUCM version.
B.    The IOS MTP is experiencing high CPU and is missing its keep-alive.
C.    A Firewall is blocking port 2000 intermittently between IOS Device and CUCM.
D.    Another IOS Media device is attempting to register with the same name.

Answer: D

QUESTION 401
A collaboration engineer is designing Cisco IM&P implementation to support instant messaging logging for compliance.
Which two external databases can be used to support that functionality? (Choose two.)

A.    Oracle database
B.    MySQL database
C.    Microsoft SQL database
D.    PostgreSQL database
E.    Informix SQL database

Answer: AD

QUESTION 402
Refer to the exhibit. A cisco collaboration engineer is troubleshooting a gateway and gatekeeper problem and sees this output from a debug command.
Which two configuration can cause this problem? (Choose two)
 

A.    The same zone prefix is configured in two different gatekeepers
B.    The same H323-ID is configured in two different gateways
C.    The same gw-type-prefix is configured in two different zone subnets IDs
D.    The same zone subnet ID is configured in two different gatekeepers
E.    The same E164-ID is configured in two different gateways

Answer: BE

QUESTION 403
The Cisco Unified Border Element is configured using high availability with the Hot Standby Routing Protocol. Which two pieces of information can be gathered about the calls traversing these border elements? (Choose two.)
 

A.    The total number of calls is 150.
B.    The number of non-native calls is 70.
C.    The number of native calls is 50.
D.    The number of calls preserved is 220.
E.    The total number of active calls is 100.

Answer: AB

QUESTION 404
Refer to the exhibit. Which two SIP packet handing behaviour will result with this cisco Unified Border Element (CUBE) configuration? (Choose two)
 

A.    Unsupported content/MIME pass-through
B.    SIP Refer is not support when received on this CUBE
C.    Privacy headers received on SIP message will be replaced with NON-privacy headers on this CUBE
D.    P-Preferred identities
E.    Mid-call codec changes

Answer: AE

QUESTION 405
A CUCM engineer has deployed Type B SIP Phones on a remote site and no SIP dial rules were deployed for these phones. How Will CUCM receive the DTMF after the phone goes off- hook and the button are pressed?

A.    Each digit will be received by CUCM in a SIP NOTIFY message as soon as they are pressed
B.    The first digit will be received in a sip invite and subsequent digits will be received using NOTIFY message as soon as they are pressed.
C.    Each digit bill be received by CUCM in a SIP INVITE as soon as the dial soft key has been pressed.
D.    All digits will be received by CUCM in a SIP INVITE as soon as the dial soft key has been pressed

Answer: A

QUESTION 406
The Video engineer wants to enable the LATM codec to allow video endpoint to communicate over audio With other IP devices Which two Characteristic should the voice engineer be aware of before enabling LATM on the Cisco Unified border element router? (Choose two)

A.    Dual tone Multi-frequency interworking with LATM codec is not supported
B.    Codec transcoding between LATM and other codecs is not supported
C.    SIP UPDATE message outlined in RFC3311 is not supported
D.    Box-to-Box High availability support feature is not supported
E.    Configure LATM under a voice class or dial peer is not supported
F.    Basic calls using flow-around or flow-through is not supported

Answer: AB


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