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QUESTION 1
Refer to the exhibit
c1
Which option describes how this Cisco IOS SIP gateway, with an analog phone attached to its FXS port, handles an incoming informational SIP 180 response message without SDP?
A. It will enable early media cut-through.
B. It will generate local ring back.
C. It will do nothing because the message is informational.
D. It will terminate the call because this is an unsupported message format.
E. It will take the FXS port offhook.
Correct Answer: B
Explanation
Explanation/Reference:
The Session Initiation Protocol (SIP) feature allows you to specify whether 180 messages with Session Description Protocol (SDP) are handled in the same way as 183 responses with SDP. The 180 Ringing message is a provisional or
informational response used to indicate that the INVITE message has been received by the user agent and that alerting is taking place. The 183 Session Progress response indicates that information about the call state is present in the
message body media information. Both 180 and 183 messages may contain SDP, which allows an early media session to be established prior to the call being answered.
Prior to this feature, Cisco gateways handled a 180 Ringing response with SDP in the same manner as a 183 Session Progress response; that is, the SDP was assumed to be an indication that the far end would send early media. Cisco
gateways handled a 180 response without SDP by providing local ringback, rather than early media cut-through. This feature provides the capability to ignore the presence or absence of SDP in 180 messages, and as a result, treat all 180
messages in a uniform manner. The SIP–Enhanced 180 Provisional Response Handling feature allows you to specify which call treatment, early media or local ringback, is provided for 180 responses with SDP.
Reference:
http://www.cisco.com/c/en/us/td/docs/ios/voice/cube/configuration/guide/vb_book/vb_book/ vb_1506.html

QUESTION 2
Refer to the exhibit.
exam2pass 400-051 question
Which number is sent as the caller ID when a user at extension 5001 places a call that matches this translation profile?
A. 14087775001
B. +4087775001
C. 4087750001
D. +14087775001
Correct Answer: D
Explanation
Explanation/Reference:
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 3
Which Cisco IOS multipoint video conferencing profile is also known as best-effort video on the Cisco Integrated Router Generation 2 with packet voice and video digital signal processor 3?
A. homogeneous
B. guaranteed-audio
C. rendezvous
D. heterogeneous
E. flex mode video
Correct Answer: B
Explanation
Explanation/Reference:
Three types of video profiles are supported: homogeneous conferences (video switching), heterogeneous conferences (video mixing), and guaranteed audio conferences (best-effort video).
As the name suggests, when Guaranteed Audio Conferences is configured, the system attempts to display video for all participants; however, it does not guarantee that the video of all participants is displayed. For those participants whose
video is not displayed, participants are downgraded to audio-only and the profile guarantees preservation of the audio portion of the call. This option gives you added flexibility because the DSPs are not all reserved when the profile is created;
the system attempts to reserve them when this profile is activated with an actual conference. For example:
dspfarm profile 1 conference video guaranteed-audio codec h264 vga codec h264 4cif
Reference:
http://www.cisco.com/c/en/us/products/collateral/unified-communications/voice-video- conferencing-isr-routers/qa_c67-649850.html

QUESTION 4
Refer to the exhibit
exam2pass 400-051 question
IP phone 1 has MAC address of 1111.1111.1111, and IP phone 2 has MAC address of 2222.2222.2222. The first two incoming calls rang both phones and were answered by IP phone 2.
Which option describes what will happen to the third incoming call?
A. Both phones ring, but only IP phone 1 can answer the call.
B. Both phones ring and either phone can answer the call.
C. Only IP phone 1 rings and can answer the call.
D. Neither phone rings and the call is forwarded to 2100.
E. Neither phone rings and the call is forwarded to 2200.
Correct Answer: C
Explanation
Explanation/Reference:
As we can see busy-trigger-per-button set to 2 in voice register pool 1(IP Phone 1). So, IP Phone 1’s channel is free for receiving incoming calls and right now IP Phone 2 is busy answering call.

QUESTION 5
Which SIP message element is mapped to QSIG FACILITY messages being tunneled across a SIP trunk between two Cisco IOS gateways?
A. SIP UPDATE
B. SIP OPTIONS
C. SIP SUBSCRIBE
D. SIP INFO
E. SIP NOTIFY
Correct Answer: D
Explanation
Explanation/Reference:
Mapping of QSIG Message Elements to SIP Message Elements
This section lists QSIG message elements and their associated SIP message elements when QSIG messages are tunneled over a SIP trunk.

QSIG FACILITY/NOTIFY/INFO
<—>
SIP INFO

QSIG SETUP
<—>
SIP INVITE

QSIG ALERTING
<—>
SIP 180 RINGING

QSIG PROGRESS
<—>
SIP 183 PROGRESS

QSIG CONNECT
<—>
SIP 200 OK

<—>
SIP BYE/CANCEL/4xx—6xx Response
Reference:
http://www.cisco.com/c/en/us/td/docs/ios/voice/sip/configuration/guide/15_0/sip_15_0_book/tunneling_qsig.html

QUESTION 6
Which two types of line codes are configurable for an E1 PRI controller on a Cisco IOS router? (Choose two.)
A. CRC4
B. AMI
C. B8ZS
D. HDB3
E. ESF
F. SF
Correct Answer: BD
Explanation
Explanation/Reference:
Configuring an NM-xCE1T1-PRI Card for an E1 Interface Perform this task to select and configure an NM-xCE1T1-PRI network module card as E1.
SUMMARY STEPS
1. enable
2. configure terminal
3. card type e1 slot
4. controller e1 slot / port
5. linecode {ami | hdb3}
6. framing {crc4 | no-crc4}
Reference:
http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/interface/configuration/12-4/ir-12-4- book/ir-12-port-chann-nm.html

QUESTION 7
Which two statements describe characteristics of Cisco Unified Border Element high availability, prior to Cisco IOS release 15.2.3T, using a box-to-box redundancy configuration? (Choose two.)
A. It leverages HSRP for router redundancy and GLBP for load sharing between a pair of routers.
B. Cisco Unified Border Element session information is check-pointed across the active and standby router pair.
C. It supports media and signal preservation when a switchover occurs.
D. Only media streams are preserved when a switchover occurs.
E. It can leverage either HSRP or VRRP for router redundancy.
F. The SIP media signal must be bound to the loopback interface.
Correct Answer: BD
Explanation
Explanation/Reference:
Configure box-to-box redundancy when you:

Expect the behavior of the CSSs to be active/standby (only the master CSS processes flows)

Can configure a dedicated Fast Ethernet (FE) link between the CSSs for the VRRP heartbeat
Do not configure box-to-box redundancy when you:

Expect the behavior of the CSSs to be active-active (both CSSs processing flows). Use VIP redundancy instead.

Cannot configure a dedicated FE link between the CSSs.

Require the connection of a Layer 2 device between the redundant CSS peers

QUESTION 8
exam2pass 400-051 question
IP phone 1 has MAC address of 1111.1111.1111, and IP phone 2 has MAC address of 2222.2222.2222. The first two incoming calls were answered by IP phone 1, and the third incoming call was answered by IP phone 2.
Which option describes what will happen to the fourth incoming call?
A. Both phones ring, but only IP phone 2 can answer the call.
B. Both phones ring and either phone can answer the call.
C. Both phones ring, but only IP phone 1 can answer the call.
D. Neither phone rings and the call is forwarded to 2100.
E. Neither phone rings and the call is forwarded to 2200.
Correct Answer: D
Explanation
Explanation/Reference:
IP Phone 1 & 2 both have busy-trigger-per-button configured to 3 & 2 respectively. So, the 4th incoming call will get forwarded to 2100 as busy-triggers are exceeding in IP Phones.
Reference:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/command/reference/cme_cr/c me_c1ht.html#wp1570384096

QUESTION 9
Which statement describes the question mark wildcard character in a SIP trigger that is configured on Cisco Unity Express?
A. It matches any single digit in the range 0 through 9.
B. It matches one or more digits in the range 0 through 9.
C. It matches zero or more occurrences of the preceding digit or wildcard value.
D. It matches one or more occurrences of the preceding digit or wildcard value.
E. It matches any single digit in the range 0 through 9, when used within square brackets.
Correct Answer: C
Explanation
Explanation/Reference:
Table 5-2 Trigger Pattern Wildcards and Special Characters
Character
Description
Examples
X
The X wildcard matches any single digit in the range 0 through 9.
The trigger pattern 9XXX matches all numbers in the range 9000 through 9999.
!
The exclamation point (!) wildcard matches one or more digits in the range 0 through 9.
The trigger pattern 91! matches all numbers in the range 910 through
91999999999999999999999999999999.
?
The question mark (?) wildcard matches zero or more occurrences of the preceding digit or wildcard value.
The trigger pattern 91X? matches all numbers in the range 91 through
91999999999999999999999999999999.
+
The plus sign (+) wildcard matches one or more occurrences of the preceding digit or wildcard value.
The trigger pattern 91X+ matches all numbers in the range 910 through
91999999999999999999999999999999.
[ ]
The square bracket ([ ]) characters enclose a range of values.
The trigger pattern 813510[012345] matches all numbers in the range 8135100 through 8135105.

The hyphen (-) character, used with the square brackets, denotes a range of values.
The trigger pattern 813510[0-5] matches all numbers in the range 8135100 through 8135105.
^
The circumflex (^) character, used with the square brackets, negates a range of values.
Ensure that it is the first character following the opening bracket ([).
Each trigger pattern can have only one ^ character.
The trigger pattern 813510[^0-5] matches all numbers in the range 8135106

QUESTION 10
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
Explanation/Reference:
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 11
Refer to the exhibit.
exam2pass 400-051 question
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?
A. FXS
B. E&M
C. BRI
D. FXOE. DID
Correct Answer: D
Explanation
Explanation/Reference:
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.ht ml

QUESTION 12
Which call hunt mechanism is only supported by the voice hunt group in a Cisco Unified Communications Manager Express router?
A. sequential
B. peer
C. longest idle
D. parallel
E. overlay
Correct Answer: D
Explanation
Explanation/Reference:
Parallel Hunt-Group, allows a user to dial a pilot number that rings 2-10 different extensions simultaneously. The first extension to answer gets connected to the caller while all other extensions will stop ringing. A timeout value can be set
whereas if none of the extensions answer before the timer expires, all the extensions will stop ringing and one final destination number will ring indefinitely instead. The final number could be another voice hunt-group pilot number or mailbox.
The following features are supported for Voice Hunt-Group:
Calls can be forwarded to Voice Hunt-Group
Calls can be transferred to Voice Hunt-Group
Member of Voice Hunt-Group can be SCCP, ds0-group, pri-group, FXS or SIP phone/trunk
Max member of Voice Hunt-Group will be 32

QUESTION 13
Which Cisco Unified Communications Manager Express ephone button configuration separator enables overflow lines when the primary line for an overlay button is occupied by an active call?
A. o
B. c
C. w
D. x
E. :
Correct Answer: D
Explanation
Explanation/Reference:
x expansion/overflow, define additional expansion lines that are used when the primary line for an overlay button is occupied by an active call.

QUESTION 14
Which codec is supported on the Cisco PVDM2 DSP modules but not on the PVDM3 DSP modules?
A. G.728
B. G.729B
C. G.729AB
D. G.723
E. G.726
Correct Answer: D
Explanation
Explanation/Reference:
All codecs that are supported on the PVDM2 are supported on the PVDM3, except that the PVDM3 does not support the G.723 (G.723.1 and G.723.1A) codecs. The PVDM2 can be used to provide G.723 codec support or the G.729 codec
can be as an alternative on the PVDM3
Reference:
http://www.cisco.com/c/en/us/td/docs/routers/access/1900/software/configuration/guide/Sof tware_Configuration/pvdm3_config.html

QUESTION 15
Which message is used by a Cisco IOS MGCP gateway to send periodic keepalives to its call agent?
A. CRCX
B. AUCX
C. NTFY
D. RQNT
E. 200 OK
Correct Answer: C
Explanation
Explanation/Reference:
The gateway maintains this connection by sending empty MGCP Notify (NTFY) keepalive messages to Cisco CallManager at 15-second intervals. If the active Cisco CallManager fails to acknowledge receipt of the keepalive message within
30 seconds, the gateway attempts to switch over to the next highest order Cisco CallManager server that is available.
If none of the Cisco CallManager servers respond, the gateway switches into fallback mode and reverts to its default H.323 session application for basic call control support of IP telephony activity in the network.

QUESTION 16
Refer to the exhibit
exam2pass 400-051 question
Which two statements about calls that match dial-peer voice 7 voip are true? (Choose two.)
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
Explanation/Reference:
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 17
Refer to the exhibit
exam2pass 400-051 question
Which two statements about the show command output are true? (Choose two.)
A. T1 0/2/1 terminates Q.921 signaling to a Cisco Unified Communications Manager server.
B. T1 0/0/0 terminates Q.921 signaling on the gateway.
C. T1 0/0/0 terminates SIP Signaling to a Cisco Unified Communications Manager server.
D. T1 0/0/0 terminates Q.931 signaling to a Cisco Unified Communications Manager server.
E. T1 0/2/1 terminates Q.931 signaling on the gateway.
Correct Answer: BD
Explanation
Explanation/Reference:
As you can see the T1 0/0/0:23 interface is active in layer 1,2(multi frame established) & 3,it means Q.931 signaling terminates at gateway and using backhauled technique q931 messages are going to CUCM server.
But in case of T1 0/2/1 port multi frames are not established in layer 2.So, it’s not configured properly & doesn’t backhauling q931 messages to CUCM

QUESTION 18
When multiple greetings are enabled on Cisco Unity Express, which greeting will take the highest precedence?
A. standard
B. meeting
C. busy
D. closed
E. internal
Correct Answer: B
Explanation
Explanation/Reference:
Meeting greeting has the highest priority because it is set by the user when he doesn’t want to take the call and notices the caller he is online.

QUESTION 19
Refer to the exhibit
exam2pass 400-051 question
Assume the B-ACD configuration on a Cisco Unified Communications Manager Express router is operational.
Which option describes what will happen to an incoming call that entered the call queue but all members of the hunt group are in Do Not Disturb status?
A. The call is forwarded to extension 2120.
B. The call is forwarded to extension 2220.
C. The call is forwarded to extension 2003.
D. The call is disconnected with user busy.
E. The call is forwarded to extension 2100.
Correct Answer: B
Explanation
Explanation/Reference:
Because all members of hunt group are unavailable or activate DnD and incoming queued call will forward to voicemail using the param voice-mail 2220 command.
Reference:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/bacd/configuration/guide/cme 40tcl/40bacd.html#wpmkr1105714

QUESTION 20
Which two Cisco IOS multipoint video conferencing profiles are supported on the Cisco Integrated Router Generation 2 with packet voice and video digital signal processor 3? (Choose two.)
A. homogeneous
B. rendezvous
C. guaranteed-audio
D. scheduled
E. guaranteed-video
F. ad-hoc
Correct Answer: AC
Explanation
Explanation/Reference:
Q. What video conferences are supported?
A. Three types of video profiles are supported: homogeneous conferences (video switching), heterogeneous conferences (video mixing), and guaranteed audio conferences (best-effort video).
Reference: http://www.cisco.com/c/en/us/products/collateral/unified-communications/voice- video-conferencing-isr-routers/qa_c67-649850.html

【Official recommendations】

400-051 CCIE Collaboration – Cisco: https://www.cisco.com/c/en/us/training-events/training-certifications/exams/current-list/ccie-collaboration.html
This exam validates that candidates have the skills to plan, design, implement, operate, and troubleshoot enterprise collaboration and
communication networks.

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