A lot of people learn a lot of knowledge in Alnaba and focus on us to improve their skills. The designing Cisco Wireless Enterprise Networks examination requires more effort and attention to detail:
(Duration 120 minutes (90 – 110 questions)), (Available Languages English), Exam Policies.

Getting this information is important, so if you want to easily pass the 400-051 CCIE Collaboration exam Please select: https://www.pass4itsure.com/400-051.html (q&as:705 PDF + VCE) is frequently updated and reviewed,
To ensure that you successfully obtain the certificate. To help more friends Alnaba share 56 Cisco 400-051 exam dumps and 400-051 PDF downloads online for free.

[PDF] Free Cisco CCIE 400-051 dumps download from Google Drive: https://drive.google.com/open?id=1ysYqNxdbWl7JQG1U4MhvmiBJ8oQCG7Jh

[PDF] Free Full Cisco dumps download from Google Drive: https://drive.google.com/drive/folders/1dq6fv9FX6zSDDoHc3ge-WzVU9SNaUbsH

400-051 CCIE Collaboration – Cisco: https://www.cisco.com/c/en/us/training-events/training-certifications/exams/current-list/ccie-collaboration.html

Free 56 Cisco CCIE 400-051 Practice test questions and answers

QUESTION 1
Refer to the exhibit.

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.

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.

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

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.

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

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.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.

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

QUESTION 21
Refer to the exhibit.Which out-of-dialog SIP OPTIONS ping response put dial-peer tag 1111 into its current operational state?
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
Explanation/Reference:
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.

QUESTION 22
Refer to the exhibit.The exhibit shows the Cisco IOS CLI output of debug ipdhcp packet, which was captured on a router that is located at a branch office where a single IP phone is located. There is a standalone Cisco Unified Communications Manager server
at the central site, which also provides DHCP services to the IP phone at the branch office. You are troubleshooting a problem where the IP phone received an IP address in the correct subnet and with a correct subnet mask from the DHCP
server, but never completed registration with Cisco Unified CM. Assuming the IP phone is correctly defined on Unified CM, which two statements the network components are true? (Choose two.)
A. The MAC address of the IP phone is 01ec44761e3e7d.
B. The IP address of the DHCP server is 10.101.15.1.
C. The MAC address of the VLAN 101 interface is 01ec44761e3e7d.
D. The MAC address of the IP phone is ec44761e3e7d.
E. There is no IP connectivity between the VLAN 101 interface of the branch router and the ip-helper address that is configured on this interface.
F. Based on the information provided, we cannot conclude if there is IP connectivity between the IP phone and Cisco Unified CM.Correct Answer: DF
Explanation
Explanation/Reference:
In the logs the only information that we get is about the mac address of the IP phone because the IP phone is raising the boot request.

QUESTION 23
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
Correct Answer: C
Explanation
Explanation/Reference:
Ephone is configured as octo line so maximum call number is 8 and it will be divided between lines.

QUESTION 24
Refer to the exhibitYour customer sent you this debug output, captured on a Cisco IOS router (router A), to troubleshoot a problem where all H.323 calls that originate from another Cisco IOS router (router B) are being dropped almost immediately after arriving
at router A. What is the reason for these disconnected calls?
A. Calls were unsuccessful because of internal, memory-related problems on router A.
B. Calls were rejected because the called number was denied on a configured class of restriction list on router A.
C. Calls were rejected because the VoIP dial peer 1002 was not operational.
D. Calls were unsuccessful because the router B IP address was not found in the trusted source IP address list on router A.
E. Calls were rejected by router A because it received an admission reject from its gatekeeper because of toll fraud suspicion.
Correct Answer: D
Explanation
Explanation/Reference:
Trusted source IP address list on router is a list which secures the connectivity of router if it is enabled then we need to give the trusted entry for any route to reach.

QUESTION 25
Refer to the exhibit.

How many simultaneous inbound calls can be handled by these two IP phones?
A. 2
B. 4
C. 6
D. 9
E. 10
Correct Answer: A
Explanation
Explanation/Reference:
The line is configured as shared line so it will support maximum two calls at a time.

QUESTION 26
How many signaling bits are there in each T1 time slot using channel associated signaling with Super Frame?
A. 1
B. 2
C. 4
D. 8
E. 12
Correct Answer: B
ExplanationExplanation/Reference:
Each T1 CAS has 24 channels that can transmit 8 bits per channel each. This gives us a total of 192 bits. The T1 has one additional bit for framing, bringing the total to 193 bits. Two types of line coding can be used on a T1 CAS. The first
type of line coding is called Super Frame (SF). This is an older and less – efficient type of framing. Super Frame bundles 12 of these 193 – bit frames together for transport. It then uses the even ?numbered frames as signaling bits. The T1
CAS signaling then looks at every sixth frame for signaling information. This comes out to be 2 bits that are referred to as the A and B bits, which reside in frames 6 and 12.

QUESTION 27
Which two Cisco Unified Communications Manager Express hunt group mechanisms keep track of the number of hops in call delivery decisions? (Choose two.)
A. sequential
B. peer
C. longest idle
D. parallel
E. overlay
F. linear
Correct Answer: BC
Explanation
Explanation/Reference:
Peer configures hunting in a circular manner among the hunt group member DNs and starts with the DN to the right of the last DN to ring.
Longest-idle specify hunting on the DN which is idle for a longest period of time and the call will go to that DN of the hunt Group.
Reference: http://ccievoice.ksiazek.be/?p=690

QUESTION 28
In which call state does the Mobility soft key act as a toggle key to enable or disable Single Number Reach for Cisco Unified Communications Manager Express SCCP IP phones?
A. idle
B. seized
C. alerting
D. ringing
E. connected
Correct Answer: A
Explanation
Explanation/Reference:
Pressing the Mobility soft key during the idle call state enables the SNR feature. This key is a toggle; pressing it a second time disables SNR.
Reference:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/admin/configuration/guide/cm eadm/cmesnr.html

QUESTION 29
Which statement about what happens to a Cisco IOS SIP VoIP dial-peer that never received any responses to its out-of-dialog OPTIONS ping is true?
A. Its admin state will be up but operational state will be down.
B. Its admin and operational state will be down.
C. Its admin and operational state will remain up.
D. Its admin state will be up but operational state will be “busy-out”.
E. Its admin and operational state will be “busy-out”.
Correct Answer: A
Explanation
Explanation/Reference:
You can check the validity of your dial peer configuration by performing the following tasks:
If you have relatively few dial peers configured, you can use the show dial-peer voice command to verify that the configuration is correct. To display a specific dial peer or to display all configured dial peers, use this command. The following is
sample output from the show dial-peer voice command for a specific VoIP dial peer:
router# show dial-peer voice 10
VoiceOverIpPeer10
tag = 10, dest-pat = \Q’,
incall-number = \Q+14087′,
group = 0, Admin state is up, Operation state is down Permission is Answer,
type = voip, session-target = \Q’,
sess-proto = cisco, req-qos = bestEffort,
acc-qos = bestEffort,
fax-rate = voice, codec = g729r8,
Expect factor = 10,Icpif = 30, VAD = disabled, Poor QOV Trap = disabled, Connect Time = 0, Charged Units = 0
Successful Calls = 0, Failed Calls = 0
Accepted Calls = 0, Refused Calls = 0
Last Disconnect Cause is “”
Last Disconnect Text is “”
Last Setup Time = 0
To show the dial peer that matches a particular number (destination pattern), use the show dialplan number command. The following example displays the VoIP dial peer associated with the destination pattern 51234:
router# show dialplan number 51234
Macro Exp.: 14085551234
VoiceOverIpPeer1004
tag = 1004, destination-pattern = \Q+1408555….’, answer-address = \Q’,
group = 1004, Admin state is up, Operation state is up type = voip, session-target = \Qipv4:1.13.24.0′, ip precedence: 0 UDP checksum = disabled
session-protocol = cisco, req-qos = best-effort, acc-qos = best-effort,
fax-rate = voice, codec = g729r8,
Expect factor = 10, Icpif = 30,
VAD = enabled, Poor QOV Trap = disabled
Connect Time = 0, Charged Units = 0
Successful Calls = 0, Failed Calls = 0
Accepted Calls = 0, Refused Calls = 0
Last Disconnect Cause is “”
Last Disconnect Text is “”
Last Setup Time = 0
Matched: +14085551234 Digits: 7
Target: ipv4:172.13.24.0

QUESTION 30
Which two statements are requirements regarding hunt group options for B-ACD implementation on Cisco Unified Communications Manager Express routers? (Choose two.)
A. The ephone hunt group is mandatory.
B. Either the ephone hunt group or the voice hunt group is acceptable.
C. Hunt group members must be SCCP IP phones.
D. Hunt group members can include both SCCP or SIP IP phones.
E. Hunt group members must be SIP IP phones.
F. The member hunting mechanism must be set to sequential.
Correct Answer: AC
Explanation
Explanation/Reference:
The ephone hunt group is mandatory, and while ephone hunt groups only support Cisco Unified SCCP IP phones, a voice hunt group supports either a Cisco Unified SCCP IP phone or a Cisco Unified SIP IP phone.
Reference:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/command/reference/cme_v1ht.ht ml

QUESTION 31
Refer to the exhibit.

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 with 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

QUESTION 32
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?
A. MULTIPLE_FRAME_ESTABLISHED
B. TEI_ASSIGNED
C. AWAITING_ESTABLISHMENT
D. STANDBY
E. INITIALIZED
Correct Answer: B
Explanation
Explanation/Reference:
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 33
Refer to the exhibit.

IP phone 1 has the MAC address 11111.1111.1111, while IP phone 2 has the MAC address 2222.2222.2222. The first two incoming calls were answered by IP phone 1, while the third incoming call was answered by IP phone 2. What will
happen to the fourth incoming call?
A. Both phones will ring, but only IP phone 2 can answer the call.
B. Both phones will ring and either phone can answer the call.
C. Only IP phone 2 will ring and can answer the call.
D. Neither phone will ring and the call will be forwarded to 2100.
E. Neither phone will ring and the call will be forwarded to 2200.
Correct Answer: B
Explanation
Explanation/Reference:
In shared line configuration phone share the same line so it is possible for any phone to answer the call.

QUESTION 34
Refer to the exhibit.

How many calls, inbound and outbound combined, are supported on the IP phone?
A. 1
B. 2
C. 8
D. 12
E. 50
Correct Answer: E
Explanation
Explanation/Reference:
Output incomplete to figure out the answer

QUESTION 35
In Cisco IOS routers, which chipset is the PVDM-12 DSP hardware based on?
A. C542
B. C549
C. C5510
D. C5421
E. C5409
Correct Answer: B
Explanation
Explanation/Reference:
NM-HDV has five SIMM sockets (called Banks) that hold the PVDM-12 cards. Each PVDM- 12 card contains three TI 549 DSPs.
Reference:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/7x/uc7_0/media.html

QUESTION 36
Which enrollment method does a Cisco IOS VPN router trustpoint use to install a Certificate Authority Proxy Function certificate for LSC validation of a Cisco IP phone client?
A. HTTP proxy server
B. certificate authority server URL
C. terminal
D. self-signed
E. registration authority
Correct Answer: C
Explanation
Explanation/Reference:
Router(config)#crypto pki trustpoint CAPF
enrollment terminal
authorization username subjectname commonname
revocation-check none
Router(config)#crypto pki authenticate CAPF
Router(config)#
Things to Note:
The enrollment method is terminal because the certificate has to be manually installed on the Router.
Reference: http://www.cisco.com/c/en/us/support/docs/ios-nx-os-software/authentication- authorization-accounting-aaa/116313-configure-anyconnect-00.html

QUESTION 37
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
Explanation/Reference:
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 conference 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 38
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
Explanation/Reference:
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 39
Refer to the exhibit.The exhibit shows the Cisco IOS CLI output of debug ipdhcp packet, which was captured on a router that is located at a branch office where a single IP phone is located. There is a standalone Cisco Unified Communications Manager serverat the central site, which also provides DHCP services to the IP phone at the branch office. You are troubleshooting a problem where the IP phone could not register to Cisco Unified Communications Manager. You have confirmed that the IP
phone received an IP address in the correct subnet and with a correct subnet mask from the DHCP server. Assuming the IP phone is correctly defined on Unified CM, which two statements about the network components are true? (Choose
two.)
A. The MAC address of the IP phone is 01ec44761e3e.
B. The IP address of the DHCP server is 10.101.15.1.
C. The MAC address of the VLAN 101 interface is ec44761e3e7d.
D. The IP address of the VLAN 101 interface is 10.101.15.1.
E. There is IP connectivity between the VLAN 101 interface of the branch router and the ip- helper address that is configured on this interface.
F. There is IP connectivity between the IP phone and the ip-helper address on the VLAN 101 interface.
Correct Answer: DE
Explanation
Explanation/Reference:
As we can see from the logs given first line relate that dhcp request is being relayed. So it clarifies there must be ip helper address commend given by the admin on interface vlan 101. Now we can see from the second line that giaddress is
set as source address of vlan 101 by the dhcp as 10.101.15.1 to unicast the dhcp request

QUESTION 40
Assume the IP address of Cisco Unity Express is 10.1.1.1. Which URL provides Cisco Unity Express end users with a GUI interface to access and manage their messages and mailbox settings?
A. http://10.1.1.1/Web/Common/Login.do
B. http://10.1.1.1/ciscopca
C. http://10.1.1.1/user
D. http://10.1.1.1/inbox
E. http://10.1.1.1/
Correct Answer: C
Explanation
Explanation/Reference:
For user access cisco unity has predefined url and it is http://10.1.1.1/user

QUESTION 41
Refer to the exhibit.

Which ds0-group option should you select to support automated number identification information collection on inbound calls for this digital T1 voice circuit?
A. e&m-wink-start
B. e&m-delay-dial
C. e&m-delay-dial
D. e&m-lmr
E. e&m-fgd
Correct Answer: E
Explanation
Explanation/Reference:
Because it can receive ANI information and sends DNIS info. But can’t send ANI.
Reference:
http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/isdn/configuration/15-mt/vi-15-mt- book/vi-imp-t1cas-voip.html

QUESTION 42
Which two analog telephony signaling methods are most vulnerable to glare conditions? (Choose two.)
A. FXS Loop-start
B. FXO Ground-start
C. E&M Wink-start
D. E&M Delay-dial
E. E&M Immediate-start
F. E&M Feature Group D
Correct Answer: AE
Explanation
Explanation/Reference:
The loop start signaling method is more common and is typically used by residential phone lines. When a voice port is configured with loop start signaling, the device (telephone) closes the circuit loop that signals the CO voice port to provide
dial tone; an incoming call is signaled on the CO by supplying a predefined voltage on the line. The loop start signaling method has one main disadvantage in that it has no method of preventing both sides of the connection from attempting to
seize the line at the same time; this condition is referred to as glare. Because of this, loop start signaling is typically not used on high demand circuits.
With immediate-start, the calling side of the connection seizes the line by going off hook on the E-lead and address information is sent using dual-tone multifrequency (DTMF) digits. Immediate start signaling is vulnerable to glare just like loop-
start signaling.

QUESTION 43
Refer to the exhibit.

Assume the B-ACD configuration on a Cisco IOS Cisco Unified Communications Manager Express router is operational. What will happen to a call in queue that was not answered by any member of the hunt group after the maximum amount
of time allowed in the call queue expires?
A. The call will be forwarded to extension 2120.
B. The call will be forwarded to extension 2220.
C. The call will be forwarded to extension 2003.
D. The call will be disconnected with user busy.
E. The call will be forwarded to 2100.
Correct Answer: B
Explanation
Explanation/Reference:
As we can see in the configuration 2220 is configured as voice mail forwarding extension so the call will forward to voice mail.

QUESTION 44
Which two statements about the restrictions for support of H.239 are true? (Choose two.)
A. SIP to H323 video calls using H.239 are not supported.
B. Redundancy for H.323 calls is not supported.
C. H.239 calls are not supported over intercluster trunks with Cisco Unified Communications Manager.
D. H.239 is not supported with third-party endpoints.
E. Cisco Unified Communications Manager supports a maximum of three video channels when using H.239.
Correct Answer: AB
Explanation
Explanation/Reference:
Restriction for Support for H.239
The Support for H.239 feature has the following restrictions:
Interworking SIP-H.323 Video calls using H.239 is not supported.
Redundancy for H.323 calls is not supported.
A fast-start request cannot include a request to open an H.239 additional video channel as it is not supported.
H.239 systems based on H.235 is not supported.
The SBC does not support call transfer for H.323 calls. When an H.323 endpoint is placed on hold, it closes its media as well as video channels.
Reference:
http://www.cisco.com/c/en/us/td/docs/routers/asr1000/configuration/guide/sbcu/2_xe/sbcu_ 2_xe_book/sbc_h239.html

QUESTION 45
Which method allows administrators to determine the best match impedance on analog voice ports in Cisco IOS router without having to shut and no shut the ports?
A. THL tone sweep
B. original tone sweep
C. ECAN test
D. inject-tone local sweep
E. remote loop
Correct Answer: A
Explanation
Explanation/Reference:
THL tone sweep allows all available impedances for a single test call to a quiet termination point out to the PSTN. You do not need to manually disable ECAN on the voice port under test. The test feature switches impedances automatically
for the tester. The test feature calculates the arithmetic mean ERL and reports the mean for each channel profile at each impedance setting. Then, at the end of the test, the feature specifies the best match impedance setting. This test
requires minimal supervision.
Reference: http://www.cisco.com/c/en/us/support/docs/voice/ip-telephony-voice-over-ip- voip/64282-impedance-choice.html

QUESTION 46
Which digital modulation method is used to transmit caller ID information on analog FXS ports on Cisco IOS routers?
A. DTMF
B. PSK
C. FSK
D. MF
E. pulse dialing
Correct Answer: C
Explanation
Explanation/Reference:
Link:-http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/vcr1/vcr1-cr-book/vcr-c4.html

QUESTION 47
Refer to the exhibit.

Which ds0-group option should you select to send automated number identification information on outbound calls for this digital T1 voice circuit?
A. e&m-fgd
B. e&m-fgd
C. fgd-eana
D. e&m-delay-dial
E. fgd-os
Correct Answer: C
Explanation
Explanation/Reference:
E&M signaling is often the preferred option for CAS because it avoids glare, it provides answer/disconnect supervision and it can receive Automatic Number Identification (ANI) with FGD and send ANI with FGD-EANA. In other words, you can
have 1 channel-group for incoming calls and 1 channel-group for outgoing calls.

QUESTION 48
In Channel Associated Signaling on a T1 circuit using Extended Super Frame, how many signaling bits does each T1 timeslot have?
A. 1
B. 2
C. 4
D. 12
E. 24
Correct Answer: C
Explanation
Explanation/Reference:
Each T1 channel carries a sequence of frames. These frames consist of 192 bits and an additional bit designated as the framing bit, for a total of 193 bits per frame. Super Frame (SF) groups twelve of these 193 bit frames together and
designates the framing bits of the even numbered frames as signaling bits. CAS looks specifically at every sixth frame for the timeslot’s or channel’s associated signaling information. These bits are commonly referred to as A- and B-bits.
Extended super frame (ESF), due to grouping the frames in sets of twenty-four, has four signaling bits per channel or timeslot. These occur in frames 6, 12, 18, and 24 and are called the A-, B-, C-, and D-bits respectively.
Reference: http://www.cisco.com/c/en/us/support/docs/voice/digital-cas/22444-t1-cas- ios.html

QUESTION 49
Refer to the exhibit.

How many inbound calls can be handled simultaneously between ephone 1 and ephone 2 before a user busy tone is returned?
A. 6
B. 7
C. 8
D. 9
E. 11
Correct Answer: A
Explanation
Explanation/Reference:
Because hunt stop channel is set to 6 as it enables call hunting to up to six channels of this ephone-dn and remaining 2 channels are available for outgoing call features.
Reference:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/command/reference/cme_cr/c me_e1ht.html

QUESTION 50
Refer to the exhibit.The exhibit shows an outgoing SIP 401 response message from Cisco Unified Communications Manager to a SIP VoIP service provider gateway. Which action can the Cisco Unified Communications Manager systems administrator use to
change the response to “200 OK”?
A. Make sure the gateway IP address of the SIP VoIP service provider is defined correctly in Cisco Unified Communications Manager SIP trunk.
B. Enable OPTIONS ping on Cisco Unified Communications Manager SIP trunk.
C. Disable OPTIONS ping on Cisco Unified Communications Manager SIP trunk.
D. Create an SIP response alias to force outgoing 401 messages to “200 OK”.
E. Disable digest authentication on Cisco Unified Communications Manager SIP trunk.
Correct Answer: E
Explanation
Explanation/Reference:
Because Right now CUCM challenges the identity of a SIP user agent and must configure digest credentials for the application user in CUCM or you have to disable it for stop challenging by CUCM.
Reference:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/security/9_0_1/secugd/CUCM_ BK_CCB00C40_00_cucm-security-guide-90/CUCM_BK_CCB00C40_00_cucm-security- guide_chapter_011010.html

QUESTION 51
In Cisco IOS routers, which chipset is the PVDM2-32 DSP hardware based on?
A. C5441
B. C549
C. C5510
D. C5421
E. Broadcom 1500
Correct Answer: C
Explanation
Explanation/Reference:
Explanation:
Table 6-2 DSP Resources on Cisco IOS Hardware Platforms with C5510 Chipset Hardware Module or Chassis
DSP Configuration
Maximum Number of Voice Terminations (Calls) per DSP and per Module
Medium Complexity
(8 calls per DSP)
High Complexity
(6 calls per DSP)
Flex Mode1
(240 MIPS per DSP)
VG-224
Fixed at 4 DSPs
N/A
24 calls per platform
Supported codecs:
•
G.711 (a-law, mu-law)
•
G.729a
N/A
NM-HD-1V2
Fixed at 1 DSP
4 calls per NM
4 calls per NM
240 MIPS per NM
NM-HD-2V
Fixed at 1 DSP
8 calls per NM
6 calls per NM
240 MIPS per NM
NM-HD-2VE
Fixed at 3 DSPs
24 calls per NM
18 calls per NM
720 MIPS per NM
NM-HDV2
NM-HDV2-2T1/E1
NM-HDV2-1T1/E1
1 to 4 of:
PVDM2-83 (½ DSP)PVDM2-16 (1 DSP)PVDM2-32 (2 DSPs)PVDM2-48 (3 DSPs)PVDM2-64 (4 DSPs)
Calls per PVDM:
48162432
Calls per PVDM:
36121824
MIPS per PVDM:
120240480720960
Reference: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/4x/42media.html

QUESTION 52
Which Cisco packet voice and video digital signal processor 3 can be used for video mixing on a Cisco Integrated Router Generation 2?
A. PVDM3-16
B. PVDM3-32
C. PVDM3-64
D. PVDM3-128
Correct Answer: D
Explanation
Explanation/Reference:
All the PVDM3 types (that is, PVDM3-16, PVDM3-32, PVDM3-64, PVDM3-128, PVDM3- 192, and PVDM3-256) support switched-only video conferences. Only PVDM3-128 and higher modules support video conferencing with video mixing,
transcoding and transrating.
http://www.cisco.com/c/en/us/products/collateral/unified-communications/voice-video- conferencing-isr-routers/data_sheet_c78-649427.pdf

QUESTION 53
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?
A. 5 seconds
B. 10 seconds
C. 20 seconds
D. 30 seconds
E. 40 seconds
Correct Answer: B
Explanation
Explanation/Reference:
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 54
Refer to the exhibit.

This output was captured on a Cisco IOS gateway shortly after it became the active Cisco Unified Border Element in a box-to-box redundancy failover.How many calls are native to this Cisco Unified Border Element?
A. 9
B. 12
C. 19
D. 31
E. 40
Correct Answer: D
Explanation
Explanation/Reference:
Total no of calls =28+12=40.
So, native calls are =40-9=31.
Reference: http://www.cisco.com/c/en/us/support/docs/voice-unified- communications/unified-border-element/112095-cube-hsrp-config-00.html.

QUESTION 55
Which two responses from a SIP device, which is the only remote destination on a Cisco Unified Communications Manager SIP trunk with OPTIONS ping enabled, cause the trunk to be marked as “Out of Service”? (Choose two.)
A. 503 Service Unavailable
B. 408 Request Timeout
C. 505 Version Not Supported
D. 504 Server Timeout
E. 484 Address Incomplete
F. 404 Not Found
Correct Answer: AB
Explanation
Explanation/Reference:
The remote peer may be marked as Out of Service if it fails to respond to OPTIONS, if it sends 503 or 408 responses, or if the Transport Control Protocol (TCP) connection cannot be established. If at least one IP address is available, the
trunk is In Service; if all IP addresses are unavailable, the trunk is Out of Service.
Reference:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/8_5_1/ccmcfg/bccm- 851-cm/b06siprf.html

QUESTION 56
Refer to the exhibit.

Assume the B-ACD configuration on a Cisco IOS Cisco Unified Communications Manager Express router is operational. What will happen to a new call that enters the call queue when there are already two calls in queue?
A. The call will be forwarded to extension 2120.
B. The call will be forwarded to extension 2220.
C. The call will be forwarded to extension 2003.
D. The call will be disconnected with user busy.
E. The call will be forwarded to 2100.
Correct Answer: C
Explanation
Explanation/Reference:
That is because queue over flow is forwarded to 2003 and maximum number of calls in queue is configured as two.

We offer more ways to make it easier for everyone to learn, and YouTube is the best tool in the video.
Follow channels: https://www.youtube.com/channel/UCTP5RClZrtMxtRkSvIag0DQ get more useful exam content.

Latest 56 Cisco CCIE 400-051 YouTube videos:

Conclusion:
Share 56 Cisco CCIE 400-051 exam dumps and 400-051 pdf for free to help you improve your skills, please follow us and share them with your friends. If you want to pass the exam easily, Alnaba recommends that https://www.pass4itsure.com/400-051.html (q&as:705 PDF + VCE) be updated throughout the year to ensure that it is valid at all times.

[PDF] Free Cisco CCIE 400-051 dumps download from Google Drive: https://drive.google.com/open?id=1ysYqNxdbWl7JQG1U4MhvmiBJ8oQCG7Jh

[PDF] Free Full Cisco dumps download from Google Drive: https://drive.google.com/drive/folders/1dq6fv9FX6zSDDoHc3ge-WzVU9SNaUbsH

Pass4itsure Promo Code 15% Off

related: https://www.maeeonline.org/most-important-microsoft-70-462-dumps/