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Cisco 400-051 Dumps Available Here at:
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051 dumps
Question 1
Refer to the exhibit.
The MGCP debugs were captured on a Cisco IOS MGCP PRI gateway registered to a Cisco United CM.
Assume that this
gateway had no active calls and will not take any new calls for the next 3 minutes. What the it will send the
next
NTFY message to the Cisco United CM?
Options:
A. Jan 10 05:56:35.130
B. Jan 10 05:55:45.130
C. Jan 10 05:55:50.130
D. Jan 10 05:56:05.130
E. Jan 10 05:55:40.130
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Cisco 400-051
Answer: C
Question 2
DRAG DROP
An engineer is set ng up a proxy TFTP between multiple Cisco communication Manager clusters.
Drag the step from the left to the correct order on the right to properly configure the certificates for the proxy
TFTP.
Not all options will be used.
Select and Place:
Options:
A.
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Cisco 400-051
Answer: A
Question 3
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
mapping
registration?
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Cisco 400-051
Options:
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-aliie.
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 4
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?
Options:
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
C. Each digit bill be received by CUCM in a SIP INVITE as soon as the dial of key has been
pressed.
D. All digits will be received by CUCM in a SIP INVITE as soon as the dial of key has been
pressed
Answer: A
Explanation:
Type-B IP telephones over functionality based on the Key Press Markup Language (KPML) to report user
key presses.
Each one of the user input events will generate its own KPML-based message to United CM. From the
standpoint of
relaying each user action immediately to United CM, this mode of operation is very similar to that of phones
running
SCCP.
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Cisco 400-051
Every user key press triggers a SIP NOTIFY message to United CM to report a KPML event corresponding
to the key
pressed by the user. This messaging enables United CM's digit analysis to recognize partial patterns as
they are
composed by the user and to provide the appropriate feedback, such as immediate reorder tone if an
invalid number
is being dialled.
In contrast to Type-A IP phones running SIP without dial rules, Type-B SIP phones have no Dial key to
indicate the end
of user input. A user dealing 1000 would be provided call progress indication (either ringback tone or
reorder tone)
after dealing the last 0 and without having to press the Dial key. This behaviour is consistent with the user
interface on
phones running the SCCP protocol.
Question 5
Which two Cisco United Communicatons Manager Express hunt group mechanisms keep track of the
number of
hops in call delivery decisions? (Choose two.)
Options:
A. sequential
B. peer
D. parallel
E. overlay
F. linear
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Cisco 400-051
Answer: B, C
Question 6
Which three issues present a customer from seeing the presence status of a new contact in their Jabber
contact list? (Choose three.)
Options:
Answer: B, C, D
Explanation:
No Presence Information After Login
Problem
You receive no Presence information after login.
Solution
Complete these steps in order to resolve this issue:
Make sure that the DNS series the PC is pointed to can resolve the fully qualified name of the CUPS
series.
The host entry will not suits, you must resolve via DNS.
Check the SUBSCRIBE CSS on the SIP trunk to CUP.
This CSS must include the part tons of the devices you are trying to receive status on.
The CUP Sips proxy incoming access control list (ACL) is not allowing incoming SIP presence messages to
reach the
presence engine. As a test, set the incoming ACL to ALL and reset the SIP proxy and presence engine.
Log in again to
the CUPC and try to reconfigure the incoming ACL properly.
Question 7
Refer to the exhibit.
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Cisco 400-051
An engineer is trying to provision CUCME with three tt41 phones. However all phone fail to register. Which
two
changes in the configuration would allow the phones to register? (Choose two)
Options:
A. The registrar series 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
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Cisco 400-051
E. The registrar series command must be added under the sip section of voice service VoIP
Answer: C, E
Question 8
A collaboration engineer has been asked to implement secure real-time protocol between a Cisco United
CM and SIP
gateway. Which option is a consideration for this implementation?
Options:
Answer: B
Explanation:
As necessary, configure the router to use Greenwich Mean Time (GMT). SIP requires that all times be sent
in GMT. SIP
INVITE messages are sent in GMT. However, the default for routers is to use Coordinated Universal Time
(UTC). To
configure the router to use GMT, issue the clock timezone command in global configuration mode and
specify GMT.
Question 9
Refer to the exhibit.
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Cisco 400-051
A collaboration engineer configures Cisco United CM location using G./11 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)
Options:
A. 20 G./11 calls
B. 1t G./11 calls
C. 36 iLBC calls
D. 42 iLBC calls
E. 11 G./11 calls
F. 51 iLBC calls
Answer: C, E
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Cisco 400-051
Question 10
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)
Options:
Answer: B, C, E
Explanation:
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Cisco 400-051
providing a send-reci over SDP, the media path remains in an inactive state and causes calls to drop call
will
drop),CUCM sends an ACK with send only to the 200 OK
.CUCM establishes new call leg with the intended transferred destination..Once this call is connected
4.CUCM receives a new Transfer instruction from the transferring phone to connect the held party
5. Next CUCM sends a re-invite with an inactive SDP to indicate a break in media path to MOH (in attempt
to
complete transfer)
6.Next CUCM sends an inactive SDP to indicate a break in media path between transferring party &
transferred party to
remote Referring party from call
/. Next CUCM sends a DO re-invite to connect the transferred party. The far end then sends 200 OK with
the required
SDP to connect the call
Question 11
Exhibit:
Options:
A. Homogeneous
B. guaranteed-audio
C. rendezvous
D. heterogeneous
E. guaranteed-iideo
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Answer: D
Explanation:
Heterogeneous, which means that multiple codecs and multiple bit rates are supported
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