400-051 PDF

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Question 1
Which Cisco Unified CM Application user is created by default and used by Cisco Unified CM Extension Mobility?
A. CCMAdministrator
B. EMSysUser
C. TabSyncSysUser D. CCMSysUser
E. CTIGWUser Answer D
Explanation:
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 Question 2
Refer to Exhibit:
 Assume there are no classes of service restrictions and all numbers shown are reachable from this Cisco Unified IP /065 Phone. Which statement about the dialing key strokes that allow the owner of this phone to reach directory number 2000 is true?
A. Press the last button on the right hand side of the phone screen.
B. There is no way to speed dial to directory number 2000 because the speed dial entry is not assigned. C. Press / on the phone keypad, followed by the Dial soft key.
D. Press 6 on the phone keypad, followed by the Dial soft key.
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E. Press 5 on the phone keypad, followed by the AbbrDial soft key.
Answer E
Explanation:
Configure these settings for the speed-dial numbers that you access with abbreviated dialing. When the user configures up to 00 speed-dial entries, part of the speed-dial entries can get assigned to the speed- dial buttons on the IP phone; the remaining speed-dial entries get used for abbreviated dialing. When a user starts dialing digits, the AbbrDial softkey displays on the phone, and the user can access any speed- dial entry by entering the appropriate index (code) for abbreviated dialing. Configuration for Dialing 5 on the phone keypad, followed by the AbbrDial softkey.
and it showing on the phone page as
Implemented and tested
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Question 3
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 engineer be aware of before enabling LATM on the Cisco Unified border element router? (Choose two)
A. Dual tone Multfrequency interworking with LATM codec is not supported. B. Codec transcoding between LATM and other codecs is not supported.
C. SIP UPDATE massage 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 A, B
Explanation:
 Question 4
Which two statements about virtual SNR in Cisco Unified Communications Manager Express are true? (Choose two.)
A. The SNR DN must be configured as SCCP.
B. Calls cannot be pulled back from the phone associated with the DN.
C. Ephone hunt groups are supported.
D. The virtual SNR DN must be assigned to an ephone.
E. Music on hold is supported for trunk and line side calls.
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Answer AB
Explanation:
SCCP: Configuring a Virtual SNR DN
To configure a virtual SNR DN on Cisco Unified SCCP IP phones, perform the following steps.
Prerequisites
Cisco Unified CME 0.0 or a later version.
Restrictions
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.
Question 5
Exhibit:
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 A user has reported that when trying to access Visual Voicemail the following error is received "Unable to open application. Please try again later. If it continues to fail contact your administrator". The collaboration engineer is working on the problem found on the following phone logs:
6532 NOT 13:40:35.35/4t0 CVM-InstallerModule.STATUSeINSTALLeCANCELLED & STATUSeINSTALLeERROR: [thread=installer MQThread][class=cip.midp.midletsuite.InstallerModule] [function=updateStatus] Midlet Install Canceled:Error...Visual VoiceMail
How can this issue be resolved?
A. Replace the server name with the series IP on service URL fled
B. Eliminate the space in the service Name fled
C. Configure DNS on phone configuration so it can resolve series name D. Check the Enable checkbox on IP phone service configuration
Answer B
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Explanation:
Looks like a simple error in phone service’s display name: Visual Voicemail. It needs to be exactly Visual Voicemail without spaces (delete the space in the service Name fled).
Question 6
Refer to the exhibit.
 A CUBE Cluster is working in HSRP box-to-box failover model. When the phone A calls Cisco WebEx meeting series to start a conference session, no DTMF tones are recognized. Which configuration change will fix this problem when configured on both CUBEs?
A. Voice-class sip asymmetric payload dtmf in dial-peer configuration B. Dtmf-relay rtp-nte digitdrop in the dial-peer configuration
C. Media flow-around under voice service VoIP configuration
D. Modem relay nse payload-type101 underglobal sip configuration E. Asymmetric payload full configured under global sip configuration Answer E
Explanation:
Symmetric and Asymmetric Calls
Cisco UBE supports dynamic payload type negotiation and interworking for all symmetric and asymmetric payload type combinations. A call leg on Cisco UBE is considered as symmetric or asymmetric based on the payload type value exchanged during the offer and answer with the endpoint:
• A symmetric endpoint accepts and sends the same payload type.
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• An asymmetric endpoint can accept and send different payload types.
The Dynamic Payload Type Interworking for DTMF and Codec Packets for SIP-to-SIP Calls feature is enabled by default for a symmetric call. An offer is sent with a payload type based on the dial-peer configuration. The answer is sent with the same payload type as was received in the incoming offer. When the payload type values negotiated during the signaling are different, the Cisco UBE changes the Real-Time Transport Protocol (RTP) payload value in the VoIP to RTP media path.
To support asymmetric call legs, you must enable The Dynamic Payload Type Interworking for DTMF and Codec Packets for SIP-to-SIP Calls feature. The dynamic payload type value is passed across the call legs, and the RTP payload type interworking is not required. The RTP payload type handling is dependent on the endpoint receiving them.
Configuring global SIP asymmetric payload support.
Example:
Router (conf-seri-sip)l asymmetric payload full
The dtmf and dynamic-codecs keywords are internally mapped to the full keyword to provide asymmetric payload type support for audio and video codecs, DTMF, and NSEs.
Question 7
Refer to the exhibit.
A collaboration engineer is configuring dynamic call routing and DN learning between two Cisco UCM and Two Cisco UCM express. What two configuration tasks will support this? (Choose two)
A. CME B should be configure as service advertisement forwarder only
B. CME B should be configure as service advertisement client and forwarder
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C. Router A and CME B should be configure to use the same autonomous system number D. Router A and CME B should be configure to use the same autonomous system number E. Router B and CME B should be configure to use the same autonomous system number F. CME B should be configured as service advertisement client only.
Answer B, C
Question 8
A cisco Unified CM user is set up with one remote destination profile that has two remote destination numbers.
First destination number is the user's mobile phone in country A and the Second is a mobile phone located in country
B. All outbound calls are centralized from the gateway at country
A. The user reports that inbound calls are properly routed to the mobile phone as long as the user is in country A. but inbound calls are not successfully routed to country B?
What could resolve this issue? (Choose two)
A. The enable mobile connect option must be selected under the user's second remote destination number
B. The value of remote destination limits should be change to 2 instead of the default value of 4 under the end user page
C. The enable mobile voice access option must be selected under the end user page
D. The value of maximum wait time for desk pickup should be change 20000 instead of the default of 10000, under the end user page
E. The rerouting calling search space assigned to the user's remote destination profile must have access to international calls
Answer A, E
Question 9
A Cisco Unity Connection administrator receives a request from a user who wants the ability to change the caller input option 0 in their voicemail box as needed without calling for support. How does the administrator grant these rights to the user?
A. The administrator can set the caller input to "Transfer to alternate contact number" so the user can log into their voicemail account through the TUI and set their alternate contact number.
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B. The administrator can set the caller input to "Transfer to alternate contact number" so the user can log into their voicemail account through their Cisco PCA page and set their alternate contact number.
C. The administrator can create a new call handler of which the user is an owner. The user controls the destination of that call handler by logging into the call handler via greetings administrator.
D. The administrator informs the user that this feature is a built-in option to the user Cisco PCA page under caller input.
E. The administrator informs the user that this feature is a built-in option for the user in the TUI under personal settings.
Answer A
Question 10
In Cisco Unity Connection, which three configuration dialog boxes can a user assign a search space? (Choose three.)
A. Routing Rule
B. Call Handler
C. Interview Handler D. Contacts
E. Users
F. Port
G. Phone System Answer A, B, E
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