Download New Latest (May) Cisco 642-427 Actual Tests 71-80

Ensurepass

 

QUESTION 71  (Topic 1)

 

Refer to the exhibits.

 

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MOH has been configured to run from flash at the BR1 site. The HQ phones and MOH server are placed in the Default region through the Default device pool. The BR1 phones are placed in the BR1 region through the BR1 device pool. The region configuration between Default and BR1 only permits G.729 codec.

 

When an IP phone user at the HQ site places a BR1 caller on hold, the BR1 caller hears tone on hold. Which of the following can cause this issue?

 

A.

Multicast routing is not enabled on the BR1 router.

B.

The command ip pim separate-dense-mode is missing from interface VLAN 120 at the SRST router in BR1.

C.

The MOH server is unable to stream MOH using G.711 codec because of the regions configuration.

D.

The command route 10.1.120.1 must be added to the multicast moh 239.1.1.1 port 16384 command at the SRST router in BR1.

E.

The Max Hops is too small in the MOH configuration

 

Answer: B

Explanation: Explanation-The router runs IP Multicast routing and IP PIM sparse-dense mode on any physical interface that must participate in multicast (PIM is in either sparse or dense mode, but the interface can be configured to forward sparse mode, dense mode, or both).

Link-

http://www.cisco.com/en/US/technologies/tk436/tk428/technologies_white_paper0900aecd 80131281_ns465_Networking_Solutions_White_Paper.html

 

 

QUESTION 72  (Topic 1)

 

When the command utils dbreplication status is executed on the Cisco Unified Communications Manager CLI, which step should be taken next to check the database replication status?

 

A.

View the activelog file.

B.

Run the same command on all nodes of the cluster.

C.

Restart the Cisco CallManager service.

D.

The command utils dbreplication runtimetstate must be run on the publisher.

E.

The command utils dbreplication runtimestate must be run on the subscriber.

 

Answer: A

 

 

QUESTION 73  (Topic 1)

 

After a successful login using Cisco Extension Mobility, an IP phone performs a restart followed by a reset. What can cause this issue?

 

A.

The phone model to which the user logged in is a different model than the model that is configured in the user device profile.

B.

The locale that is configured on the phone is different than the locale that is configured in the user device profile.

C.

The security profile that is specified in the user device profile does not match the security profile for the phone where the user logged in.

D.

The user device profile and the phone that is used for the Cisco Extension Mobility log in do not use the same phone protocol.

 

Answer: B

 

 

QUESTION 74  (Topic 1)

 

Refer to the Exhibits.

 

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Refer to the exhibits Assume that all learned SAF routes are placed in the SAF_Pt partition. An IP phone CSS contains the following partitions in this order lnternal_Pt, 3AF_Pt When the IP phone places a call to 3001. What will occur?

 

A.

The call will succeed and will be placed via the SAF network SAF-learned routes always take precedence.

B.

The call will fail because it will be blocked by the route pattern.

C.

The call will be placed in a round-robin fashion between the SAF network and SIP_Trunk.

D.

The call will be placed in a round-robin fashion between the SAF network and SIP_Trunk. Every other call will fail.

 

Answer: B

Explanation: Explanation:

If partition is listed first in CSS, it has priority for equal qualified matches.If no single best match exist, the call-routing entry with the partition that is listed first in thecalling-device CSS is used.See CIPT II V II 5-61 and TVOICE V I 3-20

 

 

QUESTION 75  (Topic 1)

 

Which two troubleshooting tools would initially be the best to use when troubleshooting the PSTN gateway side of a call routing issue while using Cisco Unified Communications Manager? (Choose two)

 

A.

RTMT trace output

B.

Cisco IOS debug commands

C.

Dialed Number Analyzer output

D.

Cisco Unified Communications Manager alerts

E.

Cisco IOS show commands

 

Answer: BE

Explanation: Link-

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/service/5_0_1/ccmsrva/sartmt.htm l

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/dna/5_0_4/dnai.html

 

 

QUESTION 76  (Topic 1)

 

Refer to the SDI trace in the exhibit A PSTN call arrived at the MGCP gateway that is shown in the SDI trace. If the caller ID that is displayed on the IP phone is 087071 222 and the HQ_clng__pty_CSS contains the HQ_cing_pty_Pt partition, which exhibit shows the correct gateway digit manipulation”?

 

 

 

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A.

Exhibit A

B.

Exhibit B

C.

Exhibit C

D.

Exhibit D

 

Answer: D

Explanation: Explanation-Actual incoming number is 14-087071 222 but next to this information in trace we can see two digits are stripped which is international code hence D is valid answer.

 

 

QUESTION 77  (Topic 1)

 

Refer to the exhibits.

 

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Examining the SDI trace shows that extension 2001 was configured with both a device and a line CSS. A can is placed to 00014087071222. Which statement about the call is true?

 

A.

The call will be blocked because the blkJntLPt partition appears in the combined CSS.

B.

The call will work because the PSTN__Pt partition appears before the blk_intl_Pt partition.

C.

The call will work because the blk_intl_Pt partition appears last in the combined CSS.

D.

The call will not work because the blk_intl_Pt partition appears first in the combined CSS.

E.

It is not possible to tell because furthertrace analysis is required

 

Answer: C

Explanation: Link-

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a00800 94b53.shtml

 

 

QUESTION 78  (Topic 1)

 

When a Cisco Unified Border Element connects two VoIP streams using flow-around media, which of the following options describe the components of the call that flow around and the components that flow through the device?

 

A.

All security information flows through the Cisco Unified Border Element, and all call signaling and RTP flows around the device.

B.

Call signaling flows through and call media flows around the device.

C.

Call media flows through and call signaling flows around the device.

D.

The initial call-signaling traffic flows through the device to initiate the call and then all subsequent calls flow around the device.

 

Answer: B

Explanation: Configuring Media Flow-AroundThis feature adds media flow-around capability on the Cisco Unified Border Element by supporting the processing of call setup and teardown requests (VoIP call signaling) and for media streams (flow-through and flow- around). Media flow-around can be configured the global level or it must be configured on both incoming and outgoing dial peers. If configured only on either the incoming or outgoing dialpeer, the call will become a flow-through call.Media flow-around is a good choice to improve scalability and performance when network-topology hiding and bearer- level interworking features are not requiredWith the default configuration, the Cisco UBE receives media packets from the inbound call leg, terminates them, and then reoriginates the media stream on an outbound call leg. Media flow-around enables media packets to be passed directly between the endpoints, without the intervention of the Cisco UBE. The Cisco UBE continues to handle routing and billing functions.To specify media flow-around for voice class, all VoIP calls, or individual dial peers, perform the steps in this section.References:http://www.cisco.com/en/US/prod/collateral/voicesw/ps6790/gatecont/p s5640/prod_qas09186a00801da69b.htmlhttp://www.cisco.com/en/US/docs/ios/voice/cube/ configuration/guide/vb-gw-sipsip.html#wp1392896

 

 

QUESTION 79  (Topic 1)

 

Which of these reasons can cause intrasite calls within a Cisco Unified Communications Manager cluster to fail?

 

A.

The route partition that is configured in the CCD requesting service is not listed in the calling phone CSS

B.

The trunk CSS does not include the partition for the called directory number.

C.

The MGCP gateway is not registered

D.

The calling phone does not have the correct CSS configured

E.

The calling phone does not have the correct partition configured.

 

Answer: D

Explanation: Explanation-To make a successful call within CUCM cluster following condition should satisfy.

No CSS, No partitions are used for call routing, default call routing hence any phone can call any phone within same CUCM cluster.

 

 

If we want to configure call restriction then CSS and partitions are must, if we don’t configure required partition in CSS then call will not be successful.

 

Link-

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a00800 94b53.shtml

 

 

QUESTION 80  (T
opic 1)

 

Refer to the exhibit.

 

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When a Cisco IP Communicator phone roams from San Jose (SJ) to RTP, the Cisco IP Communicator physical location and the device mobility group change from SJ to RTP All route patterns are assigned a route list that points to the local route group All device pools are configured to use the local route group Which statement is true when the roaming phone places an AAR call?

 

A.

Since globalized call routing is not configured, then the SJ gateway will be used in this case

B.

The phone will use the AAR CSS that contains the SJ_PSTN partition. The call will egress at the SJ gateway

C.

The phone will use the AAR CSS that contains the RTP_PSTN partition. The call will egress at the SJ gateway

D.

The phone will use the AAR CSS that contains the SJ_PSTN partition. The call will egress at the RTP gateway.

E.

The phone will use the AAR CSS that contains the RTP_PSTN partition The call will egress at the RTP gateway

 

Answer: D

Explanation:

Cisco Unified Communications Manager Version 7.0 introduced the Local Route Group feature.When using local route groups, gateway selection is totally independent of the matched route pattern and referenced route list and routegroup. The use of the Local Route Group feature makes no changes regarding roaming-sensitive settings. The application of these settings always makes sense when roaming between sites. The settings have no influence to the gateway selection and the dial rules that a user must follow. However, the dial planrelated part of Device Mobility changes substantially withthe new dial plan concept, This concept allows a roaming user to follow the home dial rules for external calls but use the local gateway of the roaming site In this case, When the device mobility group is not the same for San Jose and RTP, the Device Mobility related settings are not applied. The phone device keeps its San Jose-specific configuration Despite the San Jose-specific configuration on the phone, the PSTN calls that originate from the roaming phone are routed via the local PSTN gateway (RTP GW) and are based on the route list and device pool local route group settings. The San Jose-specific dial plan is used. Also, AAR remains configured with the San Jose-specific configuration, but if the San Jose dial plan and San Jose AAR CSS permit and if the AAR group contains the prefix that can be applied in RTP, then AAR can work

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