Vendor: Cisco

Exam Code: 300-075
Exam Name: Implementing Cisco IP Telephony & Video,
Part 2 v1.0
Version: 16.081
Q1Which statement is true regarding the configuration of SAF Forwarder?
A. In a multisite dial plan, SAF Forwarders may exist in multiple autonomous systems.
B. The client label that is configured in Cisco Unified Communications Manager must match the
configuration on the SAF Forwarder router.
C. There should not be multiple nodes of Cisco Unified Communications Manager clusters acting
as SAF clients.
D. The destination IP address must match the loopback address of the SAF router.
Answer: A
Q2Which three devices support the SAF Call Control Discovery protocol? (Choose three.)
A. Cisco Unified Border Element
B. Cisco Unity Connection
C. Cisco IOS Gatekeeper
D. Cisco Catalyst Switch
E. Cisco IOS Gateway
F. Cisco Unified Communications Manager
Answer: AEF
Q3Which component of Cisco Unified Communications Manager is responsible for sending
keepalive messages to the Service Advertisement Framework forwarder?
A. Call Control Discovery requesting service
B. Hosted DNs service
C. Service Advertisement Framework client control
D. Cisco Unified Communications Manager database
E. Service Advertisement Framework-enabled trunk
F. gatekeeper
Answer: C
Q4What is an advantage of TEHO?
A. TEHO implemented with ISRs eliminates PSTN toll charges.
B. TEHO implemented with ISRs can reduce PSTN toll charges.
C. TEHO implemented with AAR reduces toll charges.
D. TEHO implemented with CFUR reroutes calls.
Answer: B
Q5Which two options are valid service parameter settings that are used to set up proper video
QoS behavior across the Cisco Unified Communications Manager infrastructure? (Choose two.)
A. DSCP for Video Calls when RSVP Fails

B. Default Intraregion Min Video Call Bit Rate (Includes Audio)
C. Default Interregion Max Video Call Bit Rate (Includes Audio)
D. DSCP for Video Signaling
E. DSCP for Video Signaling when RSVP Fails
Answer: AC
Q6Which sign is prefixed to the number in global call routing?
A. B. +
C. #
D. @
E. &
F. *
Answer: B
Q7Scenario:
There are two call control systems in this item. The Cisco UCM is controlling the DX650, the
Cisco Jabber for Windows Client, and the 7965 and 9971 Video IP Phones. The Cisco VCS and
TMS control the Cisco Telepresence Conductor, the Cisco TelePresence MCU, and the Cisco
Jabber TelePresence for Windows
DNS Server:

Device Pool:

Expressway:

ILS: .

.

Locations: MRA: Speed Dial: .

.

) .SIP TRUNK: What two tasks must be completed in order to support calls between the VCS controlled endpoints and the Cisco Unified CM endpoints? (Choose two.

The Cisco VCS is controlling the SX20. B. Configure a neighbor zone on the Cisco Unified CM to point to the Cisco VCS. D. E. C. Configure a SIP trunk on the Cisco VCS to point to the Cisco Unified CM. and the Cisco Jabber TelePresence for Windows DP: Locations: . Configure a neighbor zone on the Cisco VCS to point to the Cisco Unified CM. and the 7965 and 9971 Video IP Phone.A. Answer: BE Q8Scenario: There are two call control systems in this item. the Cisco TelePresence MCU. Media Resource Group List. The Cisco UCM is controlling the Cisco Jabber for Windows Client. Configure a SIP trunk on the Cisco Unified CM to point to the Cisco VCS.

CSS: Movie Failure: .

Movie Setting: Topology: .

.

Subzones: Links: .

Pipe: .

B.) A.Both of the Cisco Telepresence Video for Windows clients are able to log into the server but can't make any calls. Wrong username and/or password. B. The bandwith is incorrectly configured. Investigation shows no connectivity problems. The roaming-sensitive parameters of the current (that is. the roaming) device pool are applied. The phone configuration is not modified. the Device Mobility-related settings are also applied. which of the following reasons could be causing this failure? A. D. Answer: D Q9Which two options for a Device Mobility-enabled IP phone are true? (Choose two. The user-specific settings determine which location-specific settings are downloaded from the Cisco Unified Communications Manager device pool. If the DMGs are the same. After reviewing the exhibits. The TMSPE is not working. C. Answer: BD Q10The network administrator of Enterprise X receives reports that at peak hours. some calls between remote offices are not passing through. . D. C. Wrong SIP domain name.

CallsRingNoAnswer B. LocationOutOfResources D. SX20 System information: . the Cisco Telepresence MCU. OutOfResources C. CallsAttempted Answer: BC Q11Scenario: There are two call control systems in this item. and the Cisco Jabber Telepresence for Windows. The Cisco UCM is controlling the DX650. the Cisco Jabber for Windows Client. RequestsThrottled E.) A. The Cisco VCS is controlling the SX20. and the 9971 Video IP Phone.The network administrator wants to estimate the volume of calls being affected by this issue. Which two RTMT counters can give more information on this? (Choose two.

DX650 Configuration: .

.

MRGL: DP: .

Locations: AARG: .

CSS: Movi Failure: .

SAF Trunk. Answer: BD Q12Assume that the Cisco IOS SAF Forwarder is configured correctly. SAF Trunk. and SAF Forwarder E. User is not associated with the device. SAF Trunk. Wrong SIP domain configured. SAF Trunk. CSF Device is not registered. and CCD Advertising Service Answer: B Q13When a SIP trunk is added for Call Control Discovery. The Enable SAF check box should be selected. F. SAF Trunk.Movie Settings: What two issues could be causing the Cisco Jabber Video for TelePresence failure shown in the exhibit? (Choose two) A. which statement is true? A. Incorrect username and password. SAF Forwarder. C. D. SAF Security Profile. IP or DNS name resolution issue. E. SAF Forwarder. SAF Security Profile. B. B. . Which minimum configurations on Cisco Unified Communications Manager are needed for the SAF registration to take place? A. The SIP trunk is added by selecting SIP Trunk and SIP Protocol. The Trunk Service Type should be Call Control Discovery. SAF Forwarder. and CCD Advertising Service B. The SIP trunk is added by selecting SIP Trunk and SIP Protocol. and CCD Requesting Service C. CCD Requesting Service. IP Phone DN not associated with the user. SAF Security Profile. CCD Requesting Service. SAF Security Profile. and CCD Advertising Service D.

show eigrp service-family ipv4 neighbors B. Static routes are not supported. B.C. The H. show saf neighbors E.323 as the protocol to be used. C. show voice saf dndball D. show saf registration E. show voice saf dndb all D. The Enable SAF check box should be selected in the trunk configuration.323 trunk is added by selecting Inter-Cluster Trunk (Non-Gatekeeper Controlled) and Device Protocol Inter-Cluster Trunk. The destination IP address field is configured as `SAF' to indicate that this trunk is used for SAF. show eigrp service-family ipv4 clients B. show eigrp address-family ipv4 neighbors C. D.323 Trunk. B. which statement is true? A. as long as it is a dynamic routing protocol. . Non-SAF routers act as an IP cloud. The H.323 trunk is added by selecting Inter-Cluster Trunk (Non-Gatekeeper Controlled) and Device Protocol Inter-Cluster Trunk. including non-SAF routers. The SIP trunk is added by selecting Call Control Discovery Trunk and then selecting SIP as the protocol to be used. show ip saf registration Answer: A Q17Which statement about Service Advertisement Framework is true? A. The destination IP address field is configured as `SAF' to indicate that this trunk is used for SAF.323 trunk is added for Call Control Discovery.323 trunk is added by selecting Call Control Discovery Trunk and then selecting H. The Trunk Service Type should be Call Control Discovery. SAF is totally independent of the underlying routing protocol. C. D. and selecting Inter-Cluster Trunk as the Device Protocol. SAF requires that the EIGRP be configured only on SAF routers. show ip saf neighbors Answer: A Q16Which Cisco IOS command is used to verify that the Cisco Unified Communications Manager Express has registered with the SAF Forwarder? A. Answer: A Q15Which Cisco IOS command is used to verify that a SAF Forwarder that is registered with Cisco Unified Communications Manager has established neighbor relations with an adjacent SAF Forwarder? A. The H. SAF operates on any dynamic or static IP routing configuration. show eigrp address-family ipv4 clients C. D. SAF has no dependency on the underlying routing protocol.323 trunk is added by selecting H. SAF requires that the EIGRP be configured on all routers. Answer: B Q14When an H. The SIP trunk is added by selecting SIP Trunk and SIP Protocol. The H.

Cisco Unified Communications Manager Answer: AEF Q19Which type of search message appears in the Cisco TelePresence Video Communication Server search history page when it receives a H. OPTIONS Answer: C Q20Scenario: There are two call control systems in this item. SETUP C. Cisco IOS Gatekeeper D. Cisco Unified Border Element B. The Cisco VCS is controlling the SX20.) A. and the Cisco Jabber Telepresence for Windows. Cisco Unity Connection C.Answer: D Q18Which three devices support the SAF Call Control Discovery protocol? (Choose three.323 call from a RAS-enabled endpoint that originates from an external zone? A. The Cisco UCM is controlling the DX650. Cisco Catalyst Switch E. ARQ B. Cisco IOS Gateway F. SX20 System information: . the Cisco Jabber for Windows Client. and the 9971 Video IP Phone. INVITE E. the Cisco Telepresence MCU. LRQ D.

DX650 Configuration: .

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MRGL: DP: .

Locations: AARG: .

CSS: Movi Failure: .

D. The DX650's MAC address is incorrect in the Cisco UCM.132D.460.164 format is true? A. The "+" symbol represents the international country code. . Answer: E Q21Which two statements regarding you configuring a traversal server and traversal client relationship are true? (Choose two.18/19 protocol for H.18/19 protocol for SIP traversal calls. C. The "+" symbol matches the preceding element one or more times. VCS supports either the Assent or the H. Answer: BD Q22Which statement about the function of the "+" symbol in the E. E. The "+" symbol matches the preceding element zero or one time.) A.Movie Settings: A new DX650 IP phone with MAC address D0C7. The "+" symbol represents the international call prefix. a TCP/TLS connection is established from the traversal client to the traversal server for SIP signaling.460. E. Device Pool cannot be default.18/19 protocol for H. B. The DX650 is the incorrect calling search space. The DX650 Phones does not support SIP. A VCS Expressway located in the public network or DMZ acts as the firewall traversal client. B. The location Hub_None has not been activated. What is causing this failure? A.323 traversal calls. If the Assent protocol is configured. IP address is 172.18-32. VCS supports only the H. C.460. D. VCS supports either the Assent or the H. B. D.119 has been added to the Cisco Unified Communications Manager.323 traversal calls. but is not registering properly.8914. C.

D. Unassign it from a media resource group. show ccm-manager fallback-mgcp E. Remove the subunit. 4 E. show running-config B. Which command should you use for this purpose? A. C. 3 D. Remove the device pool. Delete the dependency records. B. phones . show gateway D.Answer: B Q23You need to verify if the Media Gateway Control Protocol gateway is enabled and active. B. Delete the common device configuration. E. Create a single traversal client zone in VCS with the Unified CM nodes listed as location peer addresses. In VCS set Unified Communications mode to Mobile and remote access and configure each Unified CM node. what is the maximum number of Service Advertisement Framework forwarders that you can assign to a specific node? A. D. 1 B. show fallback-mgcp C.) A. show fallback-mgcp ccm-manager Answer: D Q24Which task must you perform before deleting a transcoder? A. Create a neighbor zone in VCS with the Unified CM nodes listed as location peer addresses. 6 Answer: B Q26Which two actions ensure that the call load from Cisco TelePresence Video Communication Server to a Cisco Unified Communications Manager cluster is shared across Unified CM nodes? (Choose two. 2 C. Answer: B Q25In a node-specific Service Advertisement Framework forwarder deployment model. E. C. Create one neighbor zone in VCS for each Unified CM node. 5 F. Use the Reset option. Answer: AD Q27Where do you specify the device mobility group and physical location after they have been configured? A. F. Create a VCS DNS zone and configure one DNS SRV record per Unified CM node. show running-config gateway F.

B. Which CPL configuration accomplishes this goal? . interface Fastethernet0/1 mls qos cos 1 F.323 message credential checks are delegated. Cisco Unified Communications Manager group (required). The administrator wants to stop calls from outside the organization being routed through it. SRST reference (optional). Location. interface Fastethernet0/1 mls qos trust cos E. Physical location (optional). mls qos map dscp-cos 8 10 to 2 C. Region (required) . Media resource group list (optional). A secure neighbor zone has been configured between the VCS Expressway and the VCS Control. AAR group. mls qos D. you must configure the following items if you want to choose them for the device pool. device pool E. which statement is true? A. Device mobility group (optional). DMI C. Q28Which three commands are necessary to override the default CoS to DSCP mapping on interface Fastethernet0/1? (Choose three. device mobility CSS D. H.) A. SIP registration proxy mode is set to Off in the VCS Expressway. AAR calling search space. MRGL F. Calling search space for auto-registration (optional). Device mobility calling search space. Answer: D Q30A local gateway is registered to Cisco TelePresence Video Communication Server with a prefix of 7. D. SIP registration proxy mode is set to On in the VCS Expressway. Reverted call focus priority (optional). C. interface Fastethernet0/2 mls qos cos 1 Answer: ACD Q29If delegated credentials checking has been enabled and remote workers can register to the VCS Expressway. mls qos map cos-dscp 0 10 18 26 34 46 48 56 B. locale Answer: D Explanation: Before you configure a device pool.B. Date/time group (required).

A. . B.

the Cisco Jabber for Windows Client. E. D. and the 9971 Video IP Phone. and the Cisco Jabber Telepresence for Windows. The Cisco UCM is controlling the DX650.C. Answer: A Q31Scenario: There are two call control systems in this item. SX20 System information: . the Cisco Telepresence MCU. The Cisco VCS is controlling the SX20.

DX650 Configuration: .

.

MRGL: DP: .

Locations: AARG: .

CSS: Movi Failure: .

323 A.com's Cisco TelePresence Video Communication Server allows SIP and H. Location D. AAR Group E. Media Resource Group List B.323 endpoints that register with an H. Calling Search Space Answer: E Q32Widgets. Which local zone search rule configuration allows SIP registered endpoints to connect to H.Movie Settings: Which device configuration option will allow an administrator to assign a device to a specific rights for making calls to specific DNs? A.323 registrations. 164 number only? . Device Pool C.

Expressway-E internal FQDN D. and the Cisco Jabber TelePresence for Windows DP: . E. Answer: D Q33Which configuration does Cisco recommend for the peer address on the Expressway-C secure traversal zone when the Expressway-E has one NIC enabled? A. C. The Cisco UCM is controlling the Cisco Jabber for Windows Client. the Cisco TelePresence MCU.B. Expressway-E external IP address C. Expressway-E external FQDN Answer: D Q34Scenario: There are two call control systems in this item. The Cisco VCS is controlling the SX20. D. Expressway-E internal IP address B. and the 7965 and 9971 Video IP Phone.

Locations: CSS: .

Movie Failure: Movie Setting: Topology: .

.

Subzones: Links: .

Pipe: .

B. Incorrect username and password. CSF Device is not registered.What two issues could be causing the Cisco Jabber Video for TelePresence failure shown in the exhibit? (Choose two) A. the Cisco TelePresence MCU. IP or DNS name resolution issue. The Cisco VCS is controlling the SX20. Answer: BD Q35Scenario: There are two call control systems in this item. The Cisco UCM is controlling the Cisco Jabber for Windows Client. and the 7965 and 9971 Video IP Phone. D. C. E. and the Cisco Jabber TelePresence for Windows DP: . Wrong SIP domain configured. IP Phone DN not associated with the user. F. User is not associated with the device.

Locations: CSS: .

Movie Failure: Movie Setting: Topology: .

.

Subzones: Links: .

Pipe: .

C. Device Pool. D. and the 7965 and 9971 Video IP Phones. Not enough bandwidth has been allocated. the Cisco Jabber for Windows Client. The Cisco VCS and TMS control the Cisco Telepresence Conductor. the Cisco TelePresence MCU. Answer: A Q36Scenario: There are two call control systems in this item. After reviewing the exhibits. Location. and the Cisco Jabber TelePresence for Windows DNS Server: . The pipe is not functioning. which of the following reasons could be causing this failure? A. The Cisco UCM is controlling the DX650. B.A third collaboration call fails between the backbone site and the HQ site.

Device Pool: .

Expressway: .

ILS: .

.

Locations:

MRA:

Speed Dial:

SIP TRUNK:

The intercluster URI call routing no longer allows calls between sites.
What is the reason why this would happen?

Configure a route pattern. Configure an SIP trunk to the ISR.A. Configure Cisco Unified Communications Manager AAR. Answer: C Q41Cisco Unified Communications Manager is configured with CAC for a maximum of 10 voice calls. Accept Replaces Header E. a route list. Enable Application Level Authorization C. IP or DNS name resolution issue. C. B. and route groups to a trunk and a gateway in Cisco Unified Communications Manager. B. Answer: C Q37Which two options should be selected in the SIP trunk security profile that affect the SIP trunk pointing to the VCS? (Choose two. User is not associated with the device. minimize PSTN costs B. adding the Expressway servers to the Application Servers list Answer: AC Q39What is the purpose of the local route group? A. Configure AAR in Cisco Unified Communications Manager. Accept Unsolicited Notification B. configuring a device pool with video feature enabled C. .) A. Wrong SIP domain configured. C. Configure a route pattern to a gateway in Cisco Unified Communications Manager. allowing numeric dialing from Cisco phones to Expressway B. eliminate the need for a route list D. help in the selection of the PSTN egress gateway C. Accept Presence Subscription Answer: AD Q38Which two options are configuration steps on Cisco Unified Communications Manager that are used when integrating with VCS Expressway servers? (Choose two. Configure CFUR in Cisco Unified Communications Manager.) A. B. allowing dialing to Expressway domain from Cisco phones D. creating an application user on Cisco Unified Communications Manager with assigned privileges E. allow manipulation of digits at the cost point to egress Answer: B Q40Which action configures PSTN backup for calls that are rejected by the gatekeeper CAC? A. No SIP route patterns for cisco.lab exist. Accept Out-of-Dialog REFER D. D. Which action routes the 11th call through the PSTN? A. D.

The gateway falls back to the H. The MGCP calls are queued up until the Cisco Unified Communications Manager servers are online. All MGCP call processing is interrupted until the Cisco Unified Communications Manager servers are online. D. The gateway continues to make an attempt to connect to the backup Cisco Unified Communications Manager server. ip source-address Answer: ABE . D. call-manager-fallback B. call-manager-fallback secondary dialtone 0 C. dial-peer voice 1 pots secondary dialtone 0 D. SRST without MGCP fallback C. ccm-manager secondary dialtone 0 Answer: B Q44Which action is performed by the Media Gateway Control Protocol gateway with SRST configured.C. Answer: B Q45Which three commands are mandatory to implement SRST for five Cisco IP Phones? (Choose three. SRST with VoIP dial peers to Cisco Unified Communications Manager Express Answer: C Q43Which option configures the secondary dial tone option for SRST mode to let the users hear the dial tone for PSTN calls? A.) A. Configure Cisco Unified Communications Manager RSVP-enabled locations. Configure Cisco Unified Communications Manager locations. C. The gateway continues with the MGCP call processing without any interruption. B. E.323 protocol for further call processing. SRST with MGCP fallback B. max-ephones C. voice service voip secondary dialtone 0 B. The gateway waits for the primary Cisco Unified Communications Manager server to come alive. keepalive D. Cisco Unified Communications Manager Express in SRST mode D. Answer: B Q42Which solution is needed to enable presence and extension mobility to branch office phones during a WAN failure? A. when it loses connectivity with the primary and backup Cisco Unified Communications Manager servers? A. limit-dn E. F.

D. The traversal client zone on Expressway-C Media encryption mode must be set to Auto. The traversal server zone on Expressway-C must have a TLS verify subject name configured. On the VCS.Q46Which parameter should be set to prevent H. and set Auto Registration to off. navigate to Configuration. The Cisco UCM is controlling the DX650.323. navigate to Configuration. D. Protocols. C. On the VCS. H. H. B. and set Auto Registration to off. The traversal client zone and the traversal server zone Media encryption mode must be set to Auto. Registration. The traversal client zone and the traversal server zone Media encryption mode must be set to Force encrypted. Answer: A Q47Which two statements about configuring mobile and remote access on Cisco TelePresence Video Communication Server Expressway are true? (Choose two. The traversal client zone and the traversal server zone must be set to SIP TLS with TLS verify mode set to On. Registration. navigate to Configuration. and the 9971 Video IP Phone. The Cisco VCS and TMS controll the the Cisco TelePresence MCU. and set Auto Registration to off.) A. On the VCS. Answer: BE Q48Scenario: There are two call control systems in this item.323 endpoints from registering to Cisco TelePresence Video Communication Server automatically? A. C. Configuration. and the Cisco Jabber TelePresence for Windows DP: . and set Auto Discover to off. E. Allow List. B. On the VCS. Protocols.323. navigate to Configuration. the Cisco Jabber for Windows Client.

Locations: CSS: .

SRST: .

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SRST-BR2-Config: BR2 Config: .

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729 codec over the WAN link to headquarters? A. Media Resource Group List B. Location D. .SESTPSTNCall: Which device configuration option will allow an administrator to control bandwidth between calls placed between branches? A. Device Pool C. B. Configure Cisco Unified Communications Manager regions. Regions Answer: C Q49A voicemail product that supports only the G. Which action allows branch Cisco IP phones to function with voicemail while using only the G.711 codec is installed in headquarters. AAR Group E. Configure transcoding within Cisco Unified Communications Manager.

The Cisco UCM is controlling the DX650. Answer: C Q50Scenario: There are two call control systems in this item. Configure transcoding resources in Cisco IOS and assign to the MRGL of Cisco IP phones. The Cisco VCS and TMS controll the the Cisco TelePresence MCU. D.C. and the Cisco Jabber TelePresence for Windows DP: Locations: . the Cisco Jabber for Windows Client. Configure transcoder resources in the branch Cisco IP phones. and the 9971 Video IP Phone.

CSS: SRST: .

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SRST-BR2-Config: BR2 Config: .

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CFUR eliminates the need for COR on an ISR. CFUR is designed to initiate TEHO to reduce toll charges. C. The SRST enabled router is not configured correctly. The Cisco UCM is pointing to the wrong IPv4 address of the BR router. C.SESTPSTNCall: After adding SRST functionality the SRST does not work. B. CFUR can prevent phones from unregistering. D. After reviewing the exhibits. Device Pool cannot be default. The router does not support SRST. which of the following reasons could be causing this failure? A. B. Answer: A Q51Which option is a benefit of implementing CFUR? A. CFUR can reroute calls placed to a temporarily unregistered destination phone. . D.

Media Answer: B Q55Which action configures phones in site A to use G. EF (46) C. MGCP C. C. mgcp active Answer: B Q54Which gateway does the Cisco Unified Communications Manager control all call activity? A. activate mgcp F. but uses G. CS4 (32) Answer: D Q53You want to perform Media Gateway Control Protocol gateway maintenance. mgcp enable D. Which command should you use? A. Endpoints can make calls to unknown IP addresses without the VCS querying any neighbors. SIP B.323 D. . AF41 (34) D. enable mgcp B. Which statement is true? A. B.729 to site C? A. mgcp yes E. CS3 (24) B. you disable Media Control Gateway Protocol gateway using the no mgcp command. Configure transcoder resources in Cisco Unified Communications Manager.Answer: C Q52What is the standard Layer 3 DSCP media packet value that should be set for Cisco TelePresence endpoints? A. H. Configure Cisco Unified Communications Manager locations. D. Configure a gatekeeper. For this purpose. mgcp C. Configure Cisco Unified Communications Manager regions. you want to enable the Media Control Gateway Protocol gateway.711 to site B. Answer: A Q56Refer to the exhibit. After you perform the maintenance.

SIP interworking modE.SIP interworking modE. On C. SAF forwarders on Cisco routers C. C. Off E.323 . Set QoS mode to DiffServ and tag value 34.B. it routes the call through the neighbor. isdn overlap-receiving E. or vice versa. which settings do you apply if traffic through the VCS should be tagged with DSCP AF41? A.225 trunk Answer: B Q60Which command is needed to utilize local dial peers on an MGCP-controlled ISR during an SRST failover? A. Set QoS mode to DiffServ and tag value 32. it queries its neighbors for the remote address and if permitted. .323 endpoints is made over SIP. Set QoS mode to IntServ and tag value to 34. Variable Answer: C Q58Which commands are needed to configure Cisco Unified Communications Manager Express in SRST mode? A. B. call-manager-fallback and voice-translation Answer: A Q59Which component is needed to set up SAF CCD? A. Dialing by IP address is not supported on VCS. H. H. such as where a call between two H. C.323 . SAF-enabled H. If the VCS receives a call to an unknown IP address.323 intercluster (gatekeeper controlled) trunk B. Reject B.323 .323 . ccm-manager fallback-mgcp B. telephony-service and moh C.SIP interworking modE. Registered only D. Endpoints that are registered directly to the VCS can call only an IP address of a system that is also registered directly to that VCS.323 . D.SIP interworking modE. telephony-service C.SIP interworking modE. telephony-service and srst mode B. H. voice-translation-rule Answer: A Q61When you configure QoS on VCS. SAF-enabled H. dialplan-pattern D. Which setting is recommended? A. call-manager-fallback and srst mode D. H. H. Cisco Unified Communications cluster D. Answer: A Q57You want to avoid unnecessary interworking in Cisco TelePresence Video Communication Server.

Set QoS mode to IntServ and tag value to 32. and the Cisco Jabber TelePresence for Windows DNS Server: Device Pool: . The Cisco UCM is controlling the DX650.D. Answer: C Q62Scenario: There are two call control systems in this item. The Cisco VCS and TMS control the Cisco Telepresence Conductor. the Cisco TelePresence MCU. Set QoS mode to ToS and tag value to 32. and the 7965 and 9971 Video IP Phones. E. the Cisco Jabber for Windows Client.

Expressway: .

ILS: .

.

Locations: MRA: Speed Dial: .

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SIP TRUNK: Which three configuration tasks need to be completed on the router to support the registration of Cisco Jabber clients? (Choose three.) .

C. Users find that video endpoints registered on Call manager can call each other and likewise for the endpoints registered on the VCS server. delete notification E. unlimited Answer: A Q66Company X has a Cisco Unified Communications Manager cluster and a VCS Control server with video endpoints registered on both systems.) A. Answer: B Q65How many active gatekeepers can you can define in a local zone? A. The administrator for Company X realizes he needs a SIP trunk between the two systems for any video endpoint to call any other video endpoint. create connection D. 2 C. Answer: BCE Q63With Media Gateway Control Protocol configuration on the voice gateway. 1 B. 15 F. E. which three types of messages are involved in the call flow between the call agent and the voice gateway? (Choose three. Set up a neighbor zone on the VCS server with the location of Cisco UCM using the menu option VCS Configuration > Zones > zone. Remove the device pool. end connections Answer: ACE Q64Which task must you perform before deleting a transcoder? A. C. The DNS server has the wrong IP address. C.A. 10 E.) A. restart in progress F. Delete the dependency records. Set up a SIP trunk on Cisco UCM with the option Device-Trunk with destination address of the VCS server. D. . B. Flush the DNS Cache on the client. The internal DNS Service (SRV) records need to be updated on the DNS Server. modify endpoint C. Set up a subzone on Cisco UCM with the peer address to the VCS cluster. F. The DNS AOR records are wrong. audit endpoint B. Use the Reset option. Add the appropriate DNS SRV for the Internet entries on the DNS Server. Delete the common device configuration. Which two steps must the administrator take to add the SIP trunk? (Choose two. Remove the subunit. D. B. E. Unassign it from a media resource group. B. 5 D.

Set up a traversal subzone on the VCS server to allow endpoints that are registered on Cisco UCM to communicate. E. and the Cisco Jabber TelePresence for Windows DP: Locations: . The Cisco VCS and TMS controll the the Cisco TelePresence MCU. Answer: AC Q67Scenario: There are two call control systems in this item. The Cisco UCM is controlling the DX650. the Cisco Jabber for Windows Client. and the 9971 Video IP Phone.D. Set up a SIP trunk on the VCS server with the destination address of the Cisco UCM and Transport set to TCP.

CSS: SRST: .

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SRST-BR2-Config: BR2 Config: .

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) A... direct-inward-dial port 0/0/0:13 C. the calls fail from the PSTN.. dial-peer voice 1 pots incoming called-number 228822.. Which two of the following configurations if applied to the router..S//+44&/ exit ! voice translation-profile pstn-in translate called 1 ! . would remedy this situation? (Choose two....SESTPSTNCall: After configuring the CFUR for the directory number that is applied to BR2 phone (+442288224001).. direct-inward-dial port 0/0/0:15 B. voice translation-rule 1 rule 1/rt228821. dial-peer voice 1 pots incoming called-number 228822.

Change the dial peer to dial-peer voice 901 voip. D. There should not be multiple nodes of Cisco Unified Communications Manager clusters acting as SAF clients. TEHO and MGCP fallback D. Change the dial peer to dial-peer voice 9011 pots. E911 calling . Answer: AD Q68Refer to the exhibit.voice-port 0/0/0:15 translation-profile incoming pstn-in D. Which configuration change is needed to enable NANP international dialing during MGCP fallback? A. on-net calling patterns B. SAF Forwarders may exist in multiple autonomous systems. Add the command prefix 9011 to the dial peer. C. Answer: C Q69Which statement is true regarding the configuration of SAF Forwarder? A.... Answer: A Q70Which technologies provide remote-site redundancy for Cisco IP Phones during a WAN failure? A. SRST and MGCP fallback B. The router does not need to be configured. B. C. B. SRST and AAR Answer: A Q71What are two important considerations when implementing TEHO to reduce long-distance cost? (Choose two. The destination IP address must match the loopback address of the SAF router. SRST and TEHO C.S//+44&/ exit ! voice translation-profile pstn-in translate called 1 ! voice-port 0/0/0:15 translation-profile incoming pstn-in E.) A. D. Add the command prefix 011 to the dial peer. The client label that is configured in Cisco Unified Communications Manager must match the configuration on the SAF Forwarder router. In a multisite dial plan. voice translation-rule 1 rule 1/rt228822.

D. D. Toll charges can be reduced when TEHO is implemented with MGCP fallback. D. 6 when using a local route group C. B. TEHO utilizes WAN links. number of route patterns D. but roaming-sensitive settings modify the AAR group. and device CSS. Device Mobility settings. AAR CSS. with Cisco Unified Communications Manager regions D. but illegal in others. on the other hand. B. C. Answer: A Q73How are Cisco IP Phones directly configured to utilize local route groups? A. TEHO is legal in most countries. 15 when not using a local route group B. Device Mobility settings have no impact on call routing. and AAR CSS. Local route groups add complexity to the dial plan. may have an impact on call routing because they modify the device CSS. TEHO reduces WAN utilization. B.) A.C. caller ID Answer: BD Q72Which statement about TEHO is true? A. Toll charges can be reduced when TEHO is implemented with CAC. The dial plan is simplified with local route groups. Device Mobility settings modify the device CSS and the roaming-sensitive settings modify the AAR group and AAR CSS. Roaming-sensitive settings are settings that do not have an impact on call routing. with Cisco Unified Communications Manager locations E. with Cisco Unified Communications Manager AAR Answer: A Q74When you configure TEHO for long-distance calls and use the local PSTN gateways as fallback. with Cisco Unified Communications Manager device pools B. TEHO is legal in all countries. C. C. 10 when not using a local route group Answer: B Q75Which two statements about international multisite dial plans are true? (Choose two. the roaming-sensitive settings modify the device CSS. Device Mobility settings modify the AAR group and the AAR CSS. AAR group. how many route patterns do you require for a cluster with five sites that are located in different area codes? A. 5 when using a local route group D. Answer: BD Q76What impact do roaming-sensitive settings and Device Mobility settings have on call routing? A. with Cisco Unified Communications Manager CSS and partitions C. .

users manage their FindMe accounts via VCS. The Advanced Networking or Dual Network Interfaces option key has been installed. mobile phones in the cluster that support device mobility D. B. The service parameter settings take precedence over the device mobility mode phone settings. 0-9. . Users are allowed to delete or change the address of their principal devices. mobile phones in the cluster that are in default mode Answer: A Q78Which statement is true when device mobility mode is enabled or disabled in the Phone Configuration window? A. The host portion cannot start or end with a hyphen. Answer: D Q81Which system configuration is used to set a restriction on audio bandwidth? A. D. The host portion is not case sensitive. Endpoints should register with an alias that is the same as an existing FindMe ID. physical location D.200. The host portion accepts characters a-z. and periods. The device mobility mode phone settings take precedence over the service parameter settings. to which does the cluster setting apply? A. The combined service parameter settings and the device mobility mode phone settings will be used. A-Z. B.165. The default settings will be used due to the conflicts. licensing Answer: B Q82Which two statements regarding IPv4 Static NAT address 209. all phones in the cluster that subscribed to device mobility C. C. location C. all phones in the cluster that support device mobility B.Answer: D Q77When device mobility mode is enabled or disabled for a cluster. A VCS cluster name must be configured. region B. The host portion can have two periods in a row. C.) A. D. B. C. D. If VCS is using Cisco TMS provisioning. hyphens. Answer: D Q80Which statement about setting up FindMe in Cisco TelePresence Video Communication Server is true? A. Answer: A Q79Which statement about the host portion format in Cisco Unified Communications Manager URI dialing is false? A.230 has been configured on a VCS Expressway are true? (Choose two.

15 F. This counter represents the total number of times that a call on a particular Cisco Unified Communications Manager through the location failed due to lack of bandwidth. Answer: B Q84How many active gatekeepers can you can define in a local zone? A. C.323 packets to 209. All called numbers sent out to the PSTN are in E. Answer: B .165. 10 E.230. B. With static NAT enabled on the LAN2 interface.323 payload messages.165.230 to outbound H. in Unified CM. 1 B. Answer: C Q86Which statement best describes globalized call routing in Cisco Unified Communications Manager? A. in Unified CM. VCS applies 209. All incoming calling numbers on the phones are displayed as an E. This counter represents the total number of failed video-stream requests (most likely due to lack of bandwidth) in the location where the person who initiated the video conference resides. Answer: AC Q83The network administrator has been investigating bandwidth issues between the central office and remote sites where location-based CAC is implemented. The CSS of all phones contain partitions assigned to route patterns that are in global format.200.164 with the + prefix format. C. VCS rewrites the Layer 3 source address of outbound SIP and H. This counter represents the total number of times since the last restart of the Cisco IP Voice Streaming Application that a Cisco Unified Communications Manager connection was lost.164 with the + prefix. D. E.164 with the + prefix format. where are DSCP values configured? A. All phone directory numbers are configured as an E. This counter represents the total number of times that a call through locations failed due to the lack of bandwidth.B. D. under Device > Device Settings > Device Defaults C.200. What does the Cisco Unified Communications Manager "LocationOutOfResources" counter indicate? A. Call routing is based on numbers represented as an E. under Enterprise Parameters Configuration B.323 and SIP payload traffic exiting the LAN1 interface. in Unified CM.164 with the + prefix. under Service Parameters > Cisco CallManager Service > Cluster-wide Parameters D.165.230 to outbound SIP and H.200. unlimited Answer: A Q85When video endpoints register with Cisco Unified Communications Manager. D. VCS applies 209. B. 5 D. 2 C. DSCP parameters are always configured on each individual video endpoint. C.

225 D.323 gatekeeper? A. user settings C.509 Subject Name from the VCS . SIP trunk D. the physical location and associated services have not changed. H.) A. A SIP trunk security profile must be configured with the X. which two configuration components are not changed? (Choose two. H. SIP trunk C. H.323 trunk D. Q91Which three statements about configuring an encrypted trunk between Cisco TelePresence Video Communication Server and Cisco Unified Communications Manager are true? (Choose three. IP subnet B.225 trunk (gatekeeper-controlled) B. H. A SIP trunk security profile must be configured with Device Security Mode set to TLS.323 gateway B. When an IP phone moves from one device mobility group to another. intracluster Answer: CD Q90Assume that local route groups are configured. Each site has a Cisco CallManager and an H. A SIP trunk security profile must be configured with Incoming Transport Type set to TCP+UDP.) A. region E. intercluster E. The Cisco Unified Communications Manager trunk configuration must have the destination port set to 5061. intercluster trunk (gatekeeper-controlled) Answer: D Q89Which two are gatekeeper-controlled trunk options that support gatekeeper call administration control? (Choose two. SRST reference D. Which connection method is best for these two new customers? A. H. H. The root CA of the VCS server certificate must be loaded in Cisco Unified Communications Manager. B.) A.Q87Which device is used to connect to the H.323 gateway that allows connection to the PSTN. phone button settings Answer: BE Explanation: Although the phone may have moved from one subnet to another.245 C. C.323 B. intercluster trunk (non-gatekeeper controlled) C. MGCP gateway Answer: C Q88The corporate WAN has been extended to two newly acquired sites and it includes gatekeeper support. D. E.

B. description C. The Cisco Unified Communications Manager zone configured in VCS must have SIP authentication trust mode set to On. Answer: ACE Q92Which system configuration is used to set audio codecs? A. The phones never re-register with the primary server until the active (secondary) one goes down. The secondary server keeps sending keepalive message to the primary server and when it succeeds. phone button template D. it sends out an "ALIVE" message via broadcast so that the phones re-register. location C. the phone re-registers with it. Answer: D Q94If the device pool in the phone record does not match the device pools in the matching subnet. NTP information Answer: BC Q96What is the default DSCP/PHB for video conferencing packets in Cisco Unified Communications Manager? A. what will the system consider the phone to be? A. The phone sends keepalive messages to the primary server frequently and when it succeeds. G. how does the phone recognize that the primary server is back? A. CS6/48 . D. physical location D. licensing Answer: A Q93During device failover to the secondary Cisco Unified Communications Manager server. When the primary server goes online.) A. roaming B. C. The Cisco Unified Communications Manager zone configured in VCS must have TLS verify mode set to Off. F. unregistered C. unknown D. region B. EF/46 B.certificate. new device Answer: A Q95Which two Cisco Extension Mobility attributes are available in the user device profile? (Choose two. regions B. it unregisters the phones to force them to register to the primary.

and dial peers of the Cisco Unified SRST router? A. Answer: C Q99Which command displays the detailed configuration of all the Cisco Unified IP phones. Expressway-C and Expressway-E must trust each other's server certificate. Create a set of username and password on each of the Expressway-C and Expressway-E to authenticate the neighboring peer. Speech Connect D. D. Enable username and password authentication verification on Expressway-E. on the actual Cisco phone itself because the DSCP setting is not part of its configuration file downloaded at registration E. C. E. A separate pair of traversal zones must be configured if an H. CS3/24 Answer: C Q97Which feature allows you to specify which endpoints ring when someone calls a user on a specific destination ID? A. Answer: AC Q101A video endpoint is configured with SIP only. show voice port summary Answer: A Q100Which two options should be used to create a secure traversal zone between the Expressway-C and Expressway-E? (Choose two.) A. the QoS service parameter in Cisco Unified Communications Manager D. One Cisco Unified Communications traversal zone for H. on the MGCP router at the edge of both networks B. voice ports. AF41/34 D. The administrator runs a wire trace while a video call is taking place and sees that the packets are not set to AF41 for desktop video as they should be. but only when they are registered to the VCS Control. FindME B.323 and SIP connections.C. the service parameters in the VCS Control C. The setting cannot be changed for video endpoints that are registered to Cisco Unified Communications Manager. show ephone summary D. Single Number Reach Answer: A Q98The administrator at Company X is getting user reports of inconsistent quality on video calls between endpoints registered to Cisco Unified Communications Manager. Where should the administrator look next to confirm that the correct DSCP markings are being set? A. B.323 connection is required and Interworking is disabled. show dial-peer voice summary C. show call-manager-fallback all B. What does a video endpoint use to register with the VCS Control? . Extension Mobility C.

MAC address D. which format must you use in the Search Rule? A. name@IP Address (192.com Answer: A Q106Company X currently uses a Cisco Unified Communications Manager. name@domain B.100. The administrator for Company X has deployed a Collaboration Expressway server at the edge of the network in an attempt to remove the need for VPN when doing voice. a VPN must be set up before their Jabber client can be used.0) C.example. DNS tracing C. name@hostname D. system name Answer: B Q102Which zone will the VCS Control use to route calls to the VCS Expressway? A.example. which has been configured for IP desk phones and Jabber soft phones. _cisco-uds. _cuplogin._tls. Dynamic DNS Answer: A Q105Which DNS SRV Records must be configured on the external DNS server in a mobile remote access scenario with Cisco Expressway? A.168. SIP URI C. A SIP trunk has to be set up between the Expressway-C and Cisco UCM. To do this._tcp. DNS zone C. _collab-edge.example. IP Address (192.) A.323 endpoints so that they communicate with one another. Which two additional steps are needed to complete this deployment? (Choose two. OPTIONS ping B. Users report however that whenever they are out of the office.A. which mechanism do you use to verify that the trunk has an active connection? A. neighbor zone B._tcp.100. traversal client zone D.com D._udp.com C. However. _collab-edge.com B. . Continuous ping D. devices outside cannot register.example.0) Answer: A Q104When you connect a Cisco VCS Control to Cisco Unified Communications Manager by using a SIP trunk. ENUM zone Answer: C Q103You want to configure Cisco VCS SIP endpoints and H. IP address B.168.

The firewall at Company X requires a rule to allow all traffic from the DMZ to pass to the same network that the VCS Control is on. A TMS server is needed to allow the firewall traversal to occur between the VCS Expressway and the VCS Control servers. An additional interface must be enabled on the Cisco UCM and placed in the same subnet at the Expressway. D. However. on the VCS under System > configuration > Registrations Answer: B Q109Which statement about configuring the Cisco VCS Control and Cisco VCS Expressway is true? A. The customer firewall must be configured with any rule for the IP address of the external Jabber client. on the voice-enabled firewall at the edge of the network B. but it must be located in the DMZ with the Expressway. C.B. System > Service Parameters > RSVP C. You do not need to configure search rules for traversal calls. The Cisco VCS Expressway is the Traversal Server. C. Where is this option enabled? A. Answer: AD Q107A new administrator at Company X has deployed a VCS Control on the LAN and VCS Expressway in the DMZ to facilitate VPN-less SIP calls with users outside of the network. on each MGCP gateway at all remote locations . on the VCS under Configuration > registration > configuration C. the users report that calls via the VCS are erratic and not very consistent. System > Call Manager > Clusterwide > Service Parameters > RSVP B. Jabber cannot connect to Cisco UCM unless it is on the same network or a VPN is set up from outside. C. Where does the administrator go to create a default profile? A. One such option is the control of trusted endpoints via a whitelist. B. Answer: D Q110Company X wants to implement RSVP-based Call Admission Control and move away from the current location-based configuration. The username on the Cisco VCS Control and Cisco VCS Expressway are local and do not need to match. System > Service Parameters > Call Manager > Clusterwide parameters > RSVP D. Answer: B Q108The VCS Expressway can be configured with security controls to safeguard external calls and endpoints. on the TMS server under Registrations > whitelist D. The VCS Control should not be on the LAN. B. You need to configure the firewall to allow communication from the Cisco VCS Expressway to the Cisco VCS Control. The firewall at Company X must have all SIP ALG functions disabled. What must the administrator configure on the firewall to stabilize this deployment? A. The Expressway server needs a neighbor zone created that points to Cisco UCM. E. D. D.

service parameters C. enterprise phone configuration D. enterprise parameters B. Ethernet configuration Answer: B Q112Refer to the exhibit.Answer: C Q111Where can you change the clusterwide DSCP setting for Cisco Unified Communications Manager? A. What is the correct value to use for the "DSCP for TelePresence Calls" Cisco CallManager service parameter? .

A. 28 (011100)
B. 34 (100010)
C. 41 (101001)
D. 46 (101110)
Answer: B
Q113Which two statements about remote survivability are true? (Choose two.)
A. SRST supports more Cisco IP Phones than Cisco Unified Communications Manager Express
in SRST mode.
B. Cisco Unified Communications Manager Express in SRST mode supports more Cisco IP
Phones than SRST.
C. MGCP fallback is required for ISDN call preservation.
D. MGCP fallback functions with SRST.
Answer: AD
Q114Which two options enable routers to provide basic call handling support for Cisco Unified IP
Phones if they lose connection to all Cisco Unified Communications Manager systems? (Choose
two.)
A. SCCP fallback
B. Cisco Unified Survivable Remote Site Telephony
C. Cisco Unified Communications Manager Express
D. MGCP fallback
E. Cisco Unified Communications Manager Express in SRST mode
Answer: BE
Q115When considering Cisco Unified Communications Manager failover, how many backup
servers can be configured in a Cisco Unified Communications Manager Group?
A. 1
B. 5
C. 2
D. 4
E. 3
F. 6
Answer: C
Q116Which three CLI commands are used when configuring H.323 call survivability for all calls?
(Choose three.)
A. voice service voip
B. telephony-service
C. h323
D. call preserve
E. call-router h323-annexg
F. transfer-system
Answer: ACD
Q117When configuring Cisco Unified Survivable Remote Site Telephony, which CLI command

enables this feature on the router?
A. call-manager-fallback
B. ccm-manager redundant-host
C. ccm-manager sccp local
D. ccm-manager switchback
Answer: A
Q118How long is the default keepalive period for SRST in Cisco IOS?
A. 45 sec
B. 30 sec
C. 60 sec
D. 120 sec
Answer: B
Q119Which option is a valid test scenario to verify that Cisco Unified Communications Manager
failover works and endpoints register with the backup call manager?
A. During a predetermined maintenance window, stop the Cisco IP Phone Services service on the
primary Unified CM. Devices should reregister with the backup Unified CM in the Cisco
CallManager Group.
B. During a predetermined maintenance window, stop the Unified CM service on the Publisher.
Devices should reregister with the backup Publisher in the Cisco CallManager Group.
C. During a predetermined maintenance window, stop the TFTP service on the primary call
manager. Devices should reregister with the backup Unified CM in the Cisco CallManager Group.
D. During a predetermined maintenance window, stop the Unified CM service on the primary call
manager. Devices should reregister with the backup Unified CM in the CallManager Group.
Answer: D
Q120Which three commands can be used to verify SRST fallback mode? (Choose three.)
A. show telephony-service all
B. show telephony-service ephone-dn
C. show telephony-service ephone
D. show telephony-service voice-port
E. show telephony-service tftp-bindings
Answer: ABC
Q121Company X has a Cisco Unified Communications Manager cluster and a Cisco Unity
Connection cluster at its head office and implemented SRST for its branch offices. One Monday
at 2:00 pm, the WAN connection to a branch office failed and stayed down for 45 minutes. That
day the help desk received several calls from the branch saying their voicemail was not working
but they were able to make and receive calls.
Why did the users not realize the WAN was down and prevented access to their voicemail?
A. All the phones should have started ringing the instant the WAN connection failed to signal the
start of SRST mode.
B. All calls should have dropped when the WAN failed so users would be instantly aware.
C. Although the phones were still working, the users should have noticed that the phone displays
said "SRST Fallback Active".
D. The voice administrators at the head office did not call the users to notify them.

Answer: C
Q122What are two important considerations when implementing TEHO to reduce long-distance
cost? (Choose two.)
A. on-net calling patterns
B. E911 calling
C. number of route patterns
D. caller ID
Answer: BD
Q123Which two statements about the use of the Intercluster Lookup Service in a multicluster
environment are true? (Choose two.)
A. Cisco Unified Communications Manager uses the ILS to support intercluster URI dialing.
B. ILS contains an optional directory URI replication feature that allows the clusters in an ILS
network to replicate their directory URIs to the other clusters in the ILS network.
C. Directory URI replication does not need to be enabled individually for each cluster.
D. To enable URI replication in a cluster, check the Exchange Directory URIs with Remote
Clusters check box that appears in the SIP trunk configuration menu.
E. If the ILS and directory URI replication feature is disabled on a cluster, this cluster still accepts
ILS advertisements and directory URIs from other neighbor clusters; it just does not advertise its
local directory URIs.
Answer: AB
Q124In Cisco Unified Communications Manager, where do you configure the +E.164 number that
is advertised globally via ILS?
A. ILS configuration under Advanced Features
B. +E.164 alternate number under Directory Number Settings
C. Device Information under Phone Configuration
D. Route Pattern under Route/Hunt
Answer: B
Q125When implementing a dial plan for multisite deployments, what must be present for SRST to
work successfully?
A. dial peers that address all sites in the multisite cluster
B. translation patterns that apply to the local PSTN for each gateway
C. incoming and outgoing COR lists
D. configuration of the gateway as an MGCP gateway
Answer: B
Q126Which code snippet is required for SAF to be initialized?
A. Service Family
B. External-Client
C. router eigrp
D. topology base
Answer: C

The SAF Client Control is a non-configurable inherent component of the Cisco IOS Routers. B. Answer: B Q129If you want to delete a SAF-enabled trunk from Cisco Unified Communications Manager Administration. B. C. C. which two IP phone settings will be modified by Device Mobility so that the phone can place and receive calls in New . Append an @ symbol at the end of the client label value in the SAF Forwarder configuration page. Disassociate the trunk from the CCD advertising service or CCD requesting service. D. Configure the correct node in the EIGRP configuration of the gateway router that is associated with the Cisco Unified Communications Manager node. Answer: B Q128Which statement about the SAF Client Control is correct? A. route pattern C. ILS updates D. Place the Cisco Unified Communications Manager node in standby mode. Fullmesh B. Delete the trunk from the CCD requesting service node. C. Redirect CCD advertising and requesting services to another Cisco Unified Communications Manager. Configure the publisher node only in the SAF Forwarder configuration page. Automesh C. D. If an IP phone in San Jose roams to New York. Answer: A Q130Which functionality does ILS use to link all hub clusters in an ILS network? A. route string D.Q127When using SAF. Configure the SAF Security Profile Configuration to support only a single node. The SAF Client Control is a configurable inherent component of the Cisco IOS Routers. B. multicast Answer: B Q131Which option is known as the location attribute that the global dialplan replication uses to advertise its dial plan information? A. URI Answer: C Q132Refer to the exhibit. location controller B. D. The SAF Client Control is a configurable inherent component of Cisco Unified Communications Manager. how do you prevent multiple nodes in a cluster from showing up in the Show Advance section of the SAF Forwarder configuration? A. The SAF Client Control is a non-configurable inherent component of Cisco Unified Communications Manager. what must you do first? A.

so the phone is considered to be in its home location. Device Mobility will reconfigure the roamingsensitive settings of the phone. The Extension Mobility log in fails. B. C.) A. The device mobility groups are the same. C. Create the extension mobility IP Phone Service. Answer: B Q134Which three steps are required when configuring extension mobility in Cisco Unified Communications Manager? (Choose three. The Device Mobility information is associated with one or more device pools other than the home device pool of the phone. so one of the associated device pools is chosen based on a round-robin load-sharing algorithm. . The physical locations are different. The device takes on the default device profile for its type. The user can log in but does not have access to any features. D. The physical locations are not different. Answer: BC Q133What happens when a user logs in using the Cisco Extension Mobility Service on a device for which the user has no user device profile? A. so the configuration of the phone is not modified. B. so the Device Mobility-related settings are applied in addition to the roaming-sensitive parameters. so the roaming-sensitive parameters of the roaming device pool are applied. The Device Mobility information is associated with the home device pool of the phone.York? (Choose two. The device uses the first device profile assigned to the user in Cisco Unified Communications Manager. or button templates. D. E. soft key templates.) A.

10 C. Unsubscribe all other services from the Cisco IP Phone. PVDM3-32 B. D. 6 Answer: B Q136When configuring Cisco Unified Mobility. Rerouting Calling Search Space under Remote Destination Profile Information D. Check the Enable Extension Mobility checkbox on the Directory Number Configuration page. Calling Search Space under Phone Configuration Answer: C Q137Which two bandwidth management parameters are available during the configuration of Cisco Unified Communications Manager regions? (Choose two. PVDM3-128 D. Default Video Call Rate D. PVDM3-192 . Create a user Device Profile. Cisco CallManager service under Service Parameter Configuration D.) A. Enterprise Phone Configuration Answer: C Q139Which module is the minimum PVDM3 module needed to support video transcoding? A. PVDM3-64 C. Max Audio Bit Rate C. where do you configure the default bit rate for audio and video devices? A. Check the Home Cluster checkbox on the End User Configuration page. which parameter defines the access control for a call that reaches out to a remote destination? A. E. User Local under Remote Destination Profile Information C. Max Video Call Bit Rate (Includes Audio) E. Enterprise Parameters B. Answer: AEF Q135How many Cisco Unified Mobility destinations can be configured per user? A.B. Calling Party Transformation Calling Search Space under Remote Destination Profile Information B. 1 B. Rerouting Calling Search Space under Remote Destination information E. Region under Region Information C. Subscribe the extension mobility IP Phone Service to the user Device Profile. Default Audio Call Rate B. F. Max Number of Video Sessions Answer: BD Q138In Cisco Unified Communications Manager. C. 4 D.

D. the directory number that is assigned to the Cisco TelePresence EX90 Answer: B Q144In a distributed call processing network with locations-based CAC. calls are routed to and from intercluster trunks.729 requires 24K of bandwidth per call. intercluster trunk without gatekeeper control C. TLC queries for the time zone as part of configuration. without the domain portion C. you must configure MTP resources on the gatekeeper and only use G. if you have servers in a cluster in a different time zone. which portion of the destination URI is the is the first match that is attempted? A. intercluster trunk with gatekeeper control B. C. the E. B.711 between regions. the full URI. h225 trunk Answer: B . To deploy a Cisco H.164 number that is assigned to the Cisco TelePresence EX90 D. TLC adjusts the time change appropriately. D. what determines the number of sessions that is supported on each DSP? A. G. SIP trunk D. G. The default codec does not matter if you have defined a hardware MTP in your Cisco Unified Communications Manager environment. TLC produces an error and must be run remotely. which time is used? A. the number of full-duplex media streams D. the Cisco Unified Communications Manager node setting Answer: A Q141Which statement is true when considering a Cisco VoIP environment for regional configuration? A. the destination alias. the codecs that are used in universal transcoding mode B.711 requires 128K of bandwidth per call. Which trunk type is implemented in this network? A. Answer: A Q143When a call is made from a video endpoint to a Cisco TelePresence EX90 that is registered to a Cisco VCS Control. B. the size of the cluster that is being designed C. Answer: B Q142When you use the Query wizard to configure the trace and log central feature to collect install logs. including the domain portion B. C.Answer: C Q140When implementing a Media Termination Point.323 gatekeeper. TLC uses its local time for all systems.

Q145Refer to the exhibit. cs4 B. af23 D. The configuration is automatically selected when RSVP is enabled for the location assigned to the trunk. EF/46 B. SIP precondition is configured by selecting E2E for RSVP over SIP on the default SIP profile assigned to the SIP trunk. What is the correct DSCP value to use when configuring a class map in a Cisco IOS router? A. af41 Answer: D Q146What is the default DSCP/PHB for TelePresence video conferencing packets in Cisco Unified Communications Manager? A. D. The configuration is done by selecting a SIP precondition trunk for trunk type. ef C. C. The "DSCP for Video Calls" Cisco CallManager service parameter is set to 34. CS3/24 E. SIP precondition is configured by configuring a new SIP profile and selecting E2E for RSVP over SIP. . CS4/32 Answer: E Q147How is a SIP trunk in Cisco Unified Communications Manager configured for SIP precondition? A. B. AF41/34 D. CS6/48 C.

the IP phones must have access to an RSVP agent. D. When configuring SIP precondition. When configuring SIP precondition. the SIP trunk must have access to an RSVP agent. the IP phones and SIP trunk must have access to an RSVP agent.The new SIP profile must then be assigned to the SIP trunk. E. When configuring SIP precondition. SIP trunks require RSVP agents only when fall back to local RSVP is configured. RSVP agents are only required for the IP phones. SIP trunk will always require RSVP agents regardless of what RSVP type is configured. Answer: D Q148Which statement about SIP precondition is most correct? A. B. C. How should the HQ Cisco Unified Communications Manager be configured for calls to 3XXX to be sent to the gatekeeper at 1 0 1 6 1 with PSTN backups? . Assume that NANP is being used and 9 is used for PSTN access code Long distance national calls are preceded with 1. Answer: D Q149Refer to the exhibit.

A. Configure a route pattern for 3XXX Assign this route pattern to a route list that points to two
route groups. The first route group contains the H 225 trunk. The second route group contains the
MGCP gateway with prefix digits 1 408555 for the outgoing called number.
B. Configure a route pattern for 1#3XXX Assign this route pattern to a route list that points to a
route group that lists the H 225 trunk as first choice and the MGCP gateway as a second choice.
C. Configure a route pattern for 4085543XXX.
Assign this route pattern to a route list that points to two route groups. The first route group
contains the H 226 trunk. The second route group contains MGCP gateway.
D. Configure a route pattern for 3XXX Assign this route pattern to a route list that points to two
route groups. The first route group contains the H 225 trunk. The second route group contains
MGCP gateway with prefix digits 91 408554 for the called number.
Answer: A
Q150Refer to the exhibit. IT shows an H.323 gateway configuration in a Cisco Unified
Communications Manager environment. An inbound PSTN call to this H.323 gateway fails to
connect to IP phone extension 2001. The PSTN user hears a reorder tone. Debug isdn q931 on
the H.323 gateway shows the correct called-party number as 5015552001.

Which two configuration changes can correct this issue? (Choose two.)
A. Add port 1/0:23 to dial-peer voice 123 pots.
B. Ensure that the Significant Digits for inbound calls on the H.323 gateway configuration is 4.
C. Add a voice translation profile to truncate the number from 10 digits to 4 digits. Apply the voice
translation profile to the Voice-port. The configuration field "Significant Digits for inbound calls" is
left at default (All).
D. Add the command h323-gateway voip id on interface vlan120.
E. Change the destination-pattern on the dial-peer voice 23000 VoIP to 501501? and change the
Significant Digits for inbound calls to 4.
Answer: BE
Explanation:
Choose the number of significant digits to collect, from 0 to 32. Cisco Unified Communications
Manager counts significant digits from the right (last digit) of the number that is called.
Q151Which device is needed to integrate H.320 into the Cisco video solution?
A. video gateway
B. MGCP gateway

C. H.323 gatekeeper
D. MCU
Answer: C
Explanation:
As with H.323 MCUs, H.320 gateways are provisioned in Cisco Unified CallManager as H.323
gateways, and then route patterns are configured to extend calls to these devices.
Q152The following exhibit shows configs for H.323 gateway. You have been asked to implement
TEHO from a remote branch office with area code 301 to the HQ office with area code 201 using
Cisco Unified Communications Manager. The remote office has an MGCP gateway and the HQ
office has an H.323 gateway. Once the call arrives at the HQ, it should break out to the PSTN as
a seven-digit local call. Which statement about the route pattern is true?

A. route pattern should be 91201.[2-9]XXXXXX with Discard Digit as Predot and Prefix 9
B. route pattern should be 91201.[2-9]XXXXXX with Discard Digit as Predot
C. route pattern should be 91201.[2-9]XXXXXX
D. route pattern should be 9.1201[2-9]XXXXXX with Discard Digit as Predot
E. route pattern should be 9.1201[2-9]XXXXXX with Discard Digit as Predot and Prefix 9
Answer: A
Q153Which two statements about the functionality of a gatekeeper are true? (Choose two.)
A. Cisco Unified Communications Manager has gatekeeper functionality built in.
B. Cisco Unified Communications Manager registers with a gatekeeper via SIP.
C. Cisco Unified Communications Manager registers with a gatekeeper via H.323.
D. A gatekeeper can enable CAC and AAR.
E. A gatekeeper can enable CAC, but not AAR.
Answer: CE
Q154Which option describes a function of SIP preconditions?
A. SIP preconditions enable end-to-end RSVP over an SIP trunk.
B. SIP preconditions enable RSVP between Cisco IP Phones.
C. SIP preconditions can be enabled in a gatekeeper.
D. SIP preconditions enable end-to-end RSVP for calls through the PSTN.
Answer: A
Q155Which statement about the function of a gatekeeper is true?

A gatekeeper can replace the dial plan of a Cisco Unified Communications Manager cluster. Pool Profile D. C. Answer: B Q158What user profile is used to define the settings for a user on login? A. Answer: A Q160What command is used to map internal extensions to the corresponding E. C. C. Group Profile C. Answer: C Q156For which VoIP protocol does a gatekeeper provide address translation and control access? A. dialplan-pattern C. SIP C. ephone-dn B.711 audio calls? A.323 B. B. Gatekeepers can be implemented to deploy RSVP-based CAC. SRST can be configured to function with no Cisco Unified Communications Manager cluster in the enterprise.A.248 Answer: A Q157Which CAC configuration on a gatekeeper restricts to 10 G. Cisco Unified Communications Manager Express in SRST mode can be configured to function with no Cisco Unified Communications Manager cluster in the enterprise. D. D. number-e. Specific Profile Answer: A Q159Which statement about technology implementation strategy is true? A. Use the command bandwidth 160. A gatekeeper can simplify the dial plan between many different Cisco Unified Communications Manager clusters.164 . H. D. number D. Skinny D. B. Use the command bandwidth 10. H. SRST and MGCP fallback can be configured to function with no Cisco Unified Communications Manager cluster in the enterprise. Device Profile B. A gatekeeper improves call routing between servers within a single Cisco Unified Communications Manager cluster. B.164 PSTN number when using Cisco Unified Communications Manager Express in SRST mode? A. Use the command bandwidth session 10. Cisco Unified Communications Manager Express can be configured to function with no Cisco Unified Communications Manager cluster in the enterprise. Use the command bandwidth 1280.

CAPF enrollment supports the use of authentication strings. Understanding Cisco IP Telephony Authentication and Encryption Fundamentals) A. GARP is normally used for HSRP. MICs are issued by the CAPF itself or by an external CA. C. A. After you add the user to this group and install the certificate. Symmetric encryption is commonly used to sign asymmetric keys. B. a security token is added to the system D. B. the application ensures that the user connects via the TLS port. D. D. GARP can be used for a man-in-the-middle attack. C. JTAPI. Set up an IPsec association between the application and Cisco Unified CallManager. B. an IP address of the Cisco TFTP server has been changed Answer: B Q163Which statement is not true about GARP? A. D. Q165Which two statements about symmetric encryption are true? (Choose two. With symmetric encryption. the encryption key equals the decryption key. Answer: CD Explanation: You must also add the application users or the end users to the Standard CTI Secure Connection user group in Cisco Unified Communications Manager Administration to enable TLS for the application. an LSC of the IP phone is upgraded C. Add the application user or end users to the Standard CTI Secure Connection user group.E. and TAPI applications requires which two tasks? (Choose two. Enter the encryption key into the application. C. JTAPI. LSCs are issued by the Cisco CTL client or by the CAPF. Answer: A Q162An update of the configuration using the Cisco CTL client not needed when _______. a Cisco Unified CallManager has been removed B. Configure related security parameters in the CTI. GARP can be disabled at Cisco IP phones. . C. ephone-transnumber Answer: B Q161Which statement about enrollment in the IP telephony PKI is true? (Source. Standard CTI Allow Reception of SRTP Key Material user group. and Standard CTI Enabled user group.) A. and TAPI application.) A. The CAPF itself has to enroll with the Cisco CTL client. Symmetric encryption is a good choice for real-time encryption of bulk data. Q164Enabling authentication and encryption for CTI. Answer: C Explanation: GARP (Gratuitous ARP) announce the presence of IP Phone on the network. GARP attacks require access to the target LAN or VLAN. B.

D. Symmetric encryption uses asymmetric keys. What happens when the fourth call is placed from HO to BR? . private key is kept secret. Assume that the priority queue has been provisioned correctly for three G.729 calls. Answer: AC Explanation: There are two basic techniques for encrypting information: symmetric encryption (also called secret key encryption) and asymmetric encryption (also called public key encryption. so that only you know it. Q166Refer to the exhibit. Locations-based CAC has been configured between HQ and the BR site. they can encrypt and decrypt all messages that use this key. A public key is made freely available to anyone who might want to send you a message.) in symmetric key As long as both sender and recipient know the secret key. A second.

The call will be queued until one of the existing calls drop. E. 384 kbps C.729 (at 24 kb/s). 192 kbps Answer: B Explanation: A 384-kb/s video call may comprise G. C. B. D. 768 kbps B. Q169While configuring Call Survivability in Cisco Unified Communications Manager. CallManager route patterns C. what is needed to route calls outside of the remote site location to the PSTN? A. what step is mandatory to reach remote sites while in SRST mode? A. B. Q168In what Cisco solution is Simple Network-Enabled Auto Provision technology used? A. Q170While operating in SRST. Cisco Unified SRST D. VOIP dial peers . SIP trunk B. Cisco Unified Call Survivability Answer: C Explanation: When the system automatically detects a failure. If the audio codec for a video call is G. Configure CFUR. Cisco Unified Gateway Duplication B. translation patterns D. Enable the SRST checkbox in the MGCP gateway. D. Enable the Failover Service parameter. but it will experience poor audio quality. Answer: B Q167Video calls using 384 kbps need to be supported across a gatekeeper-controlled trunk. Answer: B Explanation: Call Forward Unregistered (CFUR) functionality provides the automated rerouting of calls through the PSTN when an endpoint is considered unregistered due to a remote WAN link failure. The call will get through via the WAN.711 at 64 kb/s (for audio) plus 320 kb/s (for video). 512 kbps D.A. POTS dial peers E.323 gateway for SRST in Cisco Unified Communications Manager. C. The call will fail. Cisco Unified SRST uses Simple Network Auto Provisioning (SNAP) technology to auto-configure a branch office router to provide call processing for the Cisco Unified IP phones that are registered with the router. The call will get through without any issues. the video rate increases to maintain a total bandwidth of 384 kb/s. Cisco Unified CallManager Redundancy C. What value should be entered into the gatekeeper to support this bandwidth? A. Configure the H. Enable Cisco Remote Site Reachability. This sum does not include overhead.

Q171When using Cisco Unified Communications Manager Express in SRST mode. Q174What is the fastest way for an engineer to test the implementation of SRST in a production environment? A. 5 Answer: B Q172To preserve analog calls in an MGCP switchback event. 2 D.) A. D. IP phone display C. physical IP phone settings Answer: BE Explanation: IP Phone display and Physical phone IP settings are two locations were an end user can determine if an IP phone is working in SRST mode. Verification is not needed. 4 E. mgcp-graceful E. 3 B. Remove the null route when the operation is verified. mgcp-switchback-graceful C. Answer: C . 1 F.) A. E. B.Answer: D Explanation: in time of srst configuration on router. 6 C. Unplug the IP phones from their switch ports. Add a null route to the publisher Cisco Unified Communications Manager at the remote router. C. Shut down the Cisco Unified Communications Manager Servers. please configure a dial-peer so that call flow in SRST mode. Cisco Unified Communications Manager Administration B. Shut down the switch ports connected to the Cisco Unified Communications Manager Servers. preserve-h323 F. voice service voip D. h323 B. which three commands must be configured in the MGCP fallback router? (Choose three. how many multicast music on hold streams can be utilized by the system at any given time? A. Cisco Unified SRST Router D. Cisco Unified MGCP Fallback Router E. no h225 timeout keepalive Answer: ACF Q173Which two locations are the best locations that an end user can use to determine if an IP phone is working in SRST mode? (Choose two.

Q175Drag and Drop Question Answer: .

How to enable IP phones to establish calls to the PSTN when they have registered with the gateway? (Choose three. The default service must be enabled globally.) A. Answer: ABC Q177Which two configurations provide the best SIP trunk redundancy with Cisco Unified Communications Manager? (Choose two. Configure all SIP trunks with DNS SRV B. Configure all SIP trunks to point to a SIP gateway . B.Q176You are the Cisco Unified Communications Manager in Certpaper. D. C.) A. COR needs to be configured to disallow outbound calls. Configure all SIP trunks with Cisco Unified Border Element C. POTS dial peers must be added to the gateway to route calls from the IP phones to the PSTN.com. The command ccm-manager mgcp-fallback must be configured. You use a remote site MGCP gateway to provide redundancy when connectivity to the central Cisco Unified Communications Manager cluster is lost.

SCCP fallback D. Cisco IOS Software version D. The protocol that is used in Cisco Unified Communications Manager B. manually back to the secondary Cisco Unified Communications Manager cluster Answer: A Q181Which option configures call preservation for H. capacity of the Cisco Media Convergence Server F. call-manager-fallback preserve-call D. Configure SIP trunks to be members of route groups and route lists E. automatically back to the secondary Cisco Unified Communications Manager cluster D. or if a host name (configured on the SIP trunk) resolves to more than 10 IP addresses. Cisco Unified Communications Manager supports up to 16 IP addresses for each DNS SRV and up to 10 IP addresses for each DNS host name.323 fallback C. The order of the IP addresses depends on the DNS response and may be identical in each DNS query. MGCP fallback B. you need to configure the router for Survivable Remote Site Telephony in case the Cisco Unified Communications Manger stops working. Cisco Unified Communications Manager version C. WAN link bandwidth E.323 gateway Answer: AD Explanation: For SIP trunks.) A.323-based SRST mode? A. not because of any change in the status of any of the remote destinations. Thus. H. the status of a SIP trunk may change because of a change in the way a DNS query gets resolved. manually back to the primary Cisco Unified Communications Manager cluster C.323 C. automatically back to the primary Cisco Unified Communications Manager cluster B. On which two factors would the number of IP phones and Directory Numbers that can register to the SRST router depend? (Choose two. SIP fallback Answer: A Q180How does the system intelligently shift call processing upon restoration of WAN connectivity? A. The OPTIONS request may go to a different set of remote destinations each time if a DNS SRV record (configured on the SIP trunk) resolves to more than 16 IP addresses. dial-peer voice 1 voip call preserve . Q178When you configure Cisco Unified Communications Manager. call preservation not possible with H. voice service voip h323 call preserve B. router platform Answer: CF Q179Which remote-site redundancy technology fails over to POTS dial peers from the Cisco Unified Communications Manager dial plan during a WAN failure? A. Configure all SIP trunks to point to a gatekeeper through SIP to H. Configure all SIP trunks to allow TCP ports 5060 F.D.

ccm-manager switchover-to-backup B. Configure CSS and partitions in the SRST ISR. voice translation-rule C. D. Answer: C Q184Which command can be used to manually send the MGCP gateway to register with the secondary Cisco Unified Communications Manager server? A. application Answer: A Q183A Cisco Unified Communications Manager cluster is installed in headquarters only. ephone-dn 1 dual-line number 7001 description 7001 name 7001 ephone-dn-template 5 This DN is learned from srst fallback ephones B. C. incoming called-number D. not supported Answer: A Q185This is the configuration on the voice gateway: telephony-service max-ephones 30 max-dn 60 preference 0 srst mode auto-provision all srst dn line-mode dual srst dn template 3 srst ephone description srst fallback auto-provision phone srst ephone template 5 Which ephone-dn would be expected upon activation of SRST? A. Configure voice translations in the SRST ISR. How can international calls be blocked while national calls are allowed for branch office Cisco IP Phones during a WAN failure? A. mgcp use backup C. Configure CSS and partitions in Cisco Unified Communications Manager and apply the CSS and partitions to the SRST ISR. direct-inward-dial B. ccm-manager register backup D. ephone-dn 1 dual-line number 7001 description 7001 name 7001 .Answer: A Q182Which configuration command disables the secondary dial tone on the branch office ISR for users calling from the PSTN into the branch office during a WAN failure? A. Configure COR in the SRST ISR. B.

Initiating a process to provide call-processing backup redundancy. Automatically detecting a failure in the network. SIP proxy D. a means to allow the local site to continue to send and receive calls in the event of a WAN failure B. B2BUA B. Device Pool SRST Reference setting D. a means to route calls on-net through other sites during high utilization periods C. the ability to force a call out of a certain trunk when the Cisco Unified Communications Manager is being upgraded Answer: A Q187What component acts as a user agent for both ends of a SIP call while Cisco Unified SIP SRST is providing failover during a WAN outage? A. SRST Reference configured in Cisco Unified Communications Manager C. SIP registrar Answer: A Q188Which two configurations are needed to implement SRST in Cisco Unified Communications Manager? (Choose two. SIP SRST router E. SIP server C. a method that allows for backup calls in the event that your gateway fails D. B.) A. . Notifying the administrator of an issue for manual intervention. Call Manager Group setting E. ephone-dn 1 number 7001 description 7001 name 7001 ephone-dn-template 3 This DN is learned from srst fallback ephones Answer: A Q186Which ability does the Survivable Remote Site Telephony feature provide? A. C. Cisco Unified Communications Locations setting Answer: BC Q189Which of the following are two functions that ensure that the telephony capabilities stay operational in the remote location Cisco Unified SRST router? (Choose two) A.ephone-dn-template 3 This DN is learned from srst fallback ephones C. ephone-dn 1 number 7001 description 7001 name 7001 ephone-dn-template 5 This DN is learned from srst fallback ephones D. SRST Gateway setting in Cisco Unified Communications Manager B.

Enable SIP trunking between both remote and hub sites to provide mesh coverage. E. Answer: ABC Q191Which method can be used to address variable-length dial plans? A. Cisco recommends that you do not uncheck the check box. Cisco Unified Communications Manager does not route the call until the interdigit timer expires (even if it is possible to dial a sequence of digits to choose a current match). Answer: AB Q190Which three of the following are steps in configuring MGCP Fallback and Cisco Unified SRST? (Choose three) A.D. Implement a simplified SRST dial plan on the remote-site-gateways to ensure connectivity for remote-site phones in SRST mode. Define the SRST reference in the configuration on the IP Phones. Check this check box to interrupt interdigit timing when Cisco Unified Communications Manager must route a call immediately. Enable and configure the MGCP fallback and Cisco Unified SRST features on the IOS gateways. Use nested translation patterns to eliminate inter-digit timeout D. F. Define the SRST reference for phones in the Device Pool configuration B. Add a prefix for all calls that are longer than 10-digits long C. B. Unless your dial plan contains overlapping patterns or variable length patterns that contain!. Enable and configure the MGCP fallback on the IOS gateway but not Cisco Unified SRST since it is enabled automatically. Overlap sending and receiving. Use the @macro on the route pattern E. the Urgent Priority check box displays as checked. which support variable-length dial plans Answer: A Explanation: If the dial plan contains overlapping patterns. Which trunks would be most suitable for Connection Y? . Proactively repairing issues in the voice network. Q192Refer to the exhibit. Use MGCP gateways. C. D. By default.

A. an H.225 trunk (gatekeeper-controlled)
B. intercluster trunk (gatekeeper-controlled)
C. a SIP trunk on the U.S. cluster and an intercluster trunk on the remote cluster
D. intercluster trunk (nongatekeeper-controlled)
E. No extra connections are required. As long as IP connectivity exists, you need only configure a
route pattern for each site. The Cisco Unified Communications Manager will automatically forward
the calls over the WAN if the destination directory number is not registered locally.
Answer: D
Q193Which two features require or may require configuring a SIP trunk? (Choose two.)
A. SIP gateway
B. Call Control Discovery between a Cisco Unified Communications Manager and Cisco Unified
Communications Manager Express
C. Cisco Device Mobility
D. Cisco Unified Mobility
E. registering a SIP phone
Answer: AB
Explanation:
All protocols require that either a signaling interface (trunk) or a gateway be created to accept and
originate calls. Device mobility allows Cisco Unified Communications Manager to determine
whether the phone is at its home location or at a roaming location. Cisco Unified Mobility gives
users the ability to redirect incoming IP calls from Cisco Unified Communications Manager to
different designated phones, such as cellular phones.
Q194A Cisco 3825 needs to be added in Cisco Unified Communications Manager as an H.323
gateway. What should the gateway type be?
A. H.323 gateway
B. Cisco 3825

C. Cisco 3800 series router. The specific model will be selected after the Cisco 3800 is selected.
D. The gateway can be added either as an H.323 gateway or Cisco 3800 series router.
E. The gateway can be added either as an H.323 gateway or Cisco 3825 series router.
Answer: A
Q195When an incoming PSTN call arrives at an MGCP gateway, how does the calling number
get normalized to a global E.164 number with the + prefix in Cisco Unified Communications
Manager?
A. Normalization is done using translation patterns.
B. Normalization is done using route patterns.
C. Normalization is done using the gateway incoming called party prefixes based on number type.
D. Normalization is done using the gateway incoming calling party prefixes based on number
type.
E. Normalization is achieved by local route group that is assigned to the MGCP gateway.
Answer: D
Explanation:
Configuring calling party normalization alleviates issues with toll bypass where the call is routed
to multiple locations over the IP WAN. In addition, it allows Cisco Unified Communications
Manager to distinguish the origin of the call to globalize or localize the calling party number for the
phone user.
Q196When an incoming PSTN call arrives at an MGCP gateway, how does the called number get
normalized to an internal directory number in Cisco Unified Communications Manager?
A. Normalization is done by configuring the significant digits for inbound calls on the MGCP
gateway.
B. Normalization is done using route patterns.
C. Normalization is done using the gateway incoming called party prefixes based on number type.
D. Normalization is done using the gateway incoming calling party prefixes based on number
type.
E. Normalization is achieved by local route group that is assigned to the MGCP gateway.
Answer: A
Q197Which process can localize a global E.164 with + prefix calling numbers for inbound calls to
an IP phone so that users see the calling number in a local format?
A. Calling number localization is done using translation patterns.
B. Calling number localization is done using route patterns.
C. Calling number localization is done by configuring a calling party transformation CSS at the
phone.
D. Calling number localization is done by configuring a calling party transformation CSS at the
gateway.
E. Calling number localization is done by configuring the phone directory number in a localized
format.
Answer: C
Q198Refer to the exhibit. The exhibit shows centralized Cisco Unified Communications Manager
configuration components for TEHO calls to U.S. area code 408 from the U.K. The PSTN access
code for the U.K. is 9 and 001 for international calls to the U.S. To match the US-TEHO pattern \
+!, how should the translation pattern be configured?

A. 9001.4085551234 with the Called Party Transformation:
Discard Digits PreDot
Prefix Digits Outgoing Calls: +
B. 9.0014085551234 with the Called Party Transformation:
Discard Digits PreDot
Prefix Digits Outgoing Calls: +1
C. 900.14085551234 with the Called Party Transformation:
Discard Digits PreDot
Prefix Digits Outgoing Calls: +1
D. 900.14085551234 with the Called Party Transformation:
Discard Digits PreDot
Prefix Digits Outgoing Calls: +
E. 001.4085551234 with the Called Party Transformation:
Prefix Digits Outgoing Calls: +
Answer: D
Explanation:
The PSTN access code for the UK is 9, International call code is 001, The international escape
character, +, signifies the international access code in a complete E.164 number format.
Q199Refer to the exhibit. The exhibit shows centralized Cisco Unified Communications Manager
configuration components for TEHO calls to U.S. area code 408 from the U.K. The PSTN access
code for the U.K. is 9 and 001 for international calls to the U.S. What should the TEHO-US route
list configuration consist of?

K. First route group should be only the local route group. What should the Called Party Transformation Pattern at the U.S.K. gateway. is 9 and 001 for international calls to the U. First route group should point only to the U.S. The globalized configuration means that the appropriate gateway will be selected automatically. First route group should point only to the U. The second route group should be the local route group. gateway. Q200Refer to the exhibit. gateway be configured as? . gateway.K.S.S. The route group can serve as a trunk group by directing all calls to the primary device and then using the secondary devices when the primary is unavailable. The second route group should point to the U. Answer: C Explanation: The route group points to one or more gateways and can choose the gateways for call routing based on preference. The TEHO-US route list should contain only the local route group. gateway.A. B.S. The PSTN access code for the U. The second route group should point to the U. gateway. area code 408 from the U.S. The \+! route pattern should point directly to the U.S. C. D. One or more route lists can point to the same route group. Assuming the PSTN does not accept globalized numbers with + prefix. The exhibit shows centralized Cisco Unified Communications Manager configuration components for TEHO calls to U. E.

\+1.! with the following Called Party Transformation: Discard Digits PreDot Prefix Digits Outgoing Calls: None C. followed by 1. Assume that all outgoing PSTN numbers are localized at the egress gateway as shown in the exhibit. Both sites use MGCP gateways.! with the following Called Party Transformation: Discard Digits PreDot Prefix Digits Outgoing Calls: None E. and then the 10-digit number.! with the following Called Party Transformation: Discard Digits PreDot Prefix Digits Outgoing Calls: 1 D. an HQ or BR1 user dials access code 9. The HQ site uses area code 650.408! with the following Called Party Transformation: Discard Digits PreDot Prefix Digits Outgoing Calls: None Answer: D Q201Refer to the exhibit. \+. What should the AAR group prefix be? .! with the following Called Party Transformation: Discard Digits PreDot Prefix Digits Outgoing Calls: + B. AAR must use globalized call routing using a single route pattern. To make a long distance national call. \+1.A. The long distance national code for PSTN dialing is 1. \+408. \+1408. The BR1 site uses area code 408.

followed by 1. Assume that all outgoing PSTN numbers are localized at the egress gateway as shown in the exhibit. and then the 10-digit number. Both sites use MGCP gateways. To make a long distance national call.A. 9 B. none D. The BR1 site uses area code 408. AAR must use globalized call routing using a single route pattern. Which partition should be configured in the AAR CSS applied at the phones? . an HQ or BR1 user dials access code 9. 91 C. + E. The long distance national code for PSTN dialing is 1. +1 Answer: C Q202Refer to the exhibit. The HQ site uses area code 650.

Both sites use MGCP gateways. The long distance national code for PSTN dialing is 1. The BR1 site uses area code 408. LD partition C. and then the 10-digit number. To make a long distance national call. The HQ AAR CSS must include a partition assigned to route pattern 91408XXXXXXX. Assume that all outgoing PSTN numbers are localized at the egress gateway as shown in the exhibit. followed by 1. The BR1 AAR CSS must include a partition assigned to route pattern 91650XXXXXXX.1[2-9]XX[2-9]XXXXXX for each site that must be globalized. How many route lists and route groups should be configured for AAR at a minimum? . PSTN partition B.A. Answer: A Q203Refer to the exhibit. Otherwise the called numbers will not be localized at the egress gateway. an HQ or BR1 user dials access code 9. AAR must use globalized call routing using a single route pattern. AAR CSS must contain translation pattern 9. D. The HQ site uses area code 650.

A. user logs onto an RTP phone using the Cisco Extension Mobility feature and places an emergency call to 0000. two route lists and two route groups for each site C. Which statement about the emergency call is true? . and remote site in RTP. A U. a single route list and four route groups for each sitea D.K.K and RTP device pools. All route patterns are assigned a route list that points to the local route group. None. The AAR CSS can point directly to the route pattern. Local route groups have been configured on the U. K. Answer: A Q204Assume a centralized Cisco Unified Communications deployment with the headquarters in the A. a single route list with a local route group for each site B.

The call will fail. F. E. The call will match the U. The MGCP gateway has the following configurations: . Answer: A Q205Refer to the following exhibit.K_Emergency route pattern partition and will egress at the U.K.B. The call will match the RTP_Emergency route pattern partition and will egress at the U. D.K. The call will match the U. gateway. gateway.K_Emergency route pattern partition and will egress at the RTP gateway. The call will match the RTP_Emergency route pattern partition and will egress at the RTP gateway. C.

.

The called number is 011 49403021 56001. The called number is 01 1 49403021 56001. The calling number will be 5553001 and number type set to subscriber. The calling number will be 5215553001 and number type set to national.Pt) call. The calling number will be 5215553001 and number type set to national.called party transformation CSS HQ_cld_pty CSS (partition=HQ cld_pty. When the IP phone at extension 3001 places a call to 9011 49403021 56001# what is the resulting called and calling number that is sent to the PSTN? A. B.ng party transformation CSS HQ_clng_pty CSS (partition=HQ_clng_pty Pt) All translation patterns have the check box "Use Calling Party's External Phone Number Mask" enabled. . The called number is 4940302156001 with number type set to international. C.

4989S05552XXX and the ToDiD will be 0: D. The called number is +49403021 56001 with number type set to international. Use 10 digits in the domestic route pattern and 1 followed by the main number digits in the . You may also configure an External Phone Number Mask on all phone devices. Both +4989505552XXXand +4989531 21 2XXX will be advertised with ToDID of 0: Answer: A Q207When an external call is placed from Ajax. they would like the ANI that is sent to the PSTN to be the main number. How can this be accomplished? A. they would like to send the country code 1 and the 10 digits. For domestic calls. set the external phone number mask to the main number. 2XXX and the ToDID will be 0:+4989531 21 C. for international calls. Add a translation pattern to the dial peers in the gateway that adds the appropriate digits to the outgoing ANI. they would like 10 digits sent. Answer: A Explanation: Check the check box "Use Calling Party's External Phone Number Mask" if you want the full.D. 2XXX and the T0D1D will be 0:+498950555 B. not the extension. which pattern would be advertised? A. Q206Refer to the exhibit When the Cisco Unified Communications Manager advertises the Hosted DN Pattern. B. external phone number to be used for calling line identification (CLID) on outgoing calls. In the external call route patterns. The calling number will be 5215553001 and number type set to subscriber. + 4989631 21 2XXX and the ToDiD will be 0: E.

) A.164 transformation pattern represents phone numbers in Germany? A. Assume that the HQ phones have access to the HQ partition. Use a calling party transform mask for each route group in the corresponding route list configuration. Calling-party calls are routed to the gateway and trunks. \+49. C.! B. D. Calling-party numbers of internal calls are routed from the gateway or trunks.! C. the wildcard characters X. and octothorpe (#). D. Answer: AD Q211Refer to the exhibit.! D.323 gatekeeper-controlled network? A. H. and BR phones have access to the BR partition. This will be truncated to the 10-digit number for domestic calls and sent out in its entirety for international calls.225 C. \49. Q208Which trunk should you use in an H.X Answer: A Q210Which two statements are true regarding the implementation of globalized call-routing in terms of localized call egress? (Choose two.international route patterns. Set the explicit 10-digit main number for domestic calls and 1 followed by the main number for the international route patterns. B. asterisk (*).323 B. set the prefix digits field to the country code and the 10 digits of the main number. 49. Calling-party numbers are routed from the gateway or trunks to phones. MGCP T1/E1 trunk Answer: B Q209Which E. H. Which set of implementations would best address the overlapping directory number extensions for intersite (WAN) calling between the HQ site and the BR site? . \+49. Answer: C Explanation: calling party transformation mask value is Valid entries for the NANP include the digits 0 through 9. and the international escape character +. SIP D. Intercluster E. C. MGCP FXO trunk F. Called-party numbers are routed from the gateway or trunks to phones. In the directory number configurations.

12. Configure a translation pattern 8222. Configure a route pattern for site BR 8111.12]. Configure a translation pattern 8111. Use a route list that contains the local route group for each site. B. Use the local gateway at each site. Prefix the appropriate site code for the calling number.[1-32]XXX. Use a CSS that contains the partitions for BR phones. Configure a translation pattern 8222.[1-3]XXX. and assign it to partition HQ. and assign it to partition BR. Configure a single route pattern for both sites 8[12.323 gateway can be added in Cisco Unified Communications Manager under gateway type as H.[1-3]XXX for site BR.[12]XXX for site HQ.323 Gateway. D.[12]XXX for site HQ. An H.323 gateway and a SIP gateway? A. Use a CSS that contains the partitions for BR phones. For both translation patterns. The SIP gateway can connect to Cisco Unified Communications Manager only through a SIP . Configure a translation pattern 8111. configure the called party DDI of Predot. Configure a route pattern 8222. Configure the called party DDI of Predot. The H. For both translation patterns.323 gateway requires that dial peers be configured before PSTN calls can be placed and received. and assign it to partition HQ. B. Prefix the appropriate site code for the calling number. Configure called party DDI Predot. and assign it to partition BR. The SIP gateway requires no dial peers. Answer: C Q212What is the difference between an H. Use a CSS that contains the partitions for HQ phones. configure the called party DDI of Predot. Prefix the appropriate site code for the calling number. Use a CSS that contains the partitions for HQ phones. and assign it to partition BR.[12]XXX for site HQ. Prefix the appropriate site code for the calling number. and assign it to partition HQ.A. C.[1-3]XXX for site BR.

The MGCP gateway must be configured in Cisco Unified Communications Manager using the domain name.323 gateway. The SIP gateway requires no dial peers. A SIP gateway will show status of "Unknown". Normalization is done by configuring the significant digits for inbound calls on the H.323 gateway.323 gateway. An MGCP gateway can be added in Cisco Unified Communications Manager under the Gateway Type field using the gateway model.323 gateway does not require a call agent for PSTN calls to be placed and received. how does the called number get normalized to an internal directory number in Cisco Unified Communications Manager? A. Answer: B Q214When an incoming PSTN call arrives at an H. An MGCP gateway that dial peers be configured before PSTN calls can be placed and received. An H. B.323 gateway must be configured in Cisco Unified Communications Manager using a valid IP address on the gateway. B. how does the calling number get normalized to a global E.323 gateway can register with Cisco Unified Communications Manager.323 gateway configuration in Cisco Unified Communications Manager. Answer: D Q215When an incoming PSTN call arrives at an H. The SIP gateway must be configured in Cisco Unified Communications Manager using the domain name.trunk. Normalization is done using the gateway incoming calling party prefixes based on number type. Answer: A Q216Refer to the exhibit. D. C. D. The SIP gateway can connect to Cisco Unified Communications Manager only through a SIP trunk. Normalization is done using the gateway incoming called party prefixes based on number type. C. B. Normalization is done using the gateway incoming calling party prefixes based on number type. E. Normalization is done using route patterns. All HQ phones are configured to use HQ_MRGL and all BR phones are configured to use BR_MRGL.323 gateway. Normalization is achieved by local route group that is assigned to the H. An MGCP gateway does not require a call agent for PSTN calls to be placed and received. Normalization is done using route patterns. C. A SIP gateway requires a call agent for PSTN calls to be placed and received. D. E. An MGCP gateway can register with Cisco Unified Communications Manager. Normalization is done using translation patterns. For the HQ phones always to use the hardware conference bridge . A SIP gateway requires a call agent for PSTN calls to be placed and received. The H. C. An H. Answer: B Q213What is the difference between an MGCP gateway and a SIP gateway? A.164 number with + prefix in Cisco Unified Communications Manager? A. Normalization is achieved by local route group that is assigned to the H. A SIP gateway will show status of "Unknown". D. E. The SIP gateway must be configured in Cisco Unified Communications Manager using a valid IP address on the gateway.

Add both the HQ_MRG and HQ_MRG_2 to the HQ_MRGL and list the HQ_MRG first. B. Ensure that the instance ID for the hardware conference bridge is 0. Configure a second HQ_MRG_2 and assign the software conference bridge to it. Assign the hardware conference bridge to HQ_MRG. BR_MRGL is assigned to the BR IP phones.as a first choice. HQ_MRGL is assigned to the HQ IP phones. The HQ_MRGL_2 should be assigned to the HQ device pool. The HQ_MRGL should be assigned to the HQ phones. Ensure that both the hardware and software conference bridges are listed in the HQ_MRG. The remote site BR IP phones support only the G. Ensure that both the hardware and software conference bridges are listed in the HQ_MRG. Assign the hardware conference bridge to HQ_MRG. D. Where should the transcoder reside? . Add HQ_MRG_2 to HQ_MRGL_2. which configuration should be implemented? A. Configure a second HQ_MRG_2 and assign the software conference bridge to it. C. Add the HQ_MRG to HQ_MRGL. Answer: C Q217Refer to the exhibit.711 codec. Configure an additional HQ_MRGL_2. The hardware conference bridge must be configured first.

Which statement is true? . The ILS authentication password does not match. A transcoder is not needed. The HQ site uses area code 650. To make a long distance national call. an engineer gets the error message "Local cluster cannot connect to the ILS network". C. The transcoder should reside at the HQ site and assigned to HQ_MRG. Which three reasons for this error are true? (Choose three. and the other is using Password. Answer: B Q218When configuring intercluster URI dialing. Assume that all outgoing PSTN numbers are localized at the egress gateway as shown in the exhibit. and then the 10-digit number. The long distance national code for PSTN dialing is 1. F. One cluster is using TLS certificate. Both sites use MGCP gateways.) A. The transcoder should reside at the BR site and assigned to BR_MRG. B. The Cisco Unified Resource Identifier service needs a restart.711 codec. E. The BR1 site uses area code 408. D. The Tomcat certificates do not match. D. The SIP route patterns have not been properly configured. an HQ or BR1 user dials access code 9. The transcoder should be assigned to its own MRG. The cluster ID does not match. AAR must use globalized call routing using a single route pattern. The HQ phones will automatically change over to the G. followed by 1. which should then be assigned to the default device pool. B.A. Answer: BDF Q219Refer to the exhibit. C.

D. The AAR group system must be configured on the device configuration of the phones. The single AAR group system cannot be used. A second AAR group must be configured in order to have source and destination AAR groups. B. Which configuration elements must match for adjacent neighbors to establish a SAF neighbor relationship? A. Answer: B Q220Refer to the exhibit. The AAR group system must be configured on the line configuration of the phones.A. C. the autonomous-system number specified in the service-family ipv4 autonomous-system command C. The AAR group system must be configured under the AAR service parameters. the sf-interface configuration . the label name specified in the router eigrp command B.

What must be configured on the HQ Cisco Unified Communications Manager to allow HQ users to dial the SAF learned directory number pattern 3XXX? . The destination IP address will be learned automatically and configured on the SIP trunks after the Cisco Unified Communications Managers discover themselves. C.10. The HQ SIP trunk destination IP address should be the HQ SAF Forwarder IP address.6.D.1. Answer: B Explanation: The gatekeeper changes the IP address of this remote device dynamically to reflect the IP address of the remote device. The HQ SIP trunk destination IP address should be 10. B. The BR1 SIP trunk destination IP address should be the BR1 SAF Forwarder IP address. What should the destination IP address be configured as on the HQ and BR1 SIP trunks? A. The destination IP address is not configurable. The BR1 SIP trunk destination IP address should be 10. the topology base configurations E.5.10. Each cluster will learn about the remote trunk IP address through SAF learned routes.1. the label name specified in the router eigrp command and the autonomous-system number Answer: B Q221Refer to the exhibit. Q222Refer to the exhibit. D.

the call control discovery feature allows Cisco Unified Communications Manager to advertise itself along with other key attributes. E. The SAF partition assigned to the SAF learned patterns must be available to the HQ phone users through the phone CSS. The SAF directory number pattern 3XXX will be made available to all users automatically as soon as the SAF partition is selected. Answer: C Explanation: By adopting the SAF network service. B. Route pattern 3XXX should be configured and made available to HQ users through the phone CSS.A. Route pattern 3XXX should be configured and made available to HQ phone users through the phone AAR CSS. D. Which CSS is used at the HQ Cisco Unified Communications Manager to reroute calls via the PSTN when the SAF network is unavailable? . C. The SAF partition assigned to the SAF learned patterns must be available to the HQ phone users through the phone AAR CSS. Q223Refer to the exhibit.

If SAF patterns are accessible. How does the Cisco Unified Communications Manager advertise dnblock 1? . the phone device CSS B. the PSTN reroute is automatic. the phone line/device combined CSS D. the phone line CSS C. No special CSS is required. the SAF CSS configured on the CCD requesting service E. Answer: E Q224Refer to the exhibit.A. the phone AAR CSS configured at the phone device F.

4XXX and the ToDID will 0: B. 4XXX and the ToDID will 0:1972555 C. 4XXX D. How does the Cisco Unified Communications Manager advertise dnblock 2? . 4XXX and the ToDID will 0:+ 1972555 E.A. 19725554XXX Answer: B Q225Refer to the exhibit.

14087071222 Answer: C Q226Refer to the following exhibit. Which Cisco IOS SAF Forwarder configuration is correct? . +14087071222 with number type international C. +14087071222 D.A. 14087071222 with number type international B.

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When the Manager places a call to 3001 when the SAF network is down. Exhibit C D. Exhibit A B. Exhibit D Answer: A Q227Refer to the exhibit. what happens? A. CCD . The call is rerouted to the PSTN with the constructed PSTN number as +0002288223001 Answer: A Explanation: When the SAF forwarder loses network connection with its call-control entity. Exhibit B C. The call fails. The call is rerouted to the PSTN with the constructed PSTN number as 442288223001 D.A. the SAF forwarder withdraws those learned patterns that were published by the call control entity. B. The call is rerouted to the PSTN with the constructed PSTN number as 0002288223001 E. The call is rerouted to the PSTN with the constructed PSTN number as +442288223001 C. In this case.

service-family ! voice service saf profile trunk-route 1 session protocol sip interface Loopback1 transport tcp port 5060 ! profile dn-block 1 alias-prefix 1972555 pattern 1 type extension 4xxx ! profile callcontrol 1 dn-service trunk-route 1 dn-block 1 . router eigrp SAF i service-family ipv4 autonomous-system 1 ! topology base exit-sf-topology exit-service-family voice service saf profile trunkroute 1 session protocol sip interface Loopback1 transport tcp port 5060 ! B. and the calls gets routed through the PSTN gateway. Q228Refer to the exhibit. The exhibit shows a SAF Forwarder configuration attached to a Cisco Unified Communications Manager.requesting service marks those learned patterns as unreachable via IP. router eigrp SAF ! service-family ipv4 autonomous-system 1 ! topology base exit-sf-topology exit. Which minimum configuration for a Cisco Unified Communications Manager Express SAF Forwarder is needed to establish a SAF neighbor relationship with this SAF Forwarder? A.

set feature configuration parameters of Call Control Discovery E. < > B. Which three characters should you avoid entering in the description? (Choose three. Answer: C Q229Which Cisco IOS command is used for internal SAF Clients to check SAF learned routes? A.dn-block 2 ! channel 1 vrouter SAF asystem 1 subscribe callcontrol wildcarded publish callcontrol 1 ! C. enable enterprise parameter for Service Advertisement Framework forwarder Answer: CE Q231You are entering the description for the Service Advertisement Framework forwarder. % D. router eigrp SAF ! service-family ipv4 autonomous-system 1 ! topology base exit-sf-topology exit-service-family ! D. None of above configurations contain sufficient information. show voice saf routes C. create VPN groups B.) A. create VPN profiles C. create a new Service Advertisement Framework security profile D.) A.Patterns w/ invalid expr detected while add : 0 Patterns duplicated under the same instance : 0 Patterns rejected overall due to max capacity : 0 Attempts to delete a pattern which is invalid : 0 Last successful DB update @ 2009:12:14 15:42:45:967 Q230What are the two tasks that you must perform to configure the Service Advertisement Framework forwarder in Cisco Unified Communications Manager? (Choose two. show voice saf dndb all Answer: E Explanation: Router# show voice saf dnDb all Total no. show eigrp service-family ipv4 saf E. of patterns in db/max allowed : 1/6000 Patterns classified under dialplans (private/global) : 0/1 Informational/Error stats . configure Service Advertisement Framework forwarder information F. # . show voice saf detail D. show eigrp address-family ipv4 saf B. & C.

$ F. All the remaining Service Advertisement Framework forwarders are notified for their learned patterns. 3 D. as many as are configured Answer: B Q233Which two actions are performed by the Call Control Discovery service after the local Cisco Unified Communications Manager loses its TCP connection with the primary and secondary Service Advertisement Framework? (Choose two. 2 C. The Cisco Unified Communications Manager establishes a connection with the primary and secondary Service Advertisement Framework after the Learned Pattern IP Reachable Duration parameter expires. All learned patterns are purged from the local cache after the Call Control Discovery PSTN Failover Duration parameter expires. router eigrp SAF ! service-family ipv4 autonomous-system 1 ! topology base exit-sf-topology exit-service-family ! voice service saf profile trunk-route 1 session protocol sip interface Loopbackl transport tcp port 5060 i B. C. D.E. router eigrp SAF ! service-family ipv4 autonomous-system 1 ! topology base . @ Answer: ABC Q232In a cluster-wide deployment. The Service Advertisement Framework forwarder contacts all the remaining Service Advertisement Framework forwarders in the cluster. Answer: AB Q234Which minimum configuration is needed for the SAF Internal Client to register with this SAF Forwarder? A.) A. Calls are routed to the PSTN gateway after the Call Control Discovery Learned Pattern IP Reachable Duration parameter expires. what is the maximum number of Service Advertisement Framework forwarders to which the Cisco Unified Communications Manager can connect? A. Call Control Discovery immediately redirects all the calls to the PSTN gateway based on the learned patterns. B. 1 B. 4 E. F. 6 F. E.

router eigrp SAF ! service-family ipv4 autonomous-system 1 ! topology base exit-sf-topology exit-service-family ! voice service saf ! channel 1 vrouter SAF asystem 1 E. router eigrp SAF ! service-family ipv4 autonomous-system 1 ! topology base exit-sf-topology exit-service-family ! voice service saf profile trunk-route 1 session protocol sip interface Loopbackl transport tcp port 5060 ! profile dn-block 1 alias-prefix 1972555 pattern 1 type extension 4xxx ! profile callcontrol 1 dn-service trunk-route 1 dn-block 1 dn-block 2 ! channel 1 vrouter SAF asystem 1 subscribe callcontrol wildcarded publish callcontrol 1 i D.exit-sf-topology exit-service-family ! voice service saf profile trunk-route 1 session protocol sip interface Loopbackl transport tcp port 5060 ! profile dn-block 1 alias-prefix 1972555 pattern 1 type extension 4xxx ! profile callcontrol 1 dn-service trunk-route 1 dn-block 1 dn-block 2 i C. router eigrp SAF ! service-family ipv4 autonomous-system 1 ! topology base exit-sf-topology exit-service-family i .

D. E. which event should occur? A. the Device Mobility-related settings are also applied. To pass IP information from the CUCM to the endpoint C. Answer: A . The phone configuration is not modified. The roaming-sensitive parameters of the current (that is. The call will reroute via the PSTN with the constructed PSTN number as 442079460255. C. The user device profile line CSS combined with the device CSS of the physical phone used to log in the extension mobility user. C. The user device profile device CSS combined with the line CSS of the physical phone used to log in the extension mobility user.) A. The combined line/device CSS of the physical phone is used to log in the extension mobility user. To decode address information and route calls to and from the end points B. The call will fail because the ToDID is 0:. C. B. which CSS is used for calling privileges? A. D.Answer: A Q235Refer to the exhibit. Only the user device profile device CSS is used. To reside in the Cisco IOS software. the roaming) device pool are applied. Answer: BD Q238When Cisco Extension Mobility is implemented. The call will reroute via the PSTN with the constructed PSTN number as 00442079460255. B. The user-specific settings determine which location-specific settings are downloaded from the Cisco Unified Communications Manager device pool. The call will reroute via the PSTN with the constructed PSTN number as +442079460255. Answer: B Q236What is the purpose of a SAF Client? A. B. To learn about and advertise or subscribe information about SAF network services D. The call will fail because the called number will be 2079460255. and to communicate with the SAF forwarder Answer: C Q237Which two options for a Device Mobility-enabled IP phone are true? (Choose two. D. When a user presses a speed dial to +442079460255 when the SAF network is down. E. If the DMGs are the same. The combined line/device CSS of the user device profile.

If a user has more than one user device profile. Login will only be allowed to multiple profiles if the service parameter Allow Multiple Logins is enabled. C. D.Q239When multiple Cisco Extension Mobility profiles exist. The login will fail because only a single Cisco Extension Mobility profile is allowed. D. The audio source that is configured at the physical phone used for the Cisco Extension Mobility login is selected. The user must select the desired profile. choose an audio source from the User Hold MOH Audio Source drop-down list box from device profile configuration settings. how is the audio source for the MOH selected? A. the Enterprise user's mobile phone does not ring. Which CSS is responsible for ensuring that the correct partitions are accessed when calls are sent to the Enterprise user's mobile phone? . The audio source that is configured at the user device profile is selected. E. B. The user must login to both profiles in the order they are presented. The audio source that is configured at the home phone of the user is selected. a prompt displays on the phone and asks the user to choose a device profile for use with Cisco Extension Mobility. B. which actions take place when a user tries to log in to Cisco Extension Mobility? A. Q240When Cisco Extension Mobility is implemented. Q241Refer to the exhibit. The user may login to both profiles in any order. Answer: A Explanation: To specify the audio source that plays when a user initiates a hold action. The audio source that is configured in the IP Voice Media Streaming parameters is selected. when the PSTN phone calls the Enterprise user at extension 3001. With the Mobile Connect feature configured. C. Answer: B Explanation: Users access Cisco Extension Mobility by pressing the Services or Applications button on a Cisco Unified IP Phone and then entering login information in the form of a Cisco Unified Communications Manager UserID and a Personal Identification Number (PIN).

Cisco Unified Communications Manager determines how to route calls based on the remote destination number and the Rerouting Calling Search Space. the Phone Device CSS C. by verifying the visiting Trivial File Transfer Protocol Answer: E Q243Which two entities could be represented by device mobility groups? (Choose two. the Remote Destination Profile Rerouting CSS E. countries B. During the initial extension mobility login request. Q242You have been asked to deploy Cisco Extension Mobility Cross Cluster for a distributed call processing environment. by using a third-party automatic provisioning tool to verify user ID B. how does the visiting cluster determine if the user is a local user or a remote user? A.) A. the Phone Line (DN)CSS Answer: D Explanation: Ensure that the gateway that is configured for routing mobile calls is assigned to the partition that belongs to the Rerouting Calling Search Space. directory numbers D.A. the gateway CSS B. by verifying against the local database F. by using Extension Mobility Cross Cluster Session Initiation Protocol (SIP) trunks E. from user IDs that are created by default when the user logs in D. regions C. by broadcasting a request to all clusters to verify the user type C. the Remote Destination Profile CSS D. transcoders .

The phone creates a new device profile automatically. the Error 25 is displayed on the screen. The phone takes on the default clusterwide device profile. When a call between two HQ users is being conferenced with a remote user at BR. C. which configuration is needed? . subscribe device profile to EM phone service in case the enterprise subscription of EM Service is disabled Answer: BCD Q246Refer to the exhibit.Answer: AB Q244With Cisco Extension Mobility. when a user logs in to a phone type which has no user device profile. what will happen to the phone? A. when one of the end-users tries to login to the IP phone. associate EM Device profile with the end-user D. The phone immediately logs the user off. B. activate EM feature service under Cisco Unified Serviceability C. What three things should you do to resolve this issue? (Choose three. subscribe the MAC address of the IP Phone to EM Service E. update EM Phone Service URL to point to the publisher F. Answer: A Q245In a Centralized Call processing architecture. After the deployment of EM. The phone crashes and reboots. upgrade the firmware of the IP Phone to the latest version B.) A. D. you have deployed Extension Mobility (EM) feature.

729 codecs in Cisco Unified Communications Manager Service Parameters. B. D. The BR_MRG must contain the transcoder device. The BR_MRGL must be assigned to the BR phones. All HQ phones are configured to use HQ_MRGL and all BR phones are configured to use BR_MRGL. E. Enable the software conference bridge to support the G.A. For the HQ phones always to use the hardware conference bridge as a first choice. The HQ_MRGL must be assigned to the software conference bridge. The HQ_MRG must contain the transcoder device. A transcoder should be configured at the remote site and assigned to all remote phones through the BR_MRGL. The HQ_MRG must contain the transcoder device. The HQ_MRGL must be assigned to the HQ phones. Answer: D Q247Refer to the exhibit. which configuration should be implemented? . C.711 and G.

Add both the HQ_MRG and HQ_MRG_2 to the HQ_MRGL and list the HQ_MRG first.. Ensure that both the hardware and software conference bridges are listed in the HQ_MRG. how many calls are allowed for the BR location? . you need to have two separate MRG's and list the hardware MRG 1st in the MRGL . Ensure that both the hardware and software conference bridges are listed in the HQ_MRG. Ensure that the instance ID for the hardware conference bridge is 0. Add the HQ_MRG to HQ_MRGL. Assuming the regions configuration to BR only permits G. Answer: C Explanation: To ensure that the hardware bridge is utilized first with all its resources BEFORE the software bridge is used . The hardware conference bridge must be configured first. Q248Refer to the exhibit.. Add HQ_MRG_2 to HQ_MRGL_2. The HQ_MRGL_2 should be assigned to the HQ device pool. Configure a second HQ_MRG_2 and assign the software conference bridge to it. Assign the hardware conference bridge to HQ_MRG.A. Configure a second HQ_MRG_2 and assign the software conference bridge to it.729 codec. Assign the hardware conference bridge to HQ_MRG. C. The HQ_MRGL should be assigned to the HQ phones.. B. D. Configure an additional HQ_MRGL_2..

Total of four calls. Cisco Unified Communications Manager assumes that each G. Total of four calls in either direction. B.729 call stream consumes 24 kb/s amount of bandwidth. Q249Refer to the exhibit. Two outgoing calls. How many calls are permitted by the RSVP configuration? A. two incoming and two outgoing. Total of four calls to the BR location.A. Outgoing calls are not impacted by the location configuration. C. one G. Answer: D Explanation: In performing location bandwidth calculations for purposes of call admission control. Incoming calls are unlimited. E. Total of two calls in either direction.711 call . D.

The BR Cisco Unified Communications Manager has been configured for local RSVP. RSVP between the locations assigned to the IP phones and SIP trunks at each site are configured with mandatory RSVP.729 calls E. 128 Answer: D Q251Refer to the exhibit. four G.729 calls. 64 D.729 call and one G. The HQ Cisco Unified Communications Manager has been configured for end-to-end RSVP. 48 C.B.729 calls C. one G. 88 E. To permit three G.729 calls Answer: B Q250Refer to the exhibit. When a call is placed from the IP phone at HQ to the BR phone at the BR site.711 call D. 32 B. which statement is true? . eight G. what should the bandwidth value be for the ip rsvp bandwidth command? A. two G.

which supports loopback on RSVP reservation. B. The Cisco Unified Communications Manager at HQ will fall back to local RSVP and place the call.A. and that router is not running the latest IOS version. The BR Cisco Unified Communications Manager will use local RSVP. The call will fail. No RSVP end-to-end will occur. D. E. Q252Refer to the exhibit. The call will proceed as a normal call with no RSVP reservation. C. Make sure that the router is running the latest IOS version. RSVP end-to-end will occur. The Cisco Unified Communications Manager at HQ will use end-to-end RSVP. Which statement about the configuration between the Default and BR regions is true? . Answer: D Explanation: A possible cause is that the same router is being used as the calling and called RSVP agents.

Calls between the two regions can use only the G. B.323 Gatekeeper Call Admission Control is true? A.711 codec will be used. B. Q254Refer to the exhibit. Calls between the two regions can use either 64 kbps or 8 kbps. Gatekeeper Call Admission Control applies only to distributed Cisco Unified Communications Express deployments. Answer: B Explanation: in distributed call processing deployments on a simple hub-and-spoke topology. the G. Answer: B Q253Which statement about H. you can implement call admission control with a Cisco IOS gatekeeper. When lossy conditions are high. Cisco Unified Communications Manager Express (Unified CME). or an H.323 gateway) registers with the Cisco IOS gatekeeper and queries it each time the agent wants to place an IP WAN call. D. C. Gatekeeper Call Admission Control setting is configured in Cisco Unified Communications Manager. Only 64 kbps will be used between the two regions because the link is "lossy". D.A. In this design. Both codecs can be used depending on the loss statistics of the link. Gatekeeper Call Admission Control applies to distributed Cisco Unified Communications deployments. the call processing agent (which could be a Unified CM cluster. Gatekeeper Call Admission Control applies to centralized Cisco Unified Communications deployments that use locations based Call Admission Control. C. How many calls can be placed to Cluster B? .729 codec.

one G. Which command needs to be edited to allow the transcoder to register properly? . Answer: A Q255Refer to the exhibit.711 and three G.729 calls B.729 calls D. There is no limit for incoming calls to Cluster B. but the transcoder is failing to register with the Cisco Unified Communications Manager.729 calls. three G. Outgoing calls are limited to one G. You have configured transcoder resources in both an IOS router and a Cisco Unified Communications Manager.711 call C.A.711 and three G. When you review the configurations in both devices the IP addresses and transcoder names are correct. one G.

D. SRST E. The sccp ccm group number needs to match the associate ccm 2 command. E. Cisco Unified Communications Manager based Answer: AB Explanation: Location-based call admission control (CAC) manages WAN link bandwidth in Cisco Unified Communications Manager.1. The associate ccm 2 priority 1 command needs to be changed so the ccm value matches identifier 1 in the sccp ccm 10. Q256If your IP telephony administrator asks you to configure a local zone for your dial plan to control the volume of calls between two end points in a centralized multisite environment. gatekeeper based D. which two types of Call Admission Control can be implemented? (Choose two. B.A.) A. Answer: B Explanation: The value of the IP address should match the IP address in the ip source-address command. Automated alternate routing (AAR) provides a mechanism to reroute .1. C. automated alternate routing C. The sccp ccm group number must match the voice-card number. locations based B.1 command. The associate profile and dsp farm profile numbers need to match associate ccm 2 command. The maximum sessions command must match the number of codecs configured under the dsp farm profile.

Which two conditions can correct this issue? (Choose two. The called phone (San Jose) needs to have AAR Group value of AAR under the AAR Settings. B. With automated alternate routing. "Network Congestion Rerouting?" but AAR is otherwise transparent to the end user and works without user intervention. the caller does not need to hang up and redial the called party. Q258Refer to the exhibit. A user in RTP calls a phone in San Jose during congestion with Call Forward No Bandwidth (CFNB) configured to reach cell phone 4085550150. B. C. D. The user in RTP sees the message "Not Enough Bandwidth" on their phone and hears a fast busy tone. F. The calling phone (RTP) needs to have the AAR destination mask of 4085550150 configured under the AAR Settings.) A. Answer: BF Explanation: Automated alternate routing (AAR) provides a mechanism to reroute calls through the PSTN or . Q257Which statement is correct about AAR? A. AAR will display "not enough bandwidth" on the IP phone while it reroutes the call. The end users sees. AAR allows calls to be rerouted because of insufficient Cisco Unified Border Element controlled bandwidth to an ITSP.calls through the PSTN or other network by using an alternate number. The called phone (San Jose) needs to have the AAR destination mask of 4085550150 configured under the AAR Settings. C. The called phone (San Jose) needs to have the AAR destination mask of 914085550150 configured under the AAR Settings. E. Answer: A Explanation: Automated alternate routing (AAR) provides a mechanism to reroute calls through the PSTN or other network by using an alternate number when Cisco Unified Communications Manager blocks a call due to insufficient location bandwidth. D. AAR allows calls to be rerouted due to insufficient gatekeeper controlled IP WAN bandwidth. The calling phone (RTP) needs to have the AAR destination mask of 914085550150 configured under the AAR Settings. The calling phone (RTP) needs to have AAR Group value of AAR under the AAR Settings.

D. An RSVP agent is a Media Termination Point that the call has to flow through. G. D. Answer: C Q262What happens if location-based CAC is used and there is no bandwidth available when a remote caller is placed on hold? A. C. RSVP is topology aware. video. iSAC E. RSVP is configured in Cisco Unified Communications Manager independent of the ISR. the caller does not need to hang up and redial the called party.) A. B. Cisco Unified Communications Manager plays default MOH. Answer: A Q263On which Cisco Unified Communications Manager configuration parameter does the CODEC that a Cisco IP Phone uses for a call depend? . Answer: AD Explanation: The RSVP policy that is configured for a location pair overrides the default interlocation RSVP policy that configure in the Service Parameter Configuration window. Using RSVP for CAC simply allows admitting or denying calls based on a logical configuration that is ignoring the physical topology. B. RSVP supports audio. and data pass-through.722 B. GSM-FR F. iLBC Answer: C Q261How do RSVP-enabled locations differ from Cisco Unified Communications Manager locations? A. G. Cisco Unified Communications Manager sends TOH rather than MOH. G. You have an available dedicated bandwidth of 20% from the 2-Mb/s WAN circuit for VoIP that supports a maximum of 17 calls. With automated alternate routing. Cisco Unified Communications Manager attempts to reconnect the call immediately.711 C. C. RSVP enables AAR within Cisco Unified Communications Manager. RSVP is configured in the ISR independent of Cisco Unified Communications Manager. but only works with full mesh networks. Which codec do you configure in Cisco Unity Communications Manager to achieve this? A. RSVP is topology aware. C. E.729 D. Video data pass-through allows video and data packets to flow through RSVP agent and media termination point devices Q260You are deploying a remote office setup that connects with Cisco Unity Communications Manager at a hub location. Q259Which two statements describe RSVP-enabled locations-based CAC? (Choose two. RSVP can be enabled selectively between pairs of locations. Cisco Unified Communications Manager terminates the call. B. RSVP and RTP are used between the two endpoints. D.other network by using an alternate number when Cisco Unified Communications Manager blocks a call due to insufficient location bandwidth.

711 to site B but to use G. modify the device pool D.722 from being advertised in the cluster? (Choose two. what is the best CAC method recommended for this type of deployment? A. physical location D.323. gateway-based F. Cisco IOS software does not register transcoder resources. modify the line settings Answer: AB Q267Which action configures the registration of transcoder resources? A. Configure Cisco Unified Communications Manager regions. location-based C. to enable Cisco Unified Communications Manager to select the optimal single codec for endto. C. Configure Cisco Unified Communications Manager locations. which two are ways to prevent G. location Answer: D Q264In a Cisco Unified Communications Manager centralized call processing model. Answer: A Q266When you configure regions in a Cisco Unified Communications Manager multisite cluster. D. Answer: B Q268Which option describes the reason that transcoding resources are added in Cisco Unified Communications Manager? A. Cisco IOS software registers transcoder resources with H. D. RSVP-based D. to enable transcoding resources in a Cisco Unified Communications Manager server B.end calls C. to provide transcoding resources in Cisco IOS gateways to Cisco Unified Communications .729 to site C? A. Configure transcoder resources in Cisco Unified Communications Manager. modify the service parameter B. Configure a gatekeeper. to enable transcoding resources in Cisco IP Phones D. B.A.) A. Cisco IOS software registers transcoder resources with SIP. enterprise parameters B. region E. C. region-based E. QoS-based B. modify the enterprise parameter C. B. gatekeeper-based Answer: B Q265Which action configures phones in site A to use G. media resources C. Cisco IOS software registers transcoder resources with SCCP.

Configure Cisco Unified Communications Manager RSVP-enabled locations. and conferencing B. when you want to only use Cisco Unified Communications Manager resources Answer: C Q271Which two options are effective mechanisms to restrict the maximum number of voice calls on a WAN link? (Choose two. 8 kb/s for G.711. 64 kb/s for G. D. and 64 kb/s for G.711. and 8 kb/s for G. Configure Cisco Unified Communications Manager regions. 64 kb/s for G. when you need the ability to grow support by using DSPs D.729. D. Answer: BD Q272Which action configures CAC utilizing only Cisco Unified Communications Manager software? A. 64 kb/s for G. 8 kb/s for G. Configure Cisco Unified Communications Manager locations.729. and 64 kb/s for G. 8 kb/s for G. B.722 Answer: A Q274Which action configures AAR to route the calls that have been rejected by the gatekeeper . Answer: B Q270What is the purpose of configuring a hardware-based MTP when deploying Cisco Unified Communications Manager? A. Configure the branch office IP phones with regions. transfer. Configure a gatekeeper with an SIP trunk. Configure Cisco Unified Communications Manager MTPs. Configure the branch office IP phones with CSS and partitions. Configure the branch office IP phones with locations.722 B.711. 8 kb/s for G.722 C. Configure Cisco Unified Communications Manager regions. when you need support for up to 24 MTP sessions on the same server and 48 on a separate server C. Configure the branch office IP phones with MRGs and MRGLs.711. D. 64 kb/s for G. C. Configure Cisco Unified Communications Manager locations. and 16 kb/s for G.722 D.729. to allow for supplementary services such as hold. Configure a gatekeeper and a gatekeeper-controlled trunk in Cisco Unified Communications Manager with bandwidth control. Answer: B Q273Which bandwidth amounts are correct for configuring locations? A. B. C.Manager Answer: D Q269Which action configures transcoding resources in Cisco Unified Communications Manager to function with branch office Cisco IP Phones? A. B.729.) A. C.

B. Configure AAR to work with CTI route points. No adjustment to location setting is needed B.245 capabilities negotiation is completed. If both configurations exist. Answer: D Q275What is a prerequisite of AAR deployment? A.245 capabilities negotiation is completed. Configure AAR to work with SRST. C. Configure Cisco IP Phones for AAR. Immediately after the call setup message is received and the reservation message is received after H. Answer: B Q277How can the location setting be modified to resolve poor call quality? A. E. Regions are assigned directly in the device configuration page. The path and reservation messages are sent and received after the H. the device region configuration takes precedence. D. The path and reservation messages are sent and received immediately after the call setup message is received. C. Answer: C . B. Answer: E Q276How is the region assigned to a device such as an IP phone? A. Calls must be manually rerouted through the PSTN or other networks when Cisco Unified Communications Manager blocks a call due to insufficient location bandwidth. D. Regions can be assigned only through a device pool.CAC through the PSTN? A. This configuration is not possible using AAR. Clustering must be implemented over the WAN. Remove the audio bandwidth setting Answer: C Q278Cisco Unified border element is configured to support RSVP-based CAC. You must have a centralized call processing deployment. B. D. D. B. C. Calls must be automatically rerouted through the PSTN or other networks when Cisco Unified Communications Manager blocks a call due to insufficient location bandwidth. Regions can be assigned either directly on the device configuration page or through the device pool. You must have a single distributed call processing deployment. Regions can be assigned either directly on the device configuration page or through the device pool. When is the RSVP path and reservation message sent and received? A. the device pool region configuration takes precedence. The path is setup once the global command call rsvp-sync is configured. Mark the bandwidth between the locations as unlimited C. Decrease the audio bandwidth setting D. If both configurations exist. C.

D. Third-party endpoints are not compatible with VCS Control. B. B. C. Verify that all phones are registered to a second subscriber server. C.) A. 10 E. + . The engineer has confirmed that no traffic is being blocked for the endpoint and it is receiving a valid IP address. C. An incorrect SIP domain is configured on the VCS Control for the endpoint. 1 B. D. E. Confirm that SRST is configured on the voice gateway. Answer: ABC Q283Which symbol is required for globalized call routing? A. F. 15 F. Which two actions must you take to troubleshoot the problem? (Choose two. Check the Region settings in Cisco Unified Communications Manager. Verify that media resources fail over to a secondary subscriber server when the publisher fails. Answer: AB Q282Which three tests can you perform to verify redundancy in the customer environment? (Choose three.323 redundant connection is active. Which option could be the cause of this registration failure? A. Verify that the H. Confirm that a calling search space is assigned to the voice gateway in Cisco Unified Communications Manager. Verify that Cisco Unified IP phones running SCCP go into SRST mode when the WAN connection is disconnected. Confirm that the site has an SRST reference that is correctly associated with the Cisco Unified Communications Manager group. unlimited Answer: A Q281You recently implemented call redundancy at a new remote site.) A. 5 D. The VCS Control must be deployed together with VCS Expressway before endpoints can register to either one. only with VCS Expressway.Q279Company A has deployed a VCS Control and is attempting to register a third-party endpoint. Cisco Unified Communications Manager is required in addition to the VCS Control. Verify that HSRP is active on the Cisco Unified Communications Manager subscriber servers. Confirm that the site devices are associated with a Cisco Unified Communications Manager group and that four Cisco Unified Communications Manager servers are available. and users report that calls are dropped when the remote site supposedly is in SRST. E. Answer: C Q280How many active gatekeepers can you define in a local zone? A. Verify that SCCP fallback is configured in Cisco Unified Communications Manager. 2 C. F. Restart Cisco Unified Communications Manager services to confirm that the server is working correctly. D. B.

Configure a phone NTP reference. D. The phone reboots with an error. C. Video endpoints inside Company X have registered but are unable to receive calls from outside endpoints. D. MoH cannot be provided for the remote sites. it is loaded. Configure voice register pool. Answer: B Q287The network administrator at Enterprise X is creating the guidelines for a new IPT deployment consisting of a large number of remote offices. The user cannot log in. / E. The traversal zone on the VCS Control does not have a search rule configured. Configure telephony service. B. D. Every user within Enterprise X is assigned a directory number of 5 digits. To facilitate external calls. Which option might cause an issue in a multisite deployment? A. C.) .) A. Answer: A Q284Which three steps configure Cisco Unified Survivable Remote Site Telephony for SIP phones? (Choose three. B. The maximum number of IP phones are in use at each remote site. B. Which option could be the cause of this issue? A. . Answer: D Q286When considering Extension Mobility. what happens if a user logs into a phone for which the user does not have a user device profile? A. Configure the SIP registrar. C. Another user device profile is loaded. All media streams are necessarily routed through the central office for calls to establish correctly. F. Answer: BCE Q285Company X has deployed a VCS Control with a local zone and a traversal client zone. * C. E. Configure voice register global dn. The access control list on the VCS Control must be updated with the IP for the external users. D. Answer: A Q288Which three options are supplementary services that are affected by MTP? (Choose three. C. Configure an SRST reference. % D. Overlapping DID ranges are allocated to each site. B. The local zone on the VCS Control does not have a search rule configured. If a default device profile for this phone has been configured.B. When a traversal zone is set up on VCS Control only outbound calls are possible. VCS Expressway is deployed and traversal server zone is set up there.

) A. No Media Termination Point Required B.) A.323 registration? A. Call Back Answer: ABC Q289Which two configurable options are available to enable Early Offer for calls over a Cisco Unified Communications Manager SIP trunk? (Choose two. Shadow Answer: A Q291In the case of a VCS cluster. MRGL B. DNS SRV records pointing to the VCS cluster name B. Call Park D. Speed Dial F. network locale D. LBM Hub D. codec E. Use Trusted Relay Point Answer: BD Q290Which option best describes a service that assembles a network model from configured locations and link data in one or more clusters? A.) A.323 gateway with Cisco Unified Communications Manager? (Choose two. Early Offer support for voice and video calls Mandatory (insert MTP if needed) E. hostname of the VCS cluster configuration master D. hostname of the VCS cluster member peer Answer: A Q292Which three options are overlapping parameters for roaming when a device is configured for Device Mobility? (Choose three. Media Termination Point Required C. device pool Answer: ABC Q293Which two options are requirements for deploying an H.323 gateway must be configured use the . location C. extension F. Accept Audio Codec Preferences in Received Offer D. Call Pickup E. which configuration element is recommended for endpoint H. Call Hold B. Cisco Unified Communications Manager and the H.A. Weight C. static IP addresses C. Call Transfer C. LBM B.

g722/g711 for intraregion calling . as a suffix.) A. C. Answer: AC Q296When configuring a video ISDN gateway. g729 codec for interregion calling D. RTMT B.323 instead of SIP.Set up end users accounts for the users who need to roam . E. D. D. C. Use H.same TCP port for H. Which two configurations are required in the Cisco Unified Communications Manager regions to provide the most suitable use of bandwidth while preserving the call quality? (Choose two. Extension Mobility has not been enabled under Enterprise Parameters. phone menu C. B. C. The H. as a suffix.Set up a device profile for the type of phones users will be allowed to log in Users have reported to the administrator that they are unable to log in to the phones designated for Extension Mobility. otherwise they cannot log in elsewhere. The username must be numeric only and must match the DN. E.323 calls.245TCSTimeout timer must be set to at least 25. The user must ensure that their main endpoint is online and registered.) A. Cisco Unified Serviceability D. which two actions are requirements for the Cisco Preferred Architecture for Enterprise Collaboration? (Choose two. Perform dial string manipulation on Cisco Unified Communications Manager. Which two options are the two reasons for this issue? (Choose two.323. The Media Exchange Interface Capability Timer must be set to less than 20. Cisco voicemail ports must be active. E. Answer: AB Q294In which two locations can you verify that a phone has a standby Cisco Unified Communications Manager? (Choose two. Use SIP instead of H.) A. g722/g711 codec for interregion calling C. g729 codec for intraregion calling B. Use an ! (exclamation point) at the end of each ISDN number.) A. D. B. Use an * (asterisk) at the end of each ISDN number. B. The user device profile is not associated to the correct end user. phone webpage Answer: BD Q295The administrator at Company X is trying to set up Extension Mobility and has done these steps: . Answer: AB Q297Company X has three locations connected via a low bandwidth WAN. The Media Exchange Timer must be set to less than 20. The Extension Mobility service has not been enabled under the Cisco Unified Serviceability Page.

Cisco IM and P D. Allow access to TCP port 1720. Accept out-of-dialog refer D. Enable application-level authorization B. Block access to TCP ports 2427 and 2428. Expressway-C B. g729 codec for all calling F. Select the SRTP Allowed check box on the SIP trunk. Allow access to TCP port 2428. Cisco Unified Communications Manager C. VCS Control Answer: D Q302Refer to the exhibit. Enable the Media Termination Point Required option on the SIP trunk. B. F.E. Which option describes the effect of this configuration? . Execute the isdn switch-type primary-ni command globally. Enable the Early Offer Support for Voice and Video Calls option on the SIP profile. C. Which firewall and ACL configuration must you perform to allow the MCGP gateways to function correctly? A. g722/g711 codec for all calling Answer: CD Q298Which two options must be selected in the SIP Trunk Security Profile configuration between Cisco Unified Communications Manager and Expressway? (Choose two) A. Select the Enable Inbound FastStart check box on the Cisco Unified Communications Manager servers. Answer: A Q301Which device must be accessible from the public Internet in a Collaboration Edge environment? A. C. Expressway-E E. E. Accept presence subscription C. Transmit security status G. Allow charging header Answer: DE Q299Which two configurations can you perform to allow Cisco Unified Communications Manager SIP trunks to send an offer in the INVITE? (Choose two. D. Answer: AB Q300You are deploying a Cisco Unified Communications Manager solution with MGCP gateways at multiple locations. Block TCP port 1720. Accept replaces header F. B. Open access to all TCP and UDP ports. Accept unsolicited notification E. E. D. Select the Display IE Delivery check box in the gateway configuration.) A.

It creates dial peers.) A. E. Expressway-C to Expressway-E C. It implements HSRP. The Cisco Unified Communications Manager trunk configuration must have the destination port set to 5061. A SIP trunk security profile must be configured with Device Security Mode set to TLS. The root CA of the VCS server certificate must be loaded in Cisco Unified Communications Manager. Answer: ACE Q305Which three options describe the main functions of SAF Clients? (Choose three.A. It implements Cisco IOS redundancy. A SIP trunk security profile must be configured with the X. Expressway-E to Cisco Unified Communications Manager E. The Cisco Unified Communications Manager zone configured in VCS must have SIP authentication trust mode set to On.) . It configures a standby Cisco Unified CME. E. It configures failover. A SIP trunk security profile must be configured with Incoming Transport Type set to TCP+UDP. F. F. C. B.) A. G. Expressway-C to internal endpoint Answer: BC Q304Which three statements about configuring an encrypted trunk between Cisco TelePresence Video Communication Server and Cisco Unified Communications Manager are true? (Choose three. The Cisco Unified Communications Manager zone configured in VCS must have TLS verify mode set to Off. Expressway-E to outside-located endpoint D. C. Answer: A Q303On which two call legs is the media encryption enforced in a Collaboration Edge design? (Choose two. D. It implements Cisco United CME redundancy. D. B. Expressway-C to Cisco Unified Communications Manager B.509 Subject Name from the VCS certificate.

saf-forwader.asf@domain. 30 ms Jitter. integrating with Cisco IM and Presence for additional services Answer: ABC Q306Which option indicates the best QoS parameters for interactive video? A. Audio Codec B. They are generated in the format data-source:sub-service:instance.instance. E. 30 ms Jitter.323 and SIP endpoints. They are generated in the format data. 160 ms One-way Latency.local. They are generated in the format service:sub. translation patterns Answer: ABC Q309Which two statements about SAF service identifier numbers are true? (Choose two. Link Loss Type D.) A.data. 20% Overprovisioning C. Which component allows for standardized caller addresses between the endpoints? . registering Cisco Unified Communications Manager subscribers with the publisher E. 5% Max Loss. hosted DN patterns D. 100 ms One-way Latency.instance. route patterns E. 60 ms Jitter.matrix.) A.) A. They are 32-bit decimal identifiers. starting Cisco Unified Communications Manager services throughout the cluster F.local. They are 16-bit decimal identifiers.db.saf. They are generated in the format telco. 5 s One-way Latency. subscribing to SAF network services D. Location Description E. 1% Max Loss. 20% Overprovisioning B. Real Time Protocol Answer: ABC Q308Which three items must you configure to enable SAF Call Control Discovery? (Choose three. 30 ms Jitter. 10% Overprovisioning Answer: A Q307Which three configuration settings are included in a default region configuration? (Choose three.instance. F. Video Call Bandwidth C. 0% Max Loss.323 trunk B. 150 ms One-way Latency.saf.replicate. the SIP or H.A. Answer: AB Q310A voice engineer is enabling video capabilities between H. Immersive Bandwidth F.cucm-publisher. B.service:instance. hosted DN groups C. a calling search space F. registering the router as a client with the SAF network B.cisco.fifty. 20% Overprovisioning D. providing publishing services to the SAF network C. 1% Max Loss. C. D.

C. F. The backup node must be shut down first to allow the endpoints to realize that the primary node is online again. What happens when the primary node comes back online? A. the endpoints only failover when the node they lose connection to their registered node. transform Answer: D Q311Which two statements about Cisco Unified Communications Manager Extension Mobility are true? (Choose two. After an autogenerated device profile is created. Configure SIP trunks between Cisco Unified Communications Manager clusters. Devices can be configured to allow more than one user to be logged in at the same time. D. C. Answer: CD Q312When you configure a globalized dial plan. A device profile has most of the same attributes as a physical device.323 gateways. policy service D. Configure the called-party transformation settings for incoming calls on H. 1 D.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/8x/uc8x/dialplan. search rules B. Answer: ABF Explanation: http://www. Configure a hunt group. E.cisco. D. Answer: B Q315Company X has deployed a VCS Control with a local zone and a traversal client zone. C. 2 Answer: D Q314Company X has a primary and a backup Cisco Unified Communications Manager instance. To . E. B. B. An autogenerated device profiles can be loaded on a device at the same time as a user profile. A device can adopt a user profile even when no user is logged in. SIP route pattern C. Configure a remote site device pool. 4 B.A. Configure translation patterns in the partitions used by the gateway calling search space. B. The primary node becomes the backup node. which resulted in a failover to the backup node. The administrator had to do maintenance on the primary node and did a shutdown. Endpoints detect that the primary is back and reregisters automatically. you can associate it with one or more users. Nothing. 3 C. D.) A. in which three ways can you enable ingress gateways to process calls? (Choose three.html#wp11531 Q313How many nodes can a phone establish a connection to at the same time? A.) A. Configure the gateway with prefix digits to add necessary country and region codes.

MX-200 Answer: AB Q318Refer to the exhibit. Cisco Jabber Desktop C. C. one audio codec E. An engineer receives a ticket to troubleshoot a one-way audio issue with these symptoms: .facilitate external calls. ref2833 D. PVDM or DSP resource B. however user cannot hear user . D. VCS Expressway is deployed and traversal server zone is set up there.User A can hear user and vice versa.User can heat user C. LTI local transcode resource C. When a traversal zone is set up on VCS Control only outbound calls are possible.User A can hear user C. Answer: A Q316Which two options are requirements for hardware MTP on Cisco IOS routers? (Choose two. T1 PRI card Answer: AB Q317The Cisco Unified Communications system of a company has five types of devices: ·Cisco Jabber Desktop ·CP-7965 ·DX-650 ·EX-60 ·MX-200 Which two types of devices are affected when an engineer changes the DSCP for Video Calls service parameter? (Choose two. .) A. but are unable to receive calls from outside endpoints. The traversal zone on the VCS Control does not have a search rule configured.) . The local zone on the VCS Control does not have a search rule configured. Video endpoints inside Company X have registered. CP-7965 D. DX-650 B.) A. B. EX-60 E. Which two properties are the most likely reasons for this issue? (Choose two. . The access control list on the VCS Control must be updated with the IP for the external users. however user cannot hear user A. Which option could be the cause of this issue? A.

call-started F. Configure a phone NTP reference. registration Answer: F Q320Which three steps configure Cisco Unified Survivable Remote Site Telephony for SIP phones? (Choose three. end-call D. start-call B.com/c/en/us/support/docs/voice/voice-quality/5219-fix-1way-voice.A. The Cisco EX60 of User C is not responding to requests coming from the TMS server. Configure voice register global dn. Answer: BE Explanation: http://www. Configure voice register pool. B. The Cisco EX60 default gateway of User C is missing from the network configuration. The NAT device is allowing only RTP/RTCP ports from the internal network to the DMZ. call-ended E.cisco. E. D. The Cisco VCS Expressway is not responding to the SIP INVITE coming from the Cisco VCS Control. Configure an SRST reference. C. C. . D. in-call C. B.) A. The router does not have a route back from the DMZ to the internal network.html Q319Which message does a Cisco VCS use to monitor the Presence status of endpoints? A.

Still. transforms C. Call Policy could be justified as the correct answer based on how the question is interpreted. I would never use Call Policy in this capacity. access rules D. Which type of gateway or trunk on Cisco Unified Communication Manager for the Cisco VG310 must the administrator set up to allow the phones to have the call pickup feature? A.cisco.com/document/77096/device-mobility Q322An engineer is working on a Cisco VCS Control routing configuration and wants users to be able to dial ccnpcollab and have calls routed to ccnpcollab@cisco. MGCP gateway E. https://supportforums.323 gateway B. Answer: ACE Q321Which three options are overlapping parameters for roaming when a device is configured for Device Mobility? (Choose three.E.225 trunk D. and Network Locale. Configure telephony service. H. location C. H. F. extension Answer: BCE Explanation: The overlapping parameters for roaming-sensitive settings are Media Resource Group List. nor does Cisco recommend this.com. SCCP gateway C. network locale D. call policy Answer: B Explanation: Although Call Policy could be used. and not for other aliases that might be dialed without a domain. typically this type of alias change is done using transforms (either pre-search transforms or search rule transforms). The overlapping parameters for the Device Mobility-related settings are Calling Search Space (called Device Mobility Calling Search Space at the device pool). then call policy makes since. SIP trunk Answer: B Explanation: . MRGL F. codec E. Q323An administrator is setting up analog phones that connect to a Cisco VG310. Overlapping parameters configured at the phone have higher priority than settings at the home device pool and lower priority than settings at the roaming device pool. Configure the SIP registrar. Which option achieves this aim? A. If this change is only intended for the ccnpcollab alias being dialed. search rules B.) A. device pool B. and AAR Calling Search Space. AAR Group. Location.

2.263 E. H. BFCP Answer: E Q325An engineer must resolve a "VIDEO" call failure issue. Which protocol must be enabled in SIP profile for VCS SIP trunk on Cisco Unified Communications Manager? A. Answer: B Q327An engineer is performing an international multisite deployment and wants to create an effective backup method to access TEHO destinations in case the call limit triggers. Event="Authentication Failed" Service="SIP" Src-ip="10. RDP B.http://www. When using RTMT. AAR . lack of audio or video bandwidth E. E. Which option is the cause of the call failure? A. Which technology provides this ability? A.264 C. The Expressway-C Traversal Zone username/password do not match the Expressway-E Traversal Client username/password.html Q324An engineer must enable video desktop sharing between a Cisco Unified Communications Manager registered video endpoint and a Cisco VCS registered video endpoint. H. H. lack of transcoding resources D. Expressway-C. The Expressway-C Traversal Zone username/password do not match the Expressway-E Traversal Zone username/password.com/c/en/us/products/collateral/unified-communications/vg-seriesgateways/product_data_sheet09186a00801d87f6.224 D. lack of video bandwidth C. The Expressway-C Traversal Client username/password do not match the Expressway-E Traversal Server username/password.50. The Expressway-C Traversal Server username/password do not match the Expressway-E Traversal Zone username/password. C. the engineer notices that the Location Bandwidth Manager-OutOfResources counter is showing a positive value. B. D. lack of audio bandwidth B. lack of conferencing resources Answer: B Q326While troubleshooting a connectivity issue between Cisco Unified Communications Manager. an engineer sees this output in the Expressway-E logs. and Expressway-E.cisco.1" Src-port="25723" Detail="Incorrect authentication credential for user" Protocol "TLS" Method="OPTIONS" Level="1" What is the cause of this issue? A. The Expressway-C Traversal Server username/password do not match the Expressway-E Traversal Client username/password.

CFUR C.) A. when you require encrypted calls to endpoints on your corporate LAN C. and media encryption policies.B. Ensure FQDN is used in SIP Request header. D. when you require administrative access to the Cisco Expressway Edge from the Internet Answer: D Explanation: VCS/Expressway servers only use port 443 for administrative access. Ensure FQDN is used in SIP Identity header. authentication. Manage bandwidth to restrict standard definition endpoints from using more than 2 Mb of bandwidth. Traverse a firewall from a protected network to a public or DMZ network. No call session of any sort on a VCS or Expressway use port 443. Answer: D Q329Which two options are functionalities of subzones in a Cisco VCS deployment? (Choose two. E.com/c/dam/en/us/td/docs/telepresence/infrastructure/vcs/config_guide/X81/Mobile-Remote-Access-via-VCS-Deployment-Guide-X8-1-1. Apply registration. when video endpoints that reside on the Internet require administrative access to the Cisco Expressway Edge B. when you want to enable calls to web applications by using HTTP D. Also. Resolve names outside of the direct control of the Cisco VCS that exist on the public Internet. Resolve FQDN using DNS type SRV record. the "Unified Communications Mobile and Remote Access via Cisco VCS Deployment Guide" identified port 443 as the administrative access port from pubic internet to VCS Expressway.cisco. Which result is achieved by enabling this option? A. B. http://www.pdf . C. C. Connect to another Cisco VCS on the same side of the firewall to extend dialing capabilities. B. Answer: DE Q330Which situation requires TCP port 443 to be open for packets that are sourced from the Internet with a destination in the corporate DMZ? A. D. Resolve FQDN using DNS type A record. SRST Answer: C Q328An engineer is configuring a SIP profile for Cisco VCS SIP trunk on Cisco Unified Communications Manager and enables the option "Use Fully Qualified Domain Name" in SIP Requests. LRG D.

It configures failover.225 trunks E. CO trunks . C. D. It creates dial peers. F. Which option describes the effect of this configuration? A. H. SIP trunks D.) A. E. Answer: A Q332Which two types of trunks can support Cisco Unified Communications Manager? (Choose two. It implements HSRP. B. switch port trunks B. It configures a standby Cisco Unified E. It implements Cisco United CME redundancy.Q331Refer to the exhibit. It implements Cisco IOS redundancy. PIMG trunks C.

160 ms One-way Latency.com/c/en/us/td/docs/solutions/Enterprise/WAN_and_MAN/QoS_SRND/QoSSRND-Book/QoSIntro. 5% Max Loss.F. C. 1% Max Loss.) A. Verify that media resources fail over to a secondary subscriber server when the publisher fails. 0% Max Loss. D. the SIP or H.323 redundant connection is active. 1% Max Loss. 30 ms Jitter. B.) A. Answer: ABE Q334Which option indicates the best QoS parameters for interactive video? A. a calling search space B.cisco. 100 ms One-way Latency. hosted DN patterns C. E. 150 ms One-way Latency.html#wp1098521 Q333Which three tests can you perform to verify redundancy in the customer environment? (Choose three. 60 ms Jitter. Verify that all phones are registered to a second subscriber server. translation patterns D. route patterns E. 20% Overprovisioning Answer: C Explanation: http://www. Verify that HSRP is active on the Cisco Unified Communications Manager subscriber servers. POTS trunks Answer: CD Explanation: http://www. a Cisco Jabber Desktop .html (interactive video) Q335Which three items must you configure to enable SAF Call Control Discovery? (Choose three.323 trunk F.cisco. 30 ms Jitter. hosted DN groups Answer: BEF Q336The Cisco Unified Communications system of a company has five types of devices: ·Cisco Jabber Desktop ·CP-7965 ·DX-650 ·EX-60 ·MX-200 Which two types of devices are affected when an engineer changes the DSCP for TelePresence Calls service parameter? (Choose two.) A.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/8_5_1/ccmsys/accm-851cm/a08trnk. 20% Overprovisioning B. 20% Overprovisioning D. 10% Overprovisioning C. F. 30 ms Jitter. Verify that SCCP fallback is configured in Cisco Unified Communications Manager. Verify that Cisco Unified IP phones running SCCP go into SRST mode when the WAN connection is disconnected. Verify that the H. 5 s One-way Latency.

MX-200 E. SIP trunks E. DX-650 C.html#CUCM_TK_BDDD9061_00 . Answer: B Explanation: GDPR using ILS allows for blocking learned patterns to be configured in the CUCM. border controllers D. Which action accomplishes this task? A. EX-60 Answer: DE Q337A presales engineer is working on a quote for a major customer and must evaluate how many Cisco VCS Expressway traversal call licenses for which to plan. B. Cisco 9971 Endpoint C. SIP trunk F. Create a block transformation pattern. TAPI applications G. Create a block route pattern. a software conference bridge B. an annunciator D.) A.) A. gatekeeper E.cisco. gateway B. Create a block learned pattern. C. Calls to and from which three routes must the engineer include in the tally? (Choose three.323 endpoints Answer: ABD Q339An engineer is configuring Global Dial Plan Replication and wants to prevent the local cluster from routing the Vice President number 5555555555 to the remote cluster. CTI applications H. http://www.B. Create a block translation pattern. third-party H. CP-7965 D. Cisco Unified IP Phone 7962G C. JTAPI applications F.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/10_0_1/ccmfeat/CUCM_BK_F 3AC1C0F_00_cucm-features-services-guide-100/CUCM_BK_F3AC1C0F_00_cucm-featuresservices-guide-100_chapter_011101. D. VCS Answer: CDF Q338Which three devices or applications support call preservation? (Choose three.

H. Associate the directory URIs to directory numbers. The bandwidth settings of the site are fulfilling on-net call volume.) A.Q340An administrator is visiting a remote site that has on-net calls with headquarters and one voice gateway for PSTN calls. D. Load-balance PRI connections. Assign directory URIs to users. Load-balance route lists within the cluster. Configure the URI service parameters. F. Implement SIP to POTS. The LBM service is malfunctioning. Configure the directory URI partition and calling search space. C. B. Implement ICT trunks to remote locations. Answer: FG Q342Which two steps must you take when implementing TEHO in your environment? (Choose two. Answer: AF . Implement centralized failover. Which description about this issue is true? A.) A. F. Activate the URI service in Cisco Unified Serviceability. Configure SIP route patterns. C. D. B. B. AAR is routing some of the calls. Configure SIP trunk. G. E. but no call failure reports have been sent from this site. When using RTMT to monitor the bandwidth utilization of the remote site. E. C. Answer: B Q341An engineer is configuring URI calling within the same cluster. The location-based CAC does not work properly. D. Configure the SIP profile. Which two actions must be taken to accomplish this configuration? (Choose two. Implement local failover. the administrator notices the OutOfResources counter for the site in LBM has been increasing slowly in last two weeks.

CER stands for Cisco Emergency Responder and AAR stands for Automated Alternate Routing. B. AAR E. E. MGCP Answer: B Q346Which three globalization dialing functions are enhanced in Cisco Unified Communications Manager 7.) A.Q343You have deployed a Cisco 2821 ISR to perform as an SRST voice gateway at a remote site. show voice register session-server C. The site has exceeded the number of simultaneous calls allowed in SRST mode. MGRL B. DTMF inband RTP-NTE (rfc2833) D. TEHO C. Answer: AC Q344Which two functions can be implemented without MTP resources? A. Sip delay offer E. Some Phones at the remote site are assigned a device pool without SRST reference. which signaling protocol is used between peers to determine the best route for calls? A. show ccm-manager hosts . Multicast MOH Answer: DE Q345An engineer is setting up a Cisco VCS Cluster with SIP endpoints only. These are the correct answers according to the Cisco Collaboration System SRND. some of the phones located at the remote site are unable to make phone calls. IPV6 -IPV4 transform C. SIP B. Which two options are potential causes of the problem? (Choose two.) A.323 C.323 fast start B. D. The ccm-manager fallback-mgcp command is configured incorrectly on the voice gateway. The site has exceeded the number of SRST endpoints supported by the voice gateway. While configuring the Cisco VCS peers. CER D. h. C.x and later? (Choose three. The ccm-manager fallback command is configured incorrectly on the voice gateway. Q347Which two commands verify Cisco IP Phone registration? (Choose two. SCCP D. click-to-call Answer: BCD Explanation: TEHO stands for Tail End Hop Off. show ephone registered D. During a network failure between the remote site and the central office. SAF F.) A. H. show telephony-service ephone-dn B.

Cisco Unified Communications Manager IP address Answer: E Q349After forgetting to log out of his IP phone in the main office. NTP server B.E. The user's Extension Mobility profile is misconfigured. During troubleshooting. SRST with MGCP fallback D. gatekeeper . Which option must be configured to complete the Cisco VCS Expressway system configuration? A. The phone at the remote location is a different model than the phone in the user's main office. B. Cisco Unified Communications Manager Express in SRST mode Answer: D Q351An engineer has configured a Cisco EX60 to register with a Cisco VCS-C. LDAP server D. D. An engineer is deploying a new Cisco VCS Expressway for a company and has configured the IP address and the system name. SIP server C. SRST without MGCP fallback B. SRST with VoIP dial peers to Cisco Unified Communications Manager Express C. but the device is not showing up as registered. the engineer sees this output. Answer: C Q350Which solution is needed to enable presence and extension mobility to branch office phones during a WAN failure? A. DNS server F. The device pool is misconfigured. After logging into the Cisco VCS Expressway admin page. which component will the engineer likely find missing in the configuration? A. Which option is a possible reason for the problem? A. an Extension Mobility user is unable to log in to a different IP phone at a remote office. The user can log in to only one device at a time. DHCP server E. show sip-ua status registrar Answer: CE Q348Refer to the exhibit. C.

service parameters D. MCU C. enterprise parameters Answer: C Q353Your company's main number is 408-526-7209.digit numbers. Immersive Bandwidth B. translation pattern D. Video Call Bandwidth C. and your employee's directory numbers are 4. enterprise phone configuration B. Link Loss Type E. DNS Answer: A Q352An engineer is configuring a new DX-80 in Cisco Unified Communications Manager. Audio Codec D.html#wp1077135 Q355Refer to the exhibit. calling party transformation pattern B. AAR group C. Location Description Answer: BCD Explanation: http://www. Where can an engineer verify the default DSCP value of AF41? A. route pattern Answer: A Q354Which three configuration settings are included in a default region configuration? (Choose three. common phone profile C. Real Time Protocol F.B. Which statement about when user A calls user using SIP is true? .cisco. default gateway D. Which option should be configured if you want outgoing calls from 4-digit internal directory number to be presented as a 10-digit number? A.) A. TMS E.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/8_0_1/ccmcfg/bccm-801cm/b02regio.

State : Disabled A.com Answer: C Q357What are the tasks required to route calls from H323 to SIP and viceversa? (Choose two.... Pattern type : Regex Pattern string : @cisco.. SIP TCP/TLS ports must be opened from internal to DMZ and vice versa. There are three fields populated with "CCNPCOLAB".. Replace string : ccnpcollab Target : . Sent as CCNPCOLAB@cisco. Can not route call B.. Deploying a Cisco VCS Control inside a NAT mandates the use of the Advanced Networking option key.com" Mode : .) A.com C... Cisco VCS Control and Cisco VCS Expressway support static NAT.A.. Sent to Cisco. C. sent as CCNPCOLAB D.. RT and RTCP ports must be opened at the firewall from internal to DMZ and vice versa. Answer: A Q356An engineer is troubleshooting a dial path etc" and it gives a config snapshot...com Pattern Behaviour :. B. D.. "Add Suffix" and "@Cisco.. Config-protocols-Interworking-On .

Config-protocols-H323-H323Mode-On Answer: AE . Config-protocols-Interworking-registered only D.B. Config-protocols-Sip-Config-Mode-On E. Config-protocols-Interworking-Off C.