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QUESTION 1
Which statement is true when considering a Cisco VoIP environment for regional configuration?
A. G.711 requires 128K of bandwidth per call.
B. G.729 requires 24K of bandwidth per call.
C. The default codec does not matter if you have defined a hardware MTP in your Cisco Unified Communications Manager environment.
D. To deploy a Cisco H.323 gatekeeper, you must configure MTP resources on the gatekeeper and only use G.711 between regions.
Correct Answer: C
Explanation

 

QUESTION 2
Which 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
Correct Answer: BE
Explanation

 

QUESTION 3
On which Cisco Unified Communications Manager configuration parameter does the CODEC that a Cisco IP Phone uses for a call depend?
A. enterprise parameters
B. media resources
C. physical location
D. region
E. location
Correct Answer: D
Explanation

 

QUESTION 4
Which two formats are supported by Cisco Unified Communications Manager to stream music on hold? (Choose two.)
A. G.711
B. G.729
C. G.723
D. G.722
E. G.728
F. Advanced Audio Codec
Correct Answer: AB
Explanation

 

QUESTION 5
Which three of the following are steps in configuring MGCP Fallback and Cisco Unified SRST? (Choose three)
A. Define the SRST reference for phones in the Device Pool configuration
B. Enable and configure the MGCP fallback and Cisco Unified SRST features on the IOS gateways.
C. Implement a simplified SRST dial plan on the remote-site-gateways to ensure connectivity for remote-site phones in SRST mode.
D. Enable SIP trunking between both remote and hub sites to provide mesh coverage.
E. Define the SRST reference in the configuration on the IP Phones.
F. Enable and configure the MGCP fallback on the IOS gateway but not Cisco Unified SRST since it is enabled automatically.
Correct Answer: ABC
Explanation

 

QUESTION 6
Cisco Unified Communications Manager is configured with CAC for a maximum of 10 voice calls.
Which action routes the 11th call through the PSTN?
A. Configure an SIP trunk to the ISR.
B. Configure Cisco Unified Communications Manager AAR.
C. Configure Cisco Unified Communications Manager RSVP-enabled locations.
D. Configure Cisco Unified Communications Manager locations.
Correct Answer: B
Explanation

 

QUESTION 7
Which option describes the reason that transcoding resources are added in Cisco Unified Communications Manager?
A. to enable transcoding resources in a Cisco Unified Communications Manager server
B. to enable Cisco Unified Communications Manager to select the optimal single codec for end-to-end calls
C. to enable transcoding resources in Cisco IP Phones
D. to provide transcoding resources in Cisco IOS gateways to Cisco Unified Communications Manager
Correct Answer: D
Explanation

 

QUESTION 8
The 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?pass4itsure 300-075 exam question

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
Correct Answer: A
Explanation
Explanation/Reference:
Explanation: Incorrect answer: BCDE
Destination pattern is 91, HQ office area code is 201 .

 

QUESTION 9
An engineer implemented a VCS as a gateway to allow SIP and H323 endpoints to rregister, what recommended protocol or gateway should be used.A. SIP trunk gatekeeper
B. H323 gateway
C. H321 gateway-based
D. SIP
Correct Answer: B
Explanation

 

QUESTION 10
Enabling authentication and encryption for CTI, JTAPI, and TAPI applications requires which two tasks? (Choose two.) (SourcE. Configuring Cisco IP Telephony Authentication and Encryption)
A. Enter the encryption key into the application.
B. Set up an IPsec association between the application and Cisco Unified CallManager.
C. Configure related security parameters in the CTI, JTAPI, and TAPI application.
D. Add the application user or end users to the Standard CTI Secure Connection user group, Standard CTI Allow Reception of SRTP Key Material user group, and Standard CTI Enabled user group.
Correct Answer: CD
Explanation
Explanation/Reference:
Explanation: Incorrect answer: AB
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. After you add the user to this group and
install the certificate, the application ensures that the user connects via the TLS port.
Link:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/security/8_6_1/secugd/secucti.htm l#wp1166397

 

QUESTION 11
When device mobility mode is enabled or disabled for a cluster, to which does the cluster setting apply?
A. all phones in the cluster that support device mobility
B. all phones in the cluster that subscribed to device mobility
C. mobile phones in the cluster that support device mobility
D. mobile phones in the cluster that are in default mode
Correct Answer: A
Explanation

 

QUESTION 12
Refer to the exhibit.pass4itsure 300-075 exam question pass4itsure 300-075 exam questionTo stream multicast MOH to the remote site across the WAN, what should the minimum value for the Max Hops be configured as?
A. 1
B. 2
C. 3
D. 4
Correct Answer: C
Explanation
Explanation/Reference:
The Max Hops field in the Music On Hold (MOH) Server Configuration window indicates the maximum number of routers that an audio source is allowed to cross. If max hops is set to zero, the audio source must remain in its own subnet. If
max hops is set to one, the audio source can cross up to one router to the next subnet. Cisco recommends setting max hops to two.
Link:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmfeat/fsmoh.html

 

QUESTION 13
Refer to the exhibit.

pass4itsure 300-075 exam question

The HQ site uses area code 650. The BR1 site uses area code 408. The long distance national code for PSTN dialing is 1. To make a long distance national call, an HQ or BR1 user dials access code 9, followed by 1, and then the 10-digit
number.
Both sites use MGCP gateways. AAR must use globalized call routing using a single route pattern. Assume that all outgoing PSTN numbers are localized at the egress gateway as shown in the exhibit.
How many route lists and route groups should be configured for AAR at a minimum?
A. a single route list with a local route group for each site
B. two route lists and two route groups for each site
C. a single route list and four route groups for each sitea
D. None. The AAR CSS can point directly to the route pattern.
Correct Answer: A
Explanation

 

QUESTION 14
Refer to the exhibit.

pass4itsure 300-075 exam question

What must be configured on the HQ Cisco Unified Communications Manager to allow HQ users to dial the SAF learned directory number pattern 3XXX?
A. Route pattern 3XXX should be configured and made available to HQ users through the phone CSS.
B. Route pattern 3XXX should be configured and made available to HQ phone users through the phone AAR CSS.
C. The SAF partition assigned to the SAF learned patterns must be available to the HQ phone users through the phone CSS.
D. The SAF partition assigned to the SAF learned patterns must be available to the HQ phone users through the phone AAR CSS.
E. The SAF directory number pattern 3XXX will be made available to all users automatically as soon as the SAF partition is selected.
Correct Answer: C
Explanation
Explanation/Reference:
Explanation: Incorrect answer: ABD
By adopting the SAF network service, the call control discovery feature allows Cisco Unified Communications Manager to advertise itself along with other key attributes.
Link:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmfeat/fscallcontrol discovery.html

 

QUESTION 15
Refer to the exhibit.

pass4itsure 300-075 exam question

All HQ phones are configured to use HQ_MRGL and all BR phones are configured to use BR_MRGL. For the HQ phones always to use the hardware conference bridge as a first choice, which configuration should be implemented?
A. 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.
B. Ensure that both the hardware and software conference bridges are listed in the HQ_MRG. The hardware conference bridge must be configured first.
C. Assign the hardware conference bridge to HQ_MRG. Configure a second HQ_MRG_2 and assign the software conference bridge to it. Add both the HQ_MRG and HQ_MRG_2 to the HQ_MRGL and list the HQ_MRG first.
D. Assign the hardware conference bridge to HQ_MRG. Configure a second HQ_MRG_2 and assign the software conference bridge to it. Configure an additional HQ_MRGL_2. Add the HQ_MRG to HQ_MRGL. Add HQ_MRG_2 to
HQ_MRGL_2. The HQ_MRGL should be assigned to the HQ phones. The HQ_MRGL_2 should be assigned to the HQ device pool.
Correct Answer: CExplanation
Explanation/Reference:
Explanation: Expiation:
To ensure that the hardware bridge is utilized first with all its resources BEFORE the software bridge is used … you need to have two separate MRG’s and list the hardware MRG 1st in the MRGL …

 

QUESTION 16
Refer to the exhibit.pass4itsure 300-075 exam question

Which pattern will be advertised try the Cisco Unified Communications Manager?
A. 3XXX and the ToDID will be 0:.
B. 3XXX and the TnOID will be 0:44228822.
C. 3XXX and the ToDID will be 44228822.
D. 3XXX and the ToDID will be 0:+44228822.
E. 3XXX and the ToDID will be 0:+
Correct Answer: A
Explanation
Explanation/Reference:
Explanation: Incorrect answer: BCD
the number of digits that want stripped from the hosted DN if the call fails over to the PSTN.
Value can enter 0 and 16. Here 0 is defined.
Link:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmfeat/fscallcontrol discovery.html

 

QUESTION 17
Which task must you perform before deleting a transcoder?
A. Delete the dependency records.
B. Unassign it from a media resource group.
C. Use the Reset option.
D. Remove the device pool.
E. Remove the subunit.
F. Delete the common device configuration.
Correct Answer: B
Explanation

 

QUESTION 18
What command is used to map internal extensions to the corresponding E.164 PSTN number when using Cisco Unified Communications Manager Express in SRST mode?
A. ephone-dn
B. dialplan-pattern
C. number
D. number-e.164
E. ephone-transnumber
Correct Answer: B
Explanation

 

QUESTION 19
Refer to the exhibit

pass4itsure 300-075 exam question

The exhibit shows a SAF Forwarder configuration attached to a Cisco Unified Communications Manager.
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. 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. 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 Loopback1 transport tcp port 5060 !
profile dn-block 1 alias-prefix 1972555
pattern 1 type extension 4xxx
!
profile callcontrol 1dn-service
trunk-route 1
dn-block 1
dn-block 2
!
channel 1 vrouter SAF asystem 1
subscribe callcontrol wildcarded
publish callcontrol 1
!
C. 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.
Correct Answer: C
Explanation
Explanation/Reference:
Explanation: Incorrect answer: ABD
only following configuration is enough
router eigrp SAF
service-family ipv4 autonomous-system 1
exit-service-family
link:
http://www.cisco.com/en/US/prod/collateral/iosswrel/ps6537/ps6554/ps6599/ps10822/white paper_c11-636604.html

 

QUESTION 20
When multiple Cisco Extension Mobility profiles exist, which actions take place when a user tries to log in to Cisco Extension Mobility?
A. The login will fail because only a single Cisco Extension Mobility profile is allowed.
B. The user must select the desired profile.
C. The user must login to both profiles in the order they are presented.
D. The user may login to both profiles in any order.
E. Login will only be allowed to multiple profiles if the service parameter Allow Multiple Logins is enabled.
Correct Answer: B
Explanation
Explanation/Reference:
Explanation: Incorrect answer: ACDE
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). If a user has more than one user device profile, a prompt displays on the phone and asks the user to choose a device profile for use with Cisco Extension Mobility.
Link:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmfeat/fsem.html

 

QUESTION 21
Refer to the exhibit.pass4itsure 300-075 exam question pass4itsure 300-075 exam questionWhen a call is placed from Phone 1 to Phone 2 and RSVP Agent 2 is not registered to Cisco Unified Communications Manager, what happens?
A. The caller hears a busy tone.
B. The call proceeds as normal since the location setting for the BR site is unlimited.
C. An RSPV call is attempted but fails. The call will fall back to local QoS.
D. the call fails.
E. The phone displays a message that advises the user that the call has been queued.
Correct Answer: DExplanation

 

QUESTION 22
Which 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. direct-inward-dial
B. voice translation-rule
C. incoming called-number
D. application
Correct Answer: A
Explanation

 

QUESTION 23
What is the fastest way for an engineer to test the implementation of SRST in a production environment?
A. Shut down the Cisco Unified Communications Manager Servers.
B. Shut down the switch ports connected to the Cisco Unified Communications Manager Servers.
C. Add a null route to the publisher Cisco Unified Communications Manager at the remote router. Remove the null route when the operation is verified.
D. Unplug the IP phones from their switch ports.
E. Verification is not needed.
Correct Answer: C
Explanation

 

QUESTION 24
What 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.
Correct Answer: B
Explanation

 

QUESTION 25
Refer to the exhibit.

pass4itsure 300-075 exam question

Assuming the regions configuration to BR only permits G.729 codec, how many calls are allowed for the BR location?
A. Total of four calls; two incoming and two outgoing.
B. Total of two calls in either direction.
C. Total of four calls to the BR location. Outgoing calls are not impacted by the location configuration.
D. Total of four calls in either direction.
E. Two outgoing calls. Incoming calls are unlimited.
Correct Answer: D
Explanation
Explanation/Reference:
Explanation: Incorrect answer: ABCE
In performing location bandwidth calculations for purposes of call admission control, Cisco Unified Communications Manager assumes that each G.729 call stream consumes 24 kb/s amount of bandwidth
Link:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmsys/a02cac.html #wpxref28640

 

QUESTION 26
Refer to the exhibit.

pass4itsure 300-075 exam question

What should the destination IP address be configured as on the HQ and BR1 SIP trunks?
A. The HQ SIP trunk destination IP address should be 10.1.6.10. The BR1 SIP trunk destination IP address should be 10.1.5.10.
B. The destination IP address is not configurable. Each cluster will learn about the remote trunk IP address through SAF learned routes.
C. The destination IP address will be learned automatically and configured on the SIP trunks after the Cisco Unified Communications Managers discover themselves.
D. The HQ SIP trunk destination IP address should be the HQ SAF Forwarder IP address.
The BR1 SIP trunk destination IP address should be the BR1 SAF Forwarder IP address.
Correct Answer: B
Explanation
Explanation/Reference:
Explanation: Incorrect answer: ACD
The gatekeeper changes the IP address of this remote device dynamically to reflect the IP address of the remote device.
Link:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmsys/a08trnk.html

 

QUESTION 27
How do RSVP-enabled locations differ from Cisco Unified Communications Manager locations?
A. RSVP is configured in the ISR independent of Cisco Unified Communications Manager.
B. RSVP enables AAR within Cisco Unified Communications Manager.
C. RSVP is topology aware.D. RSVP is configured in Cisco Unified Communications Manager independent of the ISR.
Correct Answer: C
Explanation

 

QUESTION 28
Which two options are effective mechanisms to restrict the maximum number of voice calls on a WAN link? (Choose two.)
A. Configure a gatekeeper with an SIP trunk.
B. Configure a gatekeeper and a gatekeeper-controlled trunk in Cisco Unified Communications Manager with bandwidth control.
C. Configure Cisco Unified Communications Manager regions.
D. Configure Cisco Unified Communications Manager locations.
Correct Answer: BD
Explanation

 

QUESTION 29
Which statement about RSVP is true?
A. MTP is typically used for RSVP agent configuration.
B. If the RSVP agent cannot reserve the required bandwidth on first effort, it enables low latency queuing and tries to connect the call again.
C. Cisco Unified Communications Manager uses the RSVP-configured bandwidth between sites as a method of determining if there is sufficient bandwidth for the call.
D. Cisco Unified Communications Manager uses an RSVP-enabled infrastructure and an RSVP-controlled agent to request a bandwidth reservation from the network in order to place a call.
Correct Answer: D
Explanation

 

QUESTION 30
How is the region assigned to a device such as an IP phone?
A. Regions are assigned directly in the device configuration page.
B. Regions can be assigned only through a device pool.
C. Regions can be assigned either directly on the device configuration page or through the device pool. If both configurations exist, the device pool region configuration takes precedence.
D. Regions can be assigned either directly on the device configuration page or through the device pool. If both configurations exist, the device region configuration takes precedence.
Correct Answer: B
Explanation

 

QUESTION 31
DRAG DROPpass4itsure 300-075 exam question pass4itsure 300-075 exam questionExplanation
Explanation/Reference:

 

QUESTION 32
Refer to the exhibit.

pass4itsure 300-075 exam question

Which configuration elements must match for adjacent neighbors to establish a SAF neighbor relationship?
A. the label name specified in the router eigrp command
B. the autonomous-system number specified in the service-family ipv4 autonomous-system command
C. the sf-interface configuration
D. the topology base configurations
E. the label name specified in the router eigrp command and the autonomous-system number
Correct Answer: B
Explanation
Explanation/Reference:
Explanation: Incorrect answer: ACDE
service-family ipv4 autonomous-system 1 enables a Cisco SAF service family for the specified autonomous system on the router
Link:
http://www.cisco.com/en/US/docs/ios/saf/configuration/guide/saf_cg_ps10591_TSD_Produ cts_Configuration_Guide_Chapter.html#wp1056363

 

QUESTION 33
What is the default value for the Drop Ad Hoc Conference service parameter? (Source:Preventing Toll Fraud)
A. Never
B. When No On-Net Parties Remain in the Conference
C. When No Off-Net Parties Remain in the Conference
D. Drop Ad Hoc Conference When Creator Leaves
Correct Answer: A
Explanation
Explanation/Reference:
Explanation: Incorrect answer: BCD
To use the additional functionality that advanced ad hoc conferencing provides, Cisco recommends that you set this service parameter to Never. Any other setting can result in unintentional termination of a conference.
Link:
https://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmsys/a05confb.html

 

QUESTION 34
Which two entities could be represented by device mobility groups? (Choose two.)
A. countries
B. regions
C. directory numbers
D. transcoders
Correct Answer: AB
Explanation

 

QUESTION 35
Refer to the exhibit.

pass4itsure 300-075 exam question

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: +
Correct Answer: D
Explanation
Explanation/Reference:
Explanation: Incorrect answer: ABC
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
Link:
http://www.ciscosystems.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmsys/a03rp.html

 

QUESTION 36
Refer to the exhibit.pass4itsure 300-075 exam question

A PSTN call arrived at the MGCP gateway. The calling number was received as 14087071222 with number set to type international. The HQ_clng__pty_CSS contains the HQ_clng_pty__Pt partition. Which caller ID is displayed on the IP
phone?
A. +087071222
B. 14087071222
C. 087071222
D. 4087071222
E. 14087071222
Correct Answer: C
Explanation
Explanation/Reference:
Explanation: Incorrect answer: ABDE
SIP trunks and MGCP gateways can support sending the international escape character, +, for calls. H.323 gateways do not support the +. QSIG trunks do not attempt to send the +. For outgoing calls through a gateway that supports +,
Cisco Unified Communications Manager can send the + with the dialed digits to the gateway. For outgoing calls through a gateway that does not support +, the gateway strips the + when Cisco Unified Communications Manager sends the call
information to the gateway.
Link:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmfeat/fscallpn.html

 

QUESTION 37
When 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.
Correct Answer: D
Explanation
Explanation/Reference:
Explanation: Incorrect answer: ABCE
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.
Link:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmfeat/fscallpn.html

 

QUESTION 38
Refer to the exhibit.
If an IP phone in San Jose roams to New York, which two IP phone settings will be modified by Device Mobility so that the phone can place and receive calls in New York? (Choose two.)pass4itsure 300-075 exam question

A. The physical locations are not different, so the configuration of the phone is not modified.
B. The physical locations are different, so the roaming-sensitive parameters of the roaming device pool are applied.
C. The device mobility groups are the same, so the Device Mobility-related settings are applied in addition to the roaming-sensitive parameters.
D. 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.
E. The Device Mobility information is associated with the home device pool of the phone, so the phone is considered to be in its home location. Device Mobility will reconfigure the roaming-sensitive settings of the phone.
Correct Answer: BC
Explanation

 

QUESTION 39
Which two Cisco Extension Mobility attributes are available in the user device profile? (Choose two.)
A. regions
B. description
C. phone button template
D. NTP information
Correct Answer: BC
ExplanationExplanation/Reference:

 

QUESTION 40
Which three commands are mandatory to implement SRST for five Cisco IP Phones? (Choose three.)
A. call-manager-fallback
B. max-ephones
C. keepalive
D. limit-dn
E. ip source-address
Correct Answer: ABE
Explanation

 

QUESTION 41
Refer to the exhibit.

pass4itsure 300-075 exam question

The HQ Cisco Unified Communications Manager has been configured for end-to-end RSVP. 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. When a call is placed from the IP phone at HQ to the BR phone at the BR site, which statement is true?
A. The Cisco Unified Communications Manager at HQ will fall back to local RSVP and place the call. No RSVP end-to-end will occur.
B. RSVP end-to-end will occur.
C. The Cisco Unified Communications Manager at HQ will use end-to-end RSVP. The BR Cisco Unified Communications Manager will use local RSVP.
D. The call will fail.
E. The call will proceed as a normal call with no RSVP reservation.
Correct Answer: D
Explanation
Explanation/Reference:
Explanation: Incorrect answer: ABC
A possible cause is that the same router is being used as the calling and called RSVP agents, and that router is not running the latest IOS version, which supports loopback on RSVP reservation. Make sure that the router is running the latest
IOS version.
Link:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmsys/a02rsvp.html #wp1155102

 

QUESTION 42
Refer to the exhibit.

pass4itsure 300-075 exam question

What media resource should be configured in Cisco Unified Communications Manager?
A. Cisco Media Termination Point Hardware
B. Cisco IOS Enhanced Media Termination Point Cisco
C. Cisco IOS Media Termination Point
D. Cisco Media Termination Point Hardware (WS-SVC-CMM)
Correct Answer: B
Explanation
Explanation/Reference:
Explanation: Incorrect answer: ACD
Each media termination point receives a list of Cisco Unified Communications Managers, in priority order, to which it should attempt to register. Each media termination point can register with only one Cisco Unified Communications Manager
at a time.
Link:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_configuration_example0 9186a0080b4f916.shtml

 

QUESTION 43
Refer to the exhibit.

pass4itsure 300-075 exam question

How many calls can be placed to Cluster B?A. three G.729 calls
B. one G.711 call
C. one G.711 and three G.729 calls
D. There is no limit for incoming calls to Cluster B. Outgoing calls are limited to one G.711 and three G.729 calls.
Correct Answer: A
Explanation

 

QUESTION 44
Refer to the exhibit.pass4itsure 300-075 exam questionWhich statement about the configuration between the Default and BR regions is true?
A. Calls between the two regions can use either 64 kbps or 8 kbps.
B. Calls between the two regions can use only the G.729 codec.
C. Only 64 kbps will be used between the two regions because the link is “lossy”.
D. Both codecs can be used depending on the loss statistics of the link. When lossy conditions are high, the G.711 codec will be used.
Correct Answer: B
Explanation

 

QUESTION 45
When an external call is placed from Ajax, they would like the ANI that is sent to the PSTN to be the main number, not the extension. For domestic calls, they would like 10 digits sent; for international calls, they would like to send the country
code 1 and the 10 digits. How can this be accomplished?
A. Add a translation pattern to the dial peers in the gateway that adds the appropriate digits to the outgoing ANI.
B. In the external call route patterns, set the external phone number mask to the main number. Use 10 digits in the domestic route pattern and 1 followed by the main number digits in the international route patterns.
C. Use a calling party transform mask for each route group in the corresponding route list configuration. Set the explicit 10-digit main number for domestic calls and 1 followed by the main number for the international route patterns.
D. In the directory number configurations, set the prefix digits field to the country code and the 10 digits of the main number. This will be truncated to the 10-digit number for domestic calls and sent out in its entirety for international calls.
Correct Answer: C
Explanation
Explanation/Reference:
Explanation: Incorrect answer: ABD
calling party transformation mask value is Valid entries for the NANP include the digits 0 through 9; the wildcard characters X, asterisk (*), and octothorpe (#); and the international escape character +.
Link:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucmbe/admin/8_6_1/ccmcfg/b03trpat.ht ml

 

QUESTION 46
During device failover to the secondary Cisco Unified Communications Manager server, how does the phone recognize that the primary server is back?
A. The secondary server keeps sending keepalive message to the primary server and when it succeeds, it unregisters the phones to force them to register to the primary.
B. When the primary server goes online, it sends out an “ALIVE” message via broadcast so that the phones re-register.
C. The phones never re-register with the primary server until the active (secondary) one goes down.
D. The phone sends keepalive messages to the primary server frequently and when it succeeds, the phone re-registers with it.
Correct Answer: D
Explanation

 

QUESTION 47
Refer to the exhibit. You have configured transcoder resources in both an IOS router and a Cisco Unified Communications Manager. When you review the configurations in both devices the IP addresses and transcoder names are correct, but
the transcoder is failing to register with the Cisco Unified Communications Manager. Which command needs to be edited to allow the transcoder to register properly?

pass4itsure 300-075 exam question

A. The associate profile and dsp farm profile numbers need to match associate ccm 2 command.
B. The associate ccm 2 priority 1 command needs to be changed so the ccm value matches identifier 1 in the sccp ccm 10.1.1.1 command.
C. The sccp ccm group number needs to match the associate ccm 2 command.
D. The maximum sessions command must match the number of codecs configured under the dsp farm profile.
E. The sccp ccm group number must match the voice-card number.
Correct Answer: B
Explanation
Explanation/Reference:
Explanation: Incorrect answer: ACDE
The value of the IP address should match the IP address in the ip source-address command
Link:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmevtr ns.html

 

QUESTION 48
You need to verify if the Media Control Gateway Protocol gateway is enabled and active.
Which command should you use for this purpose?
A. show running-configB. show fallback-mgcp
C. show gateway
D. show ccm-manager fallback-mgcp
E. show running-config gateway
F. show fallback-mgcp ccm-manager
Correct Answer: D
Explanation

 

QUESTION 49
Which device is needed to integrate H.320 into the Cisco video solution? (SourcE.
Introducing IP Video Telephony)
A. video gateway
B. MGCP gateway
C. H.323 gatekeeper
D. MCU
Correct Answer: C
Explanation
Explanation/Reference:
Explanation: Incorrect answer: ABD
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.
Link:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/4x/42video.html#wp1046523

 

QUESTION 50
When an incoming PSTN call arrives at an H.323 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 H.323 gateway configuration in Cisco Unified Communications Manager.
B. Normalization is done using route patterns.
C. Normalization is done using the gateway incoming calling party prefixes based on number type.
D. Normalization is achieved by local route group that is assigned to the H.323 gateway.
Correct Answer: A
Explanation

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