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QUESTION 50
Please choose the location of RAI configuration from the following options.
A. On a gatekeeper
B. On a gateway
C. On both a gatekeeper and a gateway
D. On a Cisco Unified Communications Manager server
Correct Answer: B Section: (none) Explanation
Explanation/Reference:
Explanation: To allow gatekeepers to make intelligent call routing decisions, the gateway reports the status of its resource availability to its gatekeeper. Resources that are monitored are DS0 channels and DSP channels. The gateway reports its resource status to the gatekeeper with the use of RAS Resource Availability Indication (RAI). When a monitored resource falls below a configurable threshold, the gateway sends an RAI to the gatekeeper that indicates that the gateway is almost out of resources. When the available resources then cross above another configurable threshold, the gateway sends an RAI that indicates that the resource depletion condition no longer exists. This feature was included in Cisco IOS Software Release 12.0(5)T on the Cisco AS5300 gateway, and Cisco IOS Software Release 12.1(1)T for other gateways in H.323 version 2 http://www.cisco.com/en/US/tech/tk1077/ technologies_tech_note09186a0080093f67.shtml
QUESTION 51
You have been employed as a network technician in a middle-sized company. Suppose that the default dial peer is matched. Please choose a capability that you must configure.
A. disable DID
B. invoke a Tcl application
C. enable dtmf-relay
D. disable VAD
Correct Answer: A Section: (none) Explanation
Explanation/Reference:
Explanation:
Dial-peer 0(pid:0) has a default configuration that cannot be changed. The defaultdial-peer 0fails to
negotiate non-default capabilities, services, and applications such as:
QUESTION 52
Which four of the following options are Cisco-supported IP telephony deployment models?
A. Single site
B. Multisite with distributed call processing
C. Multisite with centralized call processing
D. Clustering over the IP WAN
E. Transcoding
Correct Answer: ABCD Section: (none) Explanation
QUESTION 53
MGCP use which call control model?
A. Distributed
B. Centralized
C. Ad hoc
D. Hybrid
Correct Answer: B Section: (none) Explanation
QUESTION 54
If you are required to configure a router to use MGCP on a digital port, which measure will you take?
A. Add the application mgcpapp subcommand to the dial peer
B. Add the service mgcp subcommand to the dial peer
C. Add the parameter application mgcpapp to the ds0-group controller subcommand.
D. Add the service mgcp parameter to the ds0-group controller subcommand
Correct Answer: D Section: (none) Explanation
QUESTION 55
As a network technician, you should be familiar with various commands. Which command displays a count of successful and unsuccessful control commands?
A. show mgcp calls
B. show mgcp statistics
C. show mgcp
D. debug mgcp statistics
Correct Answer: B Section: (none) Explanation
QUESTION 56
In a SIP direct call setup, which message will be sent by the originating UAC to the UAS of the recipient?
A. INVITE
B. RINGING
C. ACK
D. OK
Correct Answer: A Section: (none) Explanation
QUESTION 57
Which two of the following signaling protocols are peer-to-peer protocols? (Select two.)
A. H.323
B. MGCP
C. SIP
D. SCCP
Correct Answer: AC Section: (none) Explanation
QUESTION 58
In a Cisco UCM single-site deployment, please choose the maximum number of IP phones that can register with a UCM cluster.
A. 2500
B. 7500
C. 10,000
D. 30,000
Correct Answer: D Section: (none) Explanation
QUESTION 59
In a Cisco UCM multisite WAN with centralized call-processing deployment model, what redundancy feature should be configured on remote site routers to supply basic IP telephony services in the event of a WAN outage?
A. AAR
B. SRST
C. CAC
D. V3PN
Correct Answer: B Section: (none) Explanation
QUESTION 60
Look at the following options carefully. Which two tasks are performed by the RAS signaling function of H.225.0? (Select two.)
A. Performs bandwidth changes.
B. Transports audio messages between endpoints.
C. Performs disengage procedures between endpoints and a gatekeeper.
D. Allows endpoints to create connections between call agents.
Correct Answer: AC Section: (none) Explanation
QUESTION 61
As a network administrator, you should be familiar with various commands. Which command can be used to designate a source IP address for a voice gateway?
A. h323-gateway voip interface 186
B. h323-gateway voip h323-id
C. h323-gateway voip bind srcaddr
D. voice service
Correct Answer: C Section: (none) Explanation
QUESTION 62
Look at the following options. Which are SIP servers? (Select four.)
A. Registrar
B. Redirect
C. Location
D. Proxy
Correct Answer: ABCD Section: (none) Explanation
QUESTION 63
The knowledge about RAS message is very important. Which of the following RAS messages can be sent by using either unicast or multicast?
A. RRQ
B. ARQ
C. GRQ
D. RIP
Correct Answer: C Section: (none) Explanation
Explanation/Reference:
Explanation:
Typically, RAS communications is carried out via UDP through port 1719 (unicast) and 1718 (multicast)
QUESTION 64
Given the following configuration, what IP address will GK1 use to send and receive RAS messages?
GK1 (config)#interface serial 0/0/0
GK1 (config-if)#ip address 192.168.0.2 255.255.255.0
GK1 (config-if)#exit
GK1 (config)#interface serial 0/0/1
GK1 (config-if)#ip address 172.16.0.2 255.255.255.0
GK1 (config-if)#exit
GK1 (config)#gatekeeper
GK1 (config-gk)#zone local SanJose cisco.com 172.16.0.2
GK1 (config-gk)#zone remote Austin cisco.com 192.168.0.1
GK1 (config-gk)#zone prefix SanJose 2…
GK1 (config-gk)#zone prefix Austin 3…
A. 192.168.0.2
B. 172.16.0.2
C. 192.168.0.1
D. RAS messages will be load balanced between 192.168.0.2 and 172.16.0.2
Correct Answer: B Section: (none) Explanation
QUESTION 65
You are a network technician working in the Network Company. Recently, users complain that they cannot call the PSTN. With the help of testing, you find that the gateway is not switching to the secondary call agent when the primary call agent is unreachable. In order to permit the MGCP gateway to take use of a different call agent once the primary fails, which configuration should you make?
A. Add ccm-manager fallback-mgcp command to the gateway.
B. Add ccm-manager redundant-host command to the gateway
C. Assign a Cisco Unified CallManager group including the secondary call agent to the gateway
D. Define gateway as a non-gatekeeper-controlled intercluster trunk with the secondary Cisco Unified CallManager defined.
Correct Answer: B Section: (none) Explanation
QUESTION 66
Which RAS message does a gateway use to request admission to a network and to also request phone number to IP address resolution?
A. ARQ
B. IRQ
C. LRQ
D. RRQ
Correct Answer: A Section: (none) Explanation
Explanation/Reference:
Explanation: Admission messages between endpoints (like a gateway) and gatekeepers provide the basis for call admissions and bandwidth control. Gatekeepers authorize access to H.323 networks with the confirmation of or rejection of an admission request. This table defines the RAS admission messages http://www.cisco.com/en/US/tech/tk1077/technologies_tech_note09186a00800c5e0d.shtml
QUESTION 67
As a network technician, you should be familiar with RTCP. Which of the following statements best describes a function of RTCP?
A. RTCP provides encryption, message authentication and integrity, and anti-replay service for voice streams.
B. RTCP uses even-numbered UDP ports in the range 16,384??0?10?0?43??ì?0?1C32,767 to transport voice payloads
C. RTCP provides out-of-band control information for an RTP flow
D. RTCP caches an RTP packet-Layer 3 and Layer 4 headers in the routers at each end of a link, resulting in lower bandwidth demand for subsequent RTP packets.
Correct Answer: C Section: (none) Explanation
QUESTION 68
You are a voice technician. If you are required to solve latency issues in a VoIP network, which measures will you take? (Select three.)
A. Use dejitter buffers
B. Increase bandwidth
C. Prioritize voice packets
D. Fragment data packets
Correct Answer: BCD Section: (none) Explanation
QUESTION 69
Please choose two methods of LRQ forwarding from the following items. (Select two.)
A. LRQ init
B. LRQ blast
C. LRQ static
D. LRQ sequential
Correct Answer: BD Section: (none) Explanation
QUESTION 70
You are a network technician with many years’ experience. Many users complain that they can hear echo when their calls go out an H.323 gateway. You have made some testing for the gateway and have changed the configuration. So the ERL level turns to be 6 dB. Furthermore, the echo-cancel coverage value is raised to 64 ms. Please choose the effect on the voice quality after this modification.
A. Consonants will be chopped by the echo canceller.
B. The increase in echo-cancel coverage will have no effect on voice quality.
C. The ends of sentences will be chopped by the echo canceller.
D. The echo canceller will take 2-3 seconds longer to converge at the beginning of the call.
Correct Answer: D Section: (none) Explanation
Explanation/Reference:
Explanation: echo-cancel coveragecommand Adjusts the coverage size of the echo canceller. This command enables cancellation of voice that is sent out through the interface and received back on the same interface within the configured amount of time. If thelocal loop(the distance from the interface to the connected equipment that is producing the echo) is longer, the configured value of this command should be extended. If you configure a longer value for this command, it takes the echo canceller longer to converge. In this case, the user might hear a slight echo when the connection is initially set up. If the configured value for this command is too short, the user might hear some echo for the duration of the call because the echo canceller is not canceling the longer-delay echoes. There is no echo or echo cancellation on the network side (for example, the non-POTS side of the connection).
QUESTION 71
Refer to the exhibit.
When Alice at extension 2001 places a call to Bob at extension 3001, Bob hears Alice’s voice twice. What type of echo is this classified as?
A. Talker echo.
B. Listener echo.
C. Tail circuit echo.
D. Front end circuit echo.
Correct Answer: B Section: (none) Explanation
QUESTION 72
Refer to the exhibit.
The exhibit shows the output of debug isdn q931. An inbound PSTN call was received by an MGCP gateway that is registered with a Cisco Unified Communications Manager. The call failed to ring extension 3001. If the phone at extension 3001 is registered and reachable through the gateway inbound CSS, which two actions can resolve this issue? (Choose two.)
A. Change the significant digits for inbound calls to 4 in the gateway configuration in CiscoUnified Communications Manager.
B. Configure the digit strip 4 on the MGCP gateway configuration in Cisco UnifiedCommunications Manager under Incoming Called Party Settings.
C. Configure a translation pattern in Cisco Unified Communications Manager that can beaccessed by the gateway CSS to truncate the called number to four digits.
D. Configure a called-party transformation CSS on the gateway in Cisco UnifiedCommunications Manager that includes a pattern that transforms the number from ten digits to four digits.
E. Configure a voice translation profile in the MGCP Cisco IOS gateway with a voice translation rule that truncates the number from ten digits to four digits.
F. Configure the Cisco IOS command num-exp 2288223001 3001 on the gateway.
Correct Answer: AC Section: (none) Explanation QUESTION 73
Which command should you use to resolve a jerky speed issue?
A. playout-delay
B. show voice port
C. comfort-noise
D. echo-cancel enable
E. echo-cancel coverage
F. comfort-echo
Correct Answer: A Section: (none) Explanation
QUESTION 74
You are trying to access the GUI of Cisco Unified Communications Manager. However, it displays a “not accessible” error. In Cisco Unified Serviceability, which two services should you check for and ensure are running on the Control Center Network Services page? (Choose two.)
A. Cisco Certificate Expiry Monitor
B. Cisco CallManager
C. Cisco Trust Verification Service
D. System Application Agent
E. Cisco Tomcat Stats Servlet
F. Cisco Tomcat
Correct Answer: BF Section: (none) Explanation
QUESTION 75
As a voice administrator, you have received reports on issues with call dropping and call failures over a period of time. While troubleshooting, you find that there is a Code Yellow alert due to high CPU usage. You collect the logs that are shown below from the CLI of Cisco Unified Communications Manager.
Nov5 05:12:15, cm01, Error, Cisco CallManager, ccm: 147897: Nov 05 05:12:15.268 UTC: %CCM_CALLMANAGER-CALLMANAGER-3-CodeYellowExit: CodeYellowExit Expected Average Delay:0 Entry Latency:20 Exit Latency:8 Sample Size:10Time Spent in Code Yellow:2 Number of Calls Rejected Due to Call Throttling:60 Total Code Yellow Exit:14 High Priority Queue Depth:0 Normal Priority Queue Depth:5 Low Priority Queue Depth:4 Cluster ID:StandAloneCluster Node ID:cms01, 3653 From these logs, what does “Time Spent in Code Yellow” indicate?
A. A critical overload condition exists that may impact phone registration after 2 hours of this alert.
B. The server stayed in a Code Yellow state for 2 seconds.
C. The server stayed in a Code Yellow state for 2 milliseconds.
D. The server stayed in a Code Yellow state for 2 minutes.
E. The server needs a reboot within 2 hours.
F. There is a call failure and, as a result, one call is rejected every 2 milliseconds.
Correct Answer: C Section: (none) Explanation
QUESTION 76
A customer is trying to register an IP phone. During the registration process, the IP phone receives the configuration file (.xml) from the TFTP server. Which input can you find in the configuration file that is downloaded to the IP phone?
A. firmware to be loaded on IP phone
B. extension number
C. speed dials
D. valid locally significant certificate
E. location of the DHCP server
F. IP address of the IP phone
Correct Answer: A Section: (none) Explanation
QUESTION 77
Which port number is used as a backhaul for Media Gateway Control Protocol?
A. 2426
B. 2427
C. 2428
D. 2429
E. 2456
F. 2458
Correct Answer: C Section: (none) Explanation
QUESTION 78
You are in the final stages of upgrading the Cisco Unified Communications Manager, and you are waiting for dbreplication to complete. Which command should you execute from the Cisco Unified Communications Manager publisher to verify status reports and to check that all the tables are synchronized?
A. utils dbreplication runtimestate
B. utils dbreplication status all
C. utils dbreplication status
D. utils service list
E. utils dbreplication quickaudit
F. utils core active
Correct Answer: A Section: (none) Explanation QUESTION 79
In a Cisco Unified Communications Manager cluster, you make a few changes to the publisher server. However, the phones that are registered with the subscriber server do not receive these changes. You verify that the publisher and subscriber servers are up and running in the cluster.
What do you need to do to resolve this problem?
A. Reboot the publisher server.
B. Reboot the subscriber servers.
C. Manually reload the configuration on the phones.
D. Fix the replication between the publisher and subcriber servers.
E. Manually copy the database changes from the publisher to the subscriber.
F. Re-set up the database replication between the publisher and subscriber.
Correct Answer: D Section: (none) Explanation
QUESTION 80
Which default switchover method is used by the SCCP client to connect to another Cisco Unified Communications Manager after losing connectivity with the first Cisco Unified Communications Manager?
A. immediate
B. urgent
C. graceful
D. panic
E. recovery
F. static
Correct Answer: C Section: (none) Explanation
QUESTION 81
What are three requirements for Quality of Service for voice calls? (Choose three.)
A. jitter less than or equal to 30 ms
B. PoE-supported Layer 2 switches used to connect IP Phones
C. one-way latency less than or equal to 150 ms
D. jitter less than or equal to 45 ms
E. guaranteed bandwidth of 384 kbps for a voice call
F. loss less than or equal to 1 percent
Correct Answer: ACF Section: (none) Explanation
QUESTION 82
If you need to avoid choppy speech, what is the maximum tolerable round-trip delay between two VoIP endpoints?
A. 100 ms
B. 200 ms
C. 300 ms
D. 400 ms
E. 500 ms
F. 800 ms
Correct Answer: C Section: (none) Explanation
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Exam A
QUESTION 1
Which 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
Correct Answer: AB Section: Sip Trunk Explanation
Explanation/Reference:
Ok. See 1-97
-Trunk Types Used by Special Applications
. Call Control Discovery:
-Special trunks configured in Cisco Unified Communications Manager clusters that refer to an SAFenabled network providing Call Control Discovery features.
CCD between CCM and CUCM Express SAF-CCD requires a SIP Trunk or H.323
QUESTION 2
Refer 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?
A. First route group should point only to the U.K. gateway. The second route group should point to the
U.S. gateway.
B. First route group should be only the local route group. The second route group should point to the U.S gateway.
C. First route group should point only to the U.S. gateway. The second route group should be the local route group.
D. The TEHO-US route list should contain only the local route group. The globalized configuration means that the appropriate gateway will be elected automatically.
E. The \+!route pattern should point directly to the U.S gateway.
Correct Answer: C Section: TEHO Explanation
Explanation/Reference:
Ok. See 1-21 CIPT2 v8.0
QUESTION 3
When Cisco Extension Mobility is implemented, how is the audio source for the MOH selected?
A. The audio source that is configured at the user device profile is selected.
B. The audio source that is configured at the home phone of the user is selected.
C. The audio source that is configured at the physical phone used for the Cisco Extension Mobility login is selected.
D. The audio source that is configured in the IP Voice Media Streaming parameters is selected.
Correct Answer: A Section: Extension Mobility Explanation
Explanation/Reference:
Ok, see 4-47: Cisco Extension Mobility Configuration Elements
The device profile is configured with all the user-specific settings that are found at the device level of an IP phone (user MOH audio source, phone button templates, softkey templates, user locales, DND and privacy settings, and phone service subscriptions) and all phone buttons (lines, speed dials, and so on). One or more device profiles are applied to an end user, in the End User Configuration window.
QUESTION 4
In what Cisco solution is Simple Network-Enabled Auto Provision technology used?
A. Cisco Unified Gateway Duplication
B. Cisco UnifiedCallManager Redundancy
C. Cisco Unified SRST
D. Cisco Unified Call Survivability
Correct Answer: C Section: Configuration Explanation
Explanation/Reference:
Ok, check 1-57 and 2-87: Fallback for IP Phones The Cisco Unified SRST gateway uses Simple Network-Enabled Auto Provision (SNAP) technology to autoconfigure the branch office router to provide call processing for Cisco IP phones that are registered with the router
QUESTION 5
Which method can be used to address variable-length dial plans?
A. Overlap sending and receiving.
B. Add a prefix for all calls that are longer than 10-digits long.
C. Use nested translation patterns to eliminate inter-digit timeout.
D. Use the @macro on the route pattern.
E. Use MGCP gateways, which supportvariable-length dial plans.
Correct Answer: A Section: Dial Plan Explanation Explanation/Reference:
Ok, see Dial Plan Solutions, 1-18 and 1-62.
Variable-length numbering plans: Dial string length is determined by timeout. Overlap sending and
receiving is enabled, allowing dialed digits to be signaled one by one instead of being sent as one whole
number.
QUESTION 6
Refer to the exhibit.
To 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: B Section: MOH Explanation Explanation/Reference:
Ok, see 1-46 CIPT2 V8.0, multicast MOH from Branch Router Flash Example is not applicable (answer
Max Hops = 1) because the question asks the stream multicast across the wan, so TTL 2.
Note the MOH server configuration is not useful to answer this question.
QUESTION 7
Refer to the exhibit.
Which statement about the configuration between the Default and BK 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. Bothcodecs 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 Section: MOH Explanation
Explanation/Reference:
Ok. The matrix between Default and BR supports g.729 only.
QUESTION 8
Refer to the exhibit.
When the user of a phone registered to the Cisco Unified Communications Manager places a call to 3001 when the SAF network is down, what happens?
A. The call fails.
B. The call is rerouted to the PSTN with the constructed PSTN number as +442288223001
C. The call is rerouted to the PSTN with the constructed PSTN number as 442288223001
D. 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 +0002288223001
Correct Answer: C Section: SAF Explanation
Explanation/Reference:
OK, SAF, see 5-43 CIPT2, check the ToDID field number 0:442288223001
QUESTION 9
Refer to the exhibit.
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. What should the AAR group prefix be?
A. 9
B. 91
C. none
D. +
E. +1
Correct Answer: C Section: AAR Explanation
Explanation/Reference:
OK, check 3-68, 3-72 CIPT2v8.0:
When local route groups and globalized call routing are implemented, the egress gateway does not need
to be selected by site-specific AAR
CSS, because the egress gateway is determined by the local route group feature.
In summary, when you are using globalized call routing with local route groups, AAR implementation is
extremely simple: only a single AAR CSS and AAR group are required and applied to all phones,
regardless of their location.
QUESTION 10
When Cisco Extension Mobility is implemented, which CSS is used for calling privileges?
A. The user device profile line CSS combined with the device CSS of the physical phone used to log in the extension mobility user.
B. The user device profile device CSS combined with the line CSS of the physical phone used to log in the extension mobility user.
C. Only the user device profile device CSS is used
D. The combined line/device CSS of the physical phone is used to log in the extension mobility user.
E. The combined line/device CSS of the user device profile.
Correct Answer: A Section: Extension Mobility Explanation
Explanation/Reference:
Ok, check the Note section 4-56: Cisco Extension Mobility and CSSs
.
Cisco Extension Mobility does not modify the device CSS.
.
Cisco Extension Mobility modifies the line CSS:
-When using the line/device CSS approach (recommended for CoS implementation): Line CSS of user device profile is applied; CoS settings of the user are enforced. Device CSS is not modified; local gateway selection is allowed, depending on used device (at
any location).
If only device CSSs are used, the CSS is not updated, and no user-specific privileges can be applied. The user inherits the privileges that are configured at the device that is used for logging in.
QUESTION 11
Refer to the exhibit.
Assume that NANP is being used and 9 is used for PSTN access code Long distance national calls are
preceded with 1.
How should the HQ Cisco Unified Communications Manager be configured for calls to 3XXX to be sent to
the gatekeeper at 10.1.6.1 with PSTN backups?
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.
Correct Answer: A Section: Dial Plan Explanation
Explanation/Reference:
Ok, check 3-100 3-101. Note: For a PSTN backup, you need to perform digit manipulation in such a way that the calling number and (more importantly) the called number are transformed to always suit the needs of the device that is actually used. This transformation can be done at the route list, where digit manipulation can be configured per route group. In the example, the called number 9 1 511 555-1234 has to be changed to a 10-digit number for the H.225 trunk, because the gatekeeper is configured with area code prefixes without the long distance 1. The called number must also be changed to an 11-digit number if rerouting the call to the PSTN gateway is necessary. A better solution would be using global transformations at the egress devices (H.225 trunk and PSTN gateways). In a large multisite environment or in an international deployment, the implementation of globalized call routing would be the best solution.
QUESTION 12
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 Section: Dial Plan
Explanation Explanation/Reference:
Ok see 1-136 CIPT2 v8.0
QUESTION 13
Where do you specify the device mobility group and physical location after they have been configured?
A. phones
B. DMI
C. device mobility CSS
D. device pool
E. MRGL
F. locale
Correct Answer: D Section: Configuration Explanation
Explanation/Reference:
Ok, check 4-32, step 3: Configure Device Pools
QUESTION 14
Refer to the exhibit.
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 Section: SAF Explanation
Explanation/Reference:
Ok. See 5-40.
You can configure one SAF-enabled SIP trunk and one SAF-enabled H.323 trunk. With a SAF-enabled
H.323 trunk, you have to first add a standard nongatekeepercontrolled ICT and then check the Enable SAF check box. Once the check box is checked, the IP address field is disabled. The reason is that the configured trunk does not refer to a particular destination IP address but instead acts as a template for a dynamically created trunk once a SAF call is placed. The destination IP address is then taken from the learned SAF service data. The same concept applies to the SAF-enabled SIP trunk. The only difference is that the SAFenabled SIP trunk is a special trunk service type, which is selected before the trunk configuration page is shown. Therefore, there is no extra check box like there is at the nongatekeeper-controlled ICT. The SAF-enabled SIP trunk also does not have a destination IP address field.
QUESTION 15
Refer to the following exhibit.
Which Cisco IOS SAF Forwarder configuration is correct?
A. router eigrp SAF ! service-family ipv4 autonomous-system 1
!
sf-interface FastEthernet0/0
topology base
exit-sf-topology
external-client HQ_SAF
exit-service-family
!
service-family external-client listen ipv4 5050
external-client HQ_SAF
username SAFUSER
password SAFPASSWORD
B. router eigrp SAF ! service-family ipv4 autonomous-system 1
!
sf-interface FastEthernet0/0
topology base
exit-sf-topology
external-client HQ_SAF_FWDER
exit-service-family
!
service-family external-client listen ipv4 5050
external-client HQ_SAF_FWDER
username SAFUSER
password SAFPASSWORD
C. router eigrp SAF ! service-family ipv4 autonomous-system 1
!
sf-interface FastEthernet0/0
topology base
exit-sf-topology
external-client HQ_SAF_FWDER
exit-service-family
!
service-family external-client listen ipv4 5050
external-client HQ_SAF
username SAFUSER
password SAFPASSWORD
D. router eigrp SAF ! service-family ipv4 autonomous-system 1
!
sf-interface FastEthernet0/0
topology base
exit-sf-topology
external-client HQ_SAF
exit-service-family
!
service-family external-client listen ipv4 5050
external-client HQ_SAF_FWDER
username SAFUSER
password SAFPASSWORD
Correct Answer: A Section: SAF Explanation
Explanation/Reference:
OK 5-36, 5-39, check the Client Label.
QUESTION 16
Which 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)
A. Cisco Unified Communications Manager Administration
B. IP phone display
C. Cisco Unified SRST Router
D. Cisco Unified MGCP Fallback Router
E. physical IP phone settings
Correct Answer: BE Section: Configuration Explanation
Explanation/Reference:
Ok, end-user reference
QUESTION 17
Which statement best describes globalized call routing in Cisco Unified Communications Manager?
A. All incoming calling numbers on the phones are displayed as an E 164 with the + prefix.
B. Call routing is based on numbers represented as an E.164 with the + prefix format.
C. All called numbers sent out to the PSTN are in E-164 with the + prefix format.
D. The CSS of all phones contain partitions assigned to route patterns that are in global format.
E. All phone directory numbers are configured as an E.164 with the + prefix.
Correct Answer: B Section: Configuration Explanation Explanation/Reference:
Ok, 1-65 CIPT2 v8.0
QUESTION 18
How is a SIP trunk in Cisco Unified Communications Manager for SIP precondition?
A. The configuration is done by selecting a SIP precondition trunk type.
B. The configuration is automatically selected when RSVP is enabled for the location assigned to the trunk.
C. SIP precondition is configured by selecting E2E for RSVP over SIP on the default SIP profile assigned to the SIP trunk.
D. SIP precondition is configured by configuring a new SIP profile and selecting E2E for RSVP over SIP. The new SIP must then be assigned to the SIP trunk.
Correct Answer: D Section: Sip Trunk Explanation
Explanation/Reference:
Ok see 3-90 and 3-91 CIPT2.
SIP precondition is also called as “RSVP end to end”
QUESTION 19
Refer to the exhibit.
The exhibit shows centralized Cisco Unified Communications Manager configuration components for TEHO calls to U.S area code 408 from the UK The PSTN access code for the UK is 9 and 001 for international calls to the U.S. To match US -TEHO pattern \+!, how should the translation pattern be configured?
A. 9001.4085551 234 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: Discard Digits PreDot
Correct Answer: D Section: TEHO Explanation
Explanation/Reference:
Ok, see 1-137 and 1-148: Globalized Call Routing-TEHO Example
QUESTION 20
Refer to the exhibit.
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. Which partition should be configured in the AAR CSS applied at the phones’?
A. PSTN partition
B. LD partition
C. The HQ AAR CSS must include a partition assigned to route pattern 91408XXXXXXX. The BR1 AAR CSS must include a partition assigned to route pattern 91650XXXXXXX.
D. AAR CSS must contain translation pattern 9.1[2-9]XX[2-9]XXXXXX for each site that must be globalized. Otherwise the called numbers will not be localized at the egrees gateway.
Correct Answer: A Section: AAR Explanation
Explanation/Reference:
???, check 3-72 Doc. answer is B (Why??) agreed internally
QUESTION 21
Refer to the exhibit.
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 3XX will be made available to all user automatically as soon as the SAF partition is selected.
Correct Answer: C Section: SAF Explanation
Explanation/Reference:
OK, 5-24 CIPT2
QUESTION 22
Which statement about H.323 Gatekeeper Call Admission Control is true?
A. Gatekeeper Call Admission Control applies to centralized Cisco Unified Communications deployments that use locations based Call Admission Control.
B. Gatekeeper Call Admission Control applies to distributed Cisco Unified Communications deployments.
C. Gatekeeper Call Admission Control applies only to distributed Cisco Unified Communications Express deployments.
D. Gatekeeper Call Admission Control setting is configured in Cisco Unified Communications.
Correct Answer: B Section: Call Admission Control Explanation
Explanation/Reference:
OK.
QUESTION 23
Refer to the exhibit.
Which pattern will be advertised try the Cisco Unified Communications Manager?
A. 3XXX and the ToDID will be 0:
B. 3XXX and the ToDID 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 Section: Dial Plan Explanation
Explanation/Reference:
Ok, 5-41 and 5-42.
QUESTION 24
Refer to the exhibit.
Locations-based CAC has been configured between HQ and the BR site. Assume that the priority queue has been provisioned correctly for three G.729 calls. What happens when the fourth call is placed from HQ to BR?
A. The call will get through via the WAN but it will experience poor audio quality.
B. The call will fail.
C. The call will be queued until one of the existing calls drop.
D. The call will get through without any issues.
Correct Answer: A Section: Call Admission Control Explanation
Explanation/Reference:
OK. Audio Bandwidth is 96 kbps -> up to 4 g.729 calls are permitted (about 12 kb per call), but QoS is applied (priority 40k) so the 4th call goes on but with poor quality for all the active calls.
QUESTION 25
While configuring Call Survivability in Cisco Unified Communications Manager, what step is mandatory to reach remote sites while in SRST mode?
A. Enable Cisco Remote SiteReachability.
B. Configure CFUR.
C. Enable the SRST checkbox in the MGCP gateway.
D. Configure the H 323 gateway for SRST in Cisco Unified Communications Manager.
E. Enable the Failover Service parameter.
Correct Answer: B Section: Configuration Explanation
Explanation/Reference:
Ok, 2-25 CIPT2 V8.0 The CFUR feature has to be configured on the Cisco Unified Communications Manager to reach remote sites in SRST mode.
QUESTION 26
Which Cisco IOS command is used to verify that the Cisco Unified Communications Manager Express has registered with the SAF forwarder?
A. show eigrp service-family ipv4 clients
B. show eigrp address-family ipv4 clients
C. show voice saf dndball
D. show saf registration
E. show ip saf registration
Correct Answer: A Section: SAF Explanation
Explanation/Reference:
Ok.
QUESTION 27
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 parameterAllow Multiple Logins is enabled.
Correct Answer: B Section: Extension Mobility Explanation
Explanation/Reference:
OK, 4-50 CIPT2 Cisco Extension Mobility Operation.
QUESTION 28
Refer to the exhibit.
What media resource should be configured in Cisco Unified Communications Manager?
A. Cisco Media Termination Point Hardware
B. Cisco IOS Enhanced Media Termination Point
C. Cisco IOS Media Termination Point
D. Cisco Media Termination Point Hardware (WS-SVC-CMM)
Correct Answer: B Section: Configuration Explanation
Explanation/Reference:
Ok, check 3-63 CIPT2
QUESTION 29
Refer to the exhibit.
How does the Cisco Unified Communications Manager advertise dn-block 1?
A. 4XXX and theToDID will 0:
B. 4XXX and theToDID will 0: 1972555
C. 4XXX
D. 4XXX and theToDID will 0 + 1 972555
E. 19725554XXX
Correct Answer: B Section: Dial Plan Explanation
Explanation/Reference:
Ok, 5-50 CIPT2
QUESTION 30
Refer to the exhibit 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 1 dn-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 Section: SAF Explanation
Explanation/Reference:
OK, The command sf-interface is missing but is not mandatory. 5-36 CIPT2
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