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QUESTION 50
If 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, which two types of Call Admission Control can be implemented? (Choose two.)
A. locations based
B. automated alternate routing
C. gatekeeper based
D. SRST
E. Cisco Unified Communications Manager based
Correct Answer: AE Section: Call Admission Control Explanation

Explanation/Reference:
Ok
QUESTION 51
Which device is needed to integrate H.320 into the Cisco video solution?
A. video gateway
B. MGCP gateway
C. H.323 gatekeeper
D. MCU

Correct Answer: D Section: Configuration Explanation
Explanation/Reference:
Ok
QUESTION 52
Which statement about enrollment in the IP telephony PKI is true?
A. CAPF enrollment supports the use of authentication strings.
B. The CAPF itself has to enroll with the Cisco CTL client.
C. LSCs are issued by the Cisco CTL client or by the CAPF.
D. MICs are issued by the CAPF itself or by an external CA.

Correct Answer: A Section: Questions not in the test Explanation
Explanation/Reference:
Ok (Source: Understanding Cisco IP Telephony Authentication and Encryption Fundamentals) old question!!
QUESTION 53
Refer 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 configurationis 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-gatewayvoip 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.

Correct Answer: AC Section: Configuration Explanation
Explanation/Reference:
Ok.
QUESTION 54
Which two have to be defined in the Forward All field? (Choose two.)
A. calling search space
B. destination
C. partition
D. hunt list

Correct Answer: AB Section: Configuration Explanation
Explanation/Reference:
Ok
QUESTION 55
Refer to the exhibit. With the Mobile Connect feature configured, when the PSTN phone calls the Enterprise user at extension 3001, 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

A. the gateway CSS
B. the Phone Device CSS
C. the Remote Destination Profile CSS
D. the Remote Destination Profile Rerouting CSS
E. the PhoneLine(DN)CSS

Correct Answer: D Section: Questions not in the test Explanation
Explanation/Reference:
Ok 6-77 CIPT1 (old question!!!)
QUESTION 56
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?

A. The associate profile anddsp 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. Thesccp ccm group number needs to match the associate ccm 2 command.
D. The maximum sessions command must match the number ofcodecs configured under the dsp farm profile.
E. Thesccp ccm group number must match the voice-card number.

Correct Answer: B Section: Configuration Explanation
Explanation/Reference:
Ok 3-19, CIPT2
QUESTION 57
Which statement is correct about AAR?
A. The end users sees, “Network Congestion Rerouting?” but AAR is otherwise transparent to the end user and works without user intervention.
B. AAR will display “not enough bandwidth” on the IP phone while it reroutes the call.
C. AAR allows calls to be rerouted because of insufficient Cisco Unified Border Element controlled bandwidth to an ITSP.
D. AAR allows calls to be rerouted due to insufficient gatekeeper controlled IP WAN bandwidth.

Correct Answer: A Section: AAR Explanation
Explanation/Reference:
Ok, CIPT2 3-76
QUESTION 58
The relationship between a Region and a Location is that the Region codec parameter is used between a Region and its configured Locations.
A. TRUE
B. FALSE

Correct Answer: B Section: Configuration Explanation
Explanation/Reference:
Ok.
QUESTION 59
An update of the configuration using the Cisco CTL client not needed when _______.
A. a Cisco UnifiedCallManager has been removed
B. an LSC of the IP phone is upgraded
C. a security token is added to the system
D. an IP address of the Cisco TFTP server has been changed

Correct Answer: B Section: Questions not in the test Explanation
Explanation/Reference:
(Source: Configuring Cisco IP Telephony Authentication and Encryption)
QUESTION 60
Refer 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. The user in RTP sees the message “Not Enough Bandwidth” on their phone and hears a fast busy tone. Which two conditions can correct this issue? (Choose two.)

A. The calling phone (RTP) needs to have AAR Group value of AAR under the AAR Settings.
B. The called phone (San Jose) needs to have AAR Group value of AAR under the AAR Settings.
C. The calling phone (RTP) needs to have the AAR destination mask of 914085550150 configured under the AAR Settings.
D. The calling phone (RTP) needs to have the AAR destination mask of 4085550150 configured under the AAR Settings.
E. The called phone (San Jose) needs to have the AAR destination mask of 914085550150 configured under the AAR Settings.
F. The called phone (San Jose) needs to have the AAR destination mask of 4085550150 configured under the AAR Settings.

Correct Answer: BF Section: AAR Explanation
Explanation/Reference:
Ok
QUESTION 61
Which statement is not true about GARP?
A. GARP attacks require access to the target LAN or VLAN.
B. GARP can be used for a man-in-the-middle attack.
C. GARP is normally used for HSRP.
D. GARP can be disabled at Cisco IP phones.

Correct Answer: C Section: Questions not in the test Explanation
Explanation/Reference:
Ok – (Source: Hardening the IP Phone) old question!!!
QUESTION 62
Which two statements describe RSVP-enabled locations-based CAC? (Choose two.)
A. RSVP can be enabled selectively between pairs of locations.
B. Using RSVP for CAC simply allows admitting or denying calls based on a logical configuration that is ignoring the physical topology.
C. RSVP is topology aware, but only works with full mesh networks.
D. An RSVP agent is a Media Termination Point that the call has to flow through.
E. RSVP and RTP are used between the two endpoints.

Correct Answer: AD Section: Configuration Explanation
Explanation/Reference:
Ok. See 3-51 CIPT2
QUESTION 63
You are the Cisco Unified Communications Manager in Certpaper.com. You use a remote site MGCP gateway to provide redundancy when connectivity to the central Cisco Unified Communications Manager cluster is lost. How to enable IP phones to establish calls to the PSTN when they have registered with the gateway? (Choose three.)
A. POTS dial peers must be added to the gateway to route calls from the IP phones to the PSTN.
B. The default service must be enabled globally.
C. The command ccm-manager mgcp-fallback must be configured.
D. COR needs to be configured to disallow outbound calls.

Correct Answer: ABC Section: Configuration Explanation
Explanation/Reference:
Ok, check 2-37 and 2-47
QUESTION 64
Which two configurations provide the best SIP trunk redundancy with Cisco Unified Communications Manager? (Choose two.)
A. Configure all SIP trunks with DNS SRV.
B. Configure all SIP trunks with Cisco Unified Border Element.
C. Configure all SIP trunks to point to a SIP gateway.
D. Configure SIP trunks to be members of route groups and route lists.
E. Configure all SIP trunks to allow TCP ports 5060.
F. Configure all SIP trunks to point to a gatekeeper through SIP to H.323 gateway.

Correct Answer: AD Section: Sip Trunk Explanation
Explanation/Reference:
Ok, Configure all SIP trunks with DNS SRV.
QUESTION 65
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 Section: Configuration Explanation
Explanation/Reference:
Ok, only using a calling party transformation mask can be possible.
QUESTION 66
The relationship between a Region and a Location is that the Region codec parameter is combined with Location bandwidth when communicating with other Regions.
A. FALSE
B. TRUE

Correct Answer: A Section: Configuration Explanation
Explanation/Reference:
Ok
QUESTION 67
Video calls using 384 kbps need to be supported across a gatekeeper-controlled trunk. What value should be entered into the gatekeeper to support this bandwidth?
A. 768 kbps
B. 384 kbps
C. 512 kbps
D. 192 kbps

Correct Answer: A Section: Configuration Explanation
Explanation/Reference:
Ok see 3-97 CIPT2
QUESTION 68
Enabling authentication and encryption for CTI, JTAPI, and TAPI applications requires which two tasks? (Choose two.)
A. Enter the encryption key into the application.
B. Set up anIPsec 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 Section: Questions not in the test Explanation
Explanation/Reference:
Ok, old question
QUESTION 69
Which two statements about symmetric encryption are true? (Choose two.)
A. With symmetric encryption, the encryption key equals the decryption key.
B. Symmetric encryption is commonly used to sign asymmetric keys.
C. Symmetric encryption is a good choice for real-time encryption of bulk data.
D. Symmetric encryption uses asymmetric keys.

Correct Answer: AC Section: Questions not in the test Explanation
Explanation/Reference:
Ok, old question
QUESTION 70
You are the Cisco Unified Communications Manager in Certpaper.com. After you add the Tcl paramspace command, the application can____.
A. set aside memory for application variables
B. access the data on an internal server
C. access the data on an external server
D. share parameters between different call applications

Correct Answer: D Section: Configuration Explanation
Explanation/Reference:
Ok, it could be an old question.

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