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642-445 : Cisco IP Telephony for Release 5.0 Last Updated Monday, April 21, 2008 with 523 Questions

Cisco IP Telephony for Release 5.x
Exam Number: 642-445 Exam
Associated Certifications: Cisco IP Telephony for Release 5.x
Duration: 95 Q&As
Available Language(s): English
Exam Details
The Cisco IP Telephony for release 5.x 642-445 CIPT-5.x is the exam associated with the CCVP and Cisco IP Communications Support Specialist certifications. Candidates can prepare for this exam by taking the Cisco IP Telephony (Part 1) v5.0 and Cisco IP Telephony (Part 2) v5.0 courses. This exam tests a candidate’s knowledge of VoIP and PSTN components and technologies and the candidate’s ability to describe, install, configure and support CCM 5.x products in a Cisco network, including such features as security and video. Topics covered include VoIP, PSTN, and CCM.

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642-445 Exam Preparation from Actualtests with FULL explanations include:
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642-445 Exam: Actualtests’s Cisco IP Telephony for Release 5.x PDF
The Cisco IP Telephony for Release 5.x PDF for preparing for the 642-445 exam - Actualtests’s Cisco IP Telephony for Release 5.x. Actualtests is your premier source for practice tests, and true testing environment. Nothing will prepare you for your next exam like a Actualtests. You find it all here at ciscoexams.org.

QUESTION 1
How can the activation status of DNS services be changed?
A. DNS activation can only be changed by reinstalling Cisco Unified CallManager.
B. DNS activation can be changed at any time.
C. DNS activation is on by default and can only be turned off at installation; once DNS
activation has been turned off, the system must be reinstalled to turn it back on.
D. DNS activation is on by default and cannot be changed, but leaving the DNS server
addresses and domain name blank disables the service.
Answer: A
QUESTION 2
Which two steps must a user take to use a prepared stock quote XML application on an
IP phone? (Choose two.)
A. Initiate an application TFTP request.
B. Select the Services button and choose the stock option application.
C. Use the Settings button to activate the application on the phone.
D. Request that the administrator add their directory number to the application list.
E. Subscribe, or have the Cisco Unified CallManager administrator subscribe, the phone
to the service using the User Options web page.
F. Dial the directory number for the service.
Answer: B,E
QUESTION 3
DRAG DROP
You work as a network technician at Certkiller .com. Your boss, Miss Certkiller, is
interested in Presence terms. Match the proper Presence term with appropriate
description.
642-445
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Answer:
Explanation:
Presence is defined as the “availability and willingness of the user (presentity) for
communication”. Presence information is published by individuals to other systems users, known
as ‘watchers’ or ’subscribers’, to indicate their communication state. A user may publish a variety
of presence states to indicate their communication status.
When you configure Presence in Cisco Unified CallManager Administration, an interested party,
known as a watcher, can monitor the real-time status of a directory number or SIP URI, a
presence entity, from the device of the watcher. A phone serves as a watcher; a line on a phone
cannot serve as a watcher. Cisco Unified CallManager allows you to configure presence groups
to control the destinations that watchers can monitor.
The Presence feature allows a user to monitor the real-time status of another user at a directory
number or SIP URI.
642-445
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http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_chapter09186a00806
2b705.html#
QUESTION 4
DRAG DROP
You work as a network technician at Certkiller .com. Your boss, miss Certkiller, is
interested threat types. Match the proper threat type with appropriate threat.
Answer:
QUESTION 5
Which two SIP issues should be considered when deploying KPML? (Choose two.)
A. Additional MTP resources will be required.
B. There will be significantly more signaling traffic on the network.
C. Additional configuration steps will be required for SIP phone dial rules.
D. Social engineering will be required to educate users about how to dial a call with SIP.
E. There will be additional processing demands on the Cisco Unified CallManager
server.
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Answer: B,E
Explanation:
There are two methods of handling DTMF support for SIP devices. In-band and Out-of-Band.
Out-of-Band DTMF support for SIP devices will use Key Pad Mark-Up Language (KPML),
introduced in CCM 5.0. A SIP device can use In-band DTMF support, but is based on RFC 2833.
The KPML is also used for digit collection when dialing on a SIP phone. When no dial plan in the
phone exists, CallManager performs digit-by-digit collection and routes the call as soon as
enough digits are collected. There is a performance impact on CallManager, in terms of call
signaling bandwidth, for this KPML type of digit collection and transmission. MTP resources
would be allocated for CallManager 4.x and below, as DTMF method for all SIP devices were
using RFC 2833 In-band.
QUESTION 6
How does the Cisco Unified CallManager use RSVP?
A. it uses RSVP reservations to request double the bandwidth so calls can flow in both
directions
B. it uses the RSVP-configured bandwidth between sites as a method of determining if
there is sufficient bandwidth for a call
C. it uses an RSVP-enabled infrastructure and an RSVP agent controlled by Cisco
Unified CallManager to request a bandwidth reservation from the network in order to
place a call
D. it initiates RSVP reservations from the Cisco Unified CallManager server, which is an
active participant in the reservation process
E. it uses RSVP bandwidth and zones to determine how and if calls can be placed
Answer: C
QUESTION 7
DRAG DROP
You work as a network technician at Certkiller .com. Your boss, miss Certkiller, is
interested MoH parameters. Match the proper paramter with appropriate description.
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Answer:
QUESTION 8
A range of Meet-Me conference numbers is reserved for manager use. Which method can
be used to restrict non-managers from accessing these Meet-Me numbers without limiting
their ability to initiate ad hoc conferences?
A. Place the Meet-Me numbers in a partition that is only assigned to the CSS for the
managers’ phones.
B. Place the Meet-Me numbers in an MRGL that is only assigned to the managers’
phones.
C. Place the conference bridge resource in an MRGL that is only assigned to the
managers’ phones.
D. Place the conference bridge resource in a partition that is only assigned to the CSS for
the managers’ phones.
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Answer: A
QUESTION 9
You need to delete and add a new device pool to your Cisco Unified CallManager
cluster, but when you try to delete the existing device pool you receive an error message
stating that the device pool is still in use. Which tool can you use to determine which
devices are still using the device pool?
A. the Dial Plan report
B. Device Pool records
C. Dependency records
D. Cisco Unified CallManager group records
Answer: C
QUESTION 10
What is the relationship between a Region and a Location?
A. The Region codec parameter is used between a Region and its configured Locations.
B. The Region setting for a Location sets the number of audio and video calls that
Location can support.
C. The codec parameter configured in the Region is only used between Regions and
Location bandwidth is only used between Locations.
D. The Region codec parameter is combined with Location bandwidth when
communicating with other Regions.
Answer: C
Explanation:
Region settings specify voice codec that can be used for calls within a Region and between
Regions. Locations provide Call Admission Control for centralized call processing systems. Calls
between locations that travel over a WAN link most likely have limited available bandwidth. If the
number of calls placed over a WAN link is high enough to deplete the amount of available
bandwidth on the link, the audio quality of all the calls over that WAN link is degraded due to
over-subscription. You use Locations to avoid oversubscribing the WAN link such as decreasing
the number of calls that are completed to a particular location, thus you define the total amount of
bandwidth used for voice calls assigned to a location.

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