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QUESTION 86
Which are three required steps in digitizing voice? (Choose three)
A. Companding the signal.
B. Quantizing the amplitude.
C. Filtering the signal.
D. Sampling the soundwave.
E. Encoding the results in binary form.
Correct Answer: BDE Section: (none) Explanation
Explanation/Reference:
The answer on “Question 1” on page 15 dealing with digitizing voice should be:
1.
Quantizing the amplitude
2.
Sampling the soundwave
3.
Encoding the results in binary form
See page 2-45 of CVoice version 4.1 class books.
Not C: “Filtering” is part of the process to go from digital to analog not analog to digital (p. 2-47)
QUESTION 87
You are the VoIP engineer at Certkiller .com. A Certkiller user complains that she gets a busy tone instead
of a dial tone when she tries to call another user. You want to troubleshoot this problem.
What command should you use?
A. show voice dsp
B. show voice path
C. show voice connection
D. show voice port summary
E. show dial-peer voice summary
Correct Answer: A Section: (none) Explanation QUESTION 88
Your manager asks you for a worksheet defining items that need to be addressed for the future VOIP and
IP telephone rollout.
What items do you put on the worksheet that need to be addresses for the wiring closets? (Choose all that
apply.)
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A. Switches with Inline Power
B. A 7000 series router to backup the switch with HSRP
C. PBX failover
D. UPS systems and Backup power
E. Cooling Requirements (a heat profile)
Correct Answer: ADE Section: (none) Explanation
QUESTION 89
Which device listed below has an intelligent power management system that grants or denies power to various system components based on power availability in the system for use with IP telephony?
A. Cisco 7309 VXD Router
B. Cisco Works plug in
C. CallManager
D. Catalyst 6000 switch
Correct Answer: D Section: (none) Explanation
QUESTION 90
From the list below, what allows a Cisco IP phone to detect the absence of audio and therefore does not transmit packets over the network?
A. Ipng
B. PBX Filters
C. Voice Activation Detection
D. DHE
E. Call Waiting
Correct Answer: C Section: (none) Explanation
QUESTION 91
What Cisco Catalyst Switch command produces the following inline power output? Defalut Inline Power allocation per port: 10.00 Watts (0.23 Amps @42V) Port InlinePowered PowerAllocated Admin Oper Detected mWatt mA @42V
7/1 auto off no 0 0 7/2 auto on yes 5040 120 7/3 auto faulty yes 12600 300 7/4 auto deny yes 0 0 Actualtests.com – The Power of Knowing 642-436
7/5 off off no 0 0
A. show cam inlinepower <mod>|<mod/port>
B. show port inline <mod>|<mod/port>
C. show port inlinepower <mod>|<mod/port>
D. show port power <mod>|<mod/port>
Correct Answer: C Section: (none) Explanation
QUESTION 92
What are two constraints that you may encounter when trying to design a IP Telephone infrastructure?
A. Upper level management acceptance
B. Budgetary Constraints
C. STP reliability
D. IP convergence
Correct Answer: AB Section: (none) Explanation
QUESTION 93
What is the biggest issue affecting voice transport when you implement IPSec VPNs in a converged network?
A. Hop count.
B. Using G.729 as the codec.
C. Throughput considerations.
D. Ensuring only software encryption is running.
Correct Answer: C Section: (none) Explanation
QUESTION 94
What factors must be considered in the overall design when implementing an IPSec VPN for transport of voice?
A. Port numbers and added delay.
B. Added delay and added overhead.
C. Port numbers and longer dial plan.
D. Port numbers and added overhead.
E. Added overhead and longer dial plan.
Correct Answer: D Section: (none) Explanation
Explanation/Reference:
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642-436
QUESTION 95
When analyzing the WAN for IP Telephone deployment, you need to collect information from the WAN and
LAN devices. You need to determine current Bandwidth usage before rolling out a solution. From the
analysis you are performing, there are categories you can collect information from.
Select two from the list below.
A. Device information, which includes router models, memory, CPU, interface card modules versions and software versions
B. The serial numbers from the Meridian Phone System
C. The existing WAN topology, which includes logical design information and bandwidth subscription rates
D. LAN information as in the make and model of the Remote site closet Nortel equipment
E. You need all the CiscoWorks server configurations to make sure you can install Call Manager
Correct Answer: AC Section: (none) Explanation
QUESTION 96
If you were troubleshooting No Ringback Tone on ISDN-VoIP (H.323) Calls and had problem that POTS (PSTN/PBX) user places a call (through Cisco router/gateways) and does not hear ringback tone before call is answerered, what would you do?
A. Use the conf inline power command because it is not set in the terminating router.
B. Reset all the phone connections on the IP SoftPhones
C. Configure CiscoWorks CallManager to handle all errors automatically.
D. Configure the Cisco IOS global configuration command voice call send-alert in the terminating router
Correct Answer: D Section: (none) Explanation
QUESTION 97
Which preference key word assigns top precedence to a dial peer in a hunt-group?
A. 0
B. priority
C. 1
D. high
Correct Answer: A Section: (none) Explanation
Explanation/Reference:
Actualtests.com – The Power of Knowing 642-436
QUESTION 98
What are two basic parameters needed to setup a dial peer connected to the PSTN? (Choose two)
A. voice port
B. signaling type
C. interface bandwidth
D. destination pattern
Correct Answer: AD Section: (none) Explanation
Explanation/Reference:
Explanation: Depending on the call leg, a call is routed using one of the two types of dial peers: POTS-Dial peer that defines the characteristics of a traditional telephony network connection. POTS dial peers map a dialed string to a specific voice port on the local router, normally the voice port connecting the router to the local PSTN, PBX, or telephone. Voice-network-Dial peer that defines the characteristics of a packet network connection. Voice-network dial peers map a dialed string to a remote network device, such as the destination router that is connected to the remote telephony device. The following examples show basic configurations for POTS and VoIP dial peers: dial-peervoice 1 potsdestination-pattern555….port1/0:1dial-peervoice 2 voipdestination-pattern 555….session target ipv4:192.168.1.1
QUESTION 99
Users are complaining that they are unable to complete a call from 678-555-1212 to 770-555-1111 from
Router 1 to Router 2.
Select the correct answer to resolve the problem.
A. Incorrect dial-peer statement in Router 1. Actualtests.com – The Power of Knowing 642-436
B. Incorrect port statement in Router 1 pots dial peer.
C. Incorrect session-target statement in Router 2.
D. Incorrect destination-pattern in Router 1.
Correct Answer: B Section: (none) Explanation
Explanation/Reference:
Given the output the correct answer would be “Incorrect port assignment in router one”. Voice port 1/0/1 does not exist, according to the drawing on router 1, Voice port 1/0/0 is the correct port.under the router 1 dial-peer the port assignment is port 1/0/1. There is no problem with the destination-patterns (not D)
QUESTION 100
Which dial plan characteristic is most obviously improved by dropping a number translation step?
A. Availability
B. Post-dial delay
C. Scalability
D. Hierarchical design
Correct Answer: C Section: (none) Explanation
Explanation/Reference:
Explanation: Introduction This document provides a sample configuration for creating scalable dial plans for a VoIP network using IOS translation rules. As you install integrated voice and data networks, one issue frequently encountered is how to manage the numbering plans of the indial ranges at different locations. Depending on the type of exchange, signaling protocol standards and even location, the service provider could pass similar number ranges to the subscriber equipment at each remote site. If these calls are being routed back to a central site, there could be an overlap in the called numbers that originate from each of the remote sites. Since the PBX makes the routing decision based on unique called numbers, this could cause problems with automatic call distribution (ACD) queues on private branch exchange (PBX) systems . For example, calls from each site may need to be directed to particular operators who speak the local language from where the call originated. If the called numbers from each site overlap, there is not any way of identifying the origin of a call, therefore the PBX is not able to route the call to the correct ACD queue. Some remote sites may be provided with a 2-digit indial number range while other sites may have 3- or 4-digit indial ranges, so the called numbers could be from to . With these number ranges, the main site router would need configurations to handle 2-, 3- and 4-digit numbering plans. This could add to the overall complexity of the router configuration. The solution to these issues is to use IOS digit translation rules at each remote site to prepend digits to the number range that comes in from the telephone network. This then
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creates a standard numbering plan across the customer’s network and allows new sites to be gradually added without major changes to the rest of the network.
QUESTION 101
DRAG DROP
Drag each of the dial peers on the left to the phone number that it would match on the right.
A.
B.
C.
D.
Correct Answer: Section: (none) Explanation
Explanation/Reference:
Explanation:
Destination pattern 4081234″ would be an exact match for 4081234 and therefore would be the correct
answer.
Destination pattern .T” would match 4181234 not 4081234 due to length of match. ” Note:
Destination Pattern
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The destination pattern associates a dialed string with a specific telephony device. It is configured in a dial
peer by using the destination-pattern command. If the dialed string matches the destination pattern, the call
is routed according to the voice port in POTS dial peers, or the session target in voice-network dial peers.
For outbound voice-network dial peers, the destination pattern may also determine the dialed digits that the
router collects and then forwards to the remote telephony interface, such as a PBX, a telephone, or the
PSTN. You must configure a destination pattern for each POTS and voice-network dial peer that you
define on the router.
The destination pattern can be either a complete telephone number or a partial telephone number with
wildcard digits, represented by a period (.) character. Each “.” represents a wildcard for an individual digit
that the originating router expects to match. For example, if the destination pattern for a dial peer is defined
as “555….”, then any dialed string beginning with 555, plus at least four additional digits, matches this dial
peer. In addition to the period (.), there are several other symbols that can be used as wildcard characters
in the destination pattern. These symbols provide additional flexibility in implementing dial plans and
decrease the need for multiple dial peers in configuring telephone number ranges.
Fixed- and Variable-Length Dial Plans
Fixed-length dialing plans, in which all the dial-peer destination patterns have a fixed length, are sufficient
for most voice networks because the telephone number strings are of known lengths. Some voice
networks, however, require variable-length dial plans, particularly for international calls, which use
telephone numbers of different lengths. If you enter the timeout T-indicator at the end of the destination
pattern in an outbound voice-network dial peer, the router accepts a fixed-length dial string and then waits
for additional dialed digits. The timeout character must be an uppercase T. The following dial-peer
configuration shows how the T-indicator is set to allow variable-length dial strings:
In the example above, the router
dial-peervoice 1 voipdestination-pattern2222Tsessiontarget ipv4:10.10.1.1 accepts the digits 2222, and
then waits for an unspecified number of additional digits. The router can collect up to 31 additional digits,
as long as the interdigit timeout has not expired. When the interdigit timeout expires, the router places the
call. The default value for the interdigit timeout is 10 seconds. Unless the default value is changed, using
the T-indicator adds 10 seconds to each call setup because the call is not attempted until the timer has expired (unless the # character is used as a terminator). You should therefore reduce the voice-port interdigit timeout value if you use variable-length dial plans. You can change the interdigit timeout by using the timeouts inter-digit
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voice-port command.
QUESTION 102
When does an IP Phone receive the ring tones on the phone?
A. The phone downloads the wave file on boot.
B. The phone downloads based upon user selection.
C. The phone downloads the wave file on every request.
D. The phone downloads based on CallManager request.
Correct Answer: A Section: (none) Explanation
QUESTION 103
Which two tools are most appropriate for configuring 4,000 IP Phones prior to deploying the phones and allow phones to auto register? (Choose two.)?
A. ART
B. BAT
C. AST
D. TAPS Actualtests.com – The Power of Knowing
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Correct Answer: BD Section: (none) Explanation
QUESTION 104
Which field can be manually entered into the database when using the BAT tool, but not when using the TAPS tool??
A. Partiition
B. Directory number
C. Device MAC address
D. Calling Search Space
Correct Answer: C Section: (none) Explanation
QUESTION 105
Name two factors that will need to be considered and may provide a hurdle to move forward with your VOIP rollout and IP telephony implementation:
A. Money Flow issues.
B. Business Requirement
C. Technical Constraints
D. Budget constraints
E. Project Management Meetings
Correct Answer: AC Section: (none) Explanation
QUESTION 106
When the Directories button on the 7960 phone is pressed, what does the 7960 use to retrieve the Dirctory information?
A. XML
B. SQL
C. LDAP
D. Skinny
Correct Answer: A Section: (none) Explanation
QUESTION 107
Which file does the BAT tool use to import users into the CallManager database?
A. CSV
B. Microsoft Word Actualtests.com – The Power of Knowing 642-436
C. Microsoft Excel D. Tab-delimited text file
Correct Answer: A Section: (none) Explanation
QUESTION 108
You are the network technician at Certkiller .com. Your newly appointed Certkiller trainee wants to know
what the attributes of a scalable dialing plan are.
What will your reply be? (Choose four)
A. Logic distribution
B. Hierarchical design
C. Simplicity in provisioning
D. Reduction in pre-dial delay
E. reduction in post-dial delay
Correct Answer: ABCE Section: (none) Explanation
QUESTION 109
What happens if no incoming dial peer matches a router or gateway?
A. The incoming call leg takes an alternate path.
B. The incoming call leg matches the default dial peer.
C. The incoming call leg sends a busy to the originator.
D. The incoming call leg is denied and the call is dropped.
Correct Answer: B Section: (none) Explanation
QUESTION 110
A 9 digit number must be dialed to reach numbers on the PSTN. What process makes sure that the first 9 digit is not transmitted as part of the called number?
A. digit alternating
B. digit masking
C. digit manipulation
D. digit seizing
Correct Answer: C Section: (none) Explanation
QUESTION 111
SIMULATION
Actualtests.com – The Power of Knowing 642-436
Network topology exhibit: You’re a CCNP certified employee at Deutsche Telekom. You are working with the Certkiller .com Telephone Company, in Rammstein Germany, to assist them to set up a simple configuration to demonstrate an FXS-to-PSTN connection across an IP network. The Ethernet interfaces and routing protocols are already configured on both routers. The Certkiller 1 router will act a s gateway to the PSTN to the PSTN and is correctly configured for this task. You are required to add the voice portion to the Certkiller 2 router. Configure the pots and VoIP dial peers and insure that the pots telephone on the Certkiller 2 router can reach the PSTN connected to the Certkiller 1 router. The analog telephone is connected to voice port 1/0/0. The customer uses the access code 9 to dial out the PSTN. Insure that the Certkiller 1 and Certkiller 2 recognize the dialed digits as standard E.164 numbers. The telephone number of the pots telephone connected to the Certkiller 1 router is 49648455554321. After the configuration is complete you may click on the telephone to check your configuration. If the call is successful, you will get a ringing message and a completed call. If the call is unsuccessful, you will get a busy signal. The Certkiller 1 router has the following configured ports: Ethernet: 0/0 172.16.1.11 255.255.255.0 To configure the router click on a host icon that is connected to a router by a serial console cable.
A.
B.
C.
D.
Correct Answer: Section: (none) Explanation
Explanation/Reference:
Explanation: en config t dial-peervoice 1 pots destination-pattern 6495551212 port1/0/0 dial-peervoice 2 voip destination-pattern +9T sessiontarget ipv4:172.16.1.11
Explanation: Reference CVoice ver 4.1 class books page 4-21 and Cisco Voice over Frame Relay, ATM, and IP (from Cisco Press second printing May 2002) page 225. The
Actualtests.com – The Power of Knowing 642-436
“+” sign is optionaland is usedas the first digit to indicate an E.164 standard number. We place the “+” in front if the 9 digit in the voip dial peer because the question states to make sure that Certkiller 1 and Certkiller 2 recognize the dialed digits as standard E.164 numbers. The pots phone will not be calling itself so therefore the logical placement of the “+” would be in front of the 9 digit in the voip dial peer. Note 1: The simulation does not allow the use of the “register” command. Note 2: Instead of destination-pattern +9T it might be destination-pattern 9+T. You will know when you have it correct if, after clicking on the telephone icon, you receive a message that says ringing.
QUESTION 112
Which gateway interface connects to the standard station port of a PBX?
A. FXS
B. E&M
C. POTS
D. FXO
Correct Answer: D Section: (none) Explanation
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