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QUESTION 225
The Cisco CallManager dial plan architecture is set up to handle two general types of calls. What are they? (Choose all that apply.)
A. External calls through a SAA Gateway
B. External calls through a PSTN gateway or to another Cisco CallManager cluster
C. Internal calls From the source router to the PBX-1
D. Internal calls to Cisco IP phones registered to the Cisco CallManager cluster itself-
E. Internal calls from the IP SoftPhone to the 7200 VXR2
F. External calls through the last downstream CallManager phone set.

Correct Answer: BD Section: (none) Explanation
QUESTION 226
From the list below, what protocol is designed to provide end-to-end network transport functions for applications transmitting real-time data, such as audio, video, or simulation data, over Multicast or Unicast network services.
Actualtests.com – The Power of Knowing
642-436

A. CAM
B. IPTV
C. STP
D. RTP
E. DMVRP
F. PIM
G. IS-IS

Correct Answer: D Section: (none) Explanation
QUESTION 227
Which statement represents the definition of an MGCP endpoint?
A. The interconnection between packet and traditional telephone networks.
B. Any analog telephony device (PBX, switch, ect).
C. IP hones
D. The gatekeepers in a VoIP network.

Correct Answer: A Section: (none) Explanation
Explanation/Reference:
Explanation: A typical MGCP gateway environment connects on one side with a public switched telephone network (PSTN), and on the other side with an IP network. Specialized call agent applications control the flow of media data across the distributed environment. Call agents determine the route that data follows as it flows through the system. Multiple call agents can control call processing and data transfer. These call agents use a separate protocol to synchronize with each other and to send coherent commands to modules under their control. MGCP assumes a connection model where the basic constructs are endpoints and connections. Endpoints are sources or sinks of data and could be physical or virtual. Examples of physical endpoints are:
*
An interface on a gateway that terminates a trunk connected to a PSTN switch (e.g., Class 5, Class 4,
etc.). A gateway that terminates trunks is called a trunk gateway.
*
An interface on a gateway that terminates analog POTS connection to a phone, key system, PBX, etc. A
gateway that terminates residential POTS lines (to phones) is called a residential gateway.
An example of a virtual endpoint is an audio source in an audio- content server. Creation of physical
endpoints requires hardware installation, while creation of virtual endpoints can be done by software.

QUESTION 228
What are the three components in an MGCP environment? (Choose three)
A. Gateway Actualtests.com – The Power of Knowing 642-436
B. Gatekeeper
C. Endpoint
D. Call agent
E. Proxy server

Correct Answer: ACD Section: (none) Explanation Explanation/Reference:
Explanation: A typical MGCP gateway environment connects on one side with a public switched telephone network (PSTN), and on the other side with an IP network. Specialized call agent applications control the flow of media data across the distributed environment. Call agents determine the route that data follows as it flows through the system. Multiple call agents can control call processing and data transfer. These call agents use a separate protocol to synchronize with each other and to send coherent commands to modules under their control. Each call agent usually controls a set of gateway applications, including at least one media gateway. Media gateways convert media signals to an appropriate format depending on whether the signals are directed to a circuit switched network format or a packet switched network. Media gateways primarily perform audio signal translation functions in accordance with call agent commands. Note: Gateways connected to an SS7 controlled network must also include at least one signaling gateway for controlling SS7 signaling. The MGCP connection model consists of endpoints and connections. Endpoints represent physical or virtual sources through which data can flow (for example, PSTN ports on a media gateway). Call agents combine sets of endpoints under their control to create point-to-point or multipoint connections. Connections provide data paths for transferring and processing the data that flows through the gateway environment. In the MGCP model, call control intelligence resides in the call agents, not in the media gateways. In effect, the MGCP standard defines a master/slave relationship between call agents and media gateways, where gateways execute commands sent by the call agents. MGCP is a client-server protocol. The CA handles all aspects of setting up calls to and from endpoints. CAs or control servers provide the feature capabilities that a particular endpoint will be able to use. Endpoints connected to different CAs will likely have a different set of features they can use. Since all of the call control features are in the control server, each control server vendor decides which features are most important, and therefore different control server vendors differ in “essential features.” MGCP relies on a control server, or call agent (CA), to control call progression, tones to apply, and call characteristics. MGCP endpoints carry out instructions from the CA, which controls how calls proceed.
QUESTION 229
With regard to MGCP, what is a call?
A. It is the path between two telephones. Actualtests.com – The Power of Knowing 642-436
B. It is the RTP sessions between the endpoints.
C. It is a connection between an endpoint and the call agent.
D. It is two or more endpoints sharing the same Call ID and the same media stream.

Correct Answer: D Section: (none) Explanation
QUESTION 230
You are the network engineer at Certkiller .com. You are deploying an IP telephony solution using MGCP.
The call agent expects the gateway to use UDP port 2427 but an application on the Certkiller network is
already using that port. You want to use port 4662 instead.
Which command would allow you to change the UDP port that the call agents and gateway communicate
on?

A. Router(config)# mgcp UDP 4662
B. Router(config)# mgcp gateway 4662
C. Router(config)# mgcp call-agent 4662
D. Router(config-dial-peer)#application MGCPAPP 4662
E. Router(config)# mgcp default-package gm-package 4662

Correct Answer: C Section: (none) Explanation QUESTION 231
You are the Voice engineer at Certkiller .com. Numerous Certkiller users complain that they are unable to
complete calls through the MGCP network. You want to verify the extent of the problem by reviewing a
count of the successful and unsuccessful control commands.
Which command should you use?

A. show mgcp
B. show mgcp count
C. show mgcp statistics
D. show call active voice
E. show call history voice

Correct Answer: C Section: (none) Explanation
QUESTION 232
You are the network engineer at Certkiller .com. You want to verify the registration of the gateway with the
call agent.
Which show command should you use?

A. show mgcp Actualtests.com – The Power of Knowing 642-436
B. show call agent
C. show gateway mgcp
D. show endpoint mgcp
E. show call active voice

Correct Answer: A Section: (none) Explanation
QUESTION 233
What identifies an MGCP endpoint?
A. A two part identifier that consists of thetelephone number and local name of the user.
B. A two part identifier that consists of thetelephone number and remote name of the user.
C. A two part identifier that consists of the domain name of the user and the IP address of the gateway.
D. A two part identifier that consists of the local name of the user and the domain name of the gateway.

Correct Answer: D Section: (none) Explanation
QUESTION 234
DRAG DROP Assume a SIP voice network. Drag each characteristic to the type of SIP call setup the characteristics best describes.

A.
B.
C.
D.

Correct Answer: Section: (none) Explanation
Explanation/Reference:
Actualtests.com – The Power of Knowing 642-436

Explanation:
“Server reports back to a UA with destination coordinates” is a function of the a Redirect Server (p. 6-94 of
CVoice version 4.1 class books). Reference pages 6-91 – 6-94 of CVoice version 4.1 class books.

QUESTION 235

For Scalability and ease of management, the decision has been made to centralize the location of all SIP
endpoints in servers.
When phone A wants to call Phone B. it asks Certkiller A how to find Phone B.
What kind of device is Certkiller A?

A. Proxy
B. Redirect
C. Registrar
D. User agent client
E. User agent server

Correct Answer: B Section: (none) Explanation
Explanation/Reference:
Explanation:
SIP ServersSIP servers include:

1.
Proxy server-the proxy server is an intermediate device that receives SIP requests from a client and then forwards the requests on the client’s behalf. Basically, proxy servers receive SIP messages and forward them to the next SIP server in the network. Proxy servers can provide functions such as authentication, authorization, network access control, routing, reliable request retransmission, and security.

Actualtests.com – The Power of Knowing 642-436

2.
Redirectserver-Providesthe client with information about the next hop or hops that a message should take and then the client contacts the next hop server or UAS directly.

3.
Registrar server-Processes requests from UACs for registration of their current location. Registrar servers are often co-located with a redirect or proxy server. Redirect server: A redirect server is a server that accepts a SIP request, maps the address into zero or more new addresses and returns these addresses to the client
QUESTION 236
What is the function of a SIP location server?
A. Resolves active endpoint addresses
B. Routes service requests
C. Acquires active endpoint addresses
D. Resolves text addresses to IP addresses

Correct Answer: A Section: (none) Explanation
Explanation/Reference:
Explanation: The correct answer should be “Resolves active endpoint addresses” based on the following from CVoice version 4.1 class books on pages 6-84 and 6-89. A Location Server is defined (on page 6-84) as: An abstraction of a service providing address resolution services to SIP proxy or redirect servers. A location server embodies mechanisms to resolve addresses. On page 6-89 a Registrar Server is described as a server that acquires addresses for the location server.
QUESTION 237

Given the SIP network shown in the diagram identify which three actions are initiated by the UAC (user agent client)? (Choose three)
A. Initiates a SIP requests.
B. Originated the BYE method to indicate call termination.
C. Originates the ACK method to indicate that it has receives a response to its invitation.
D. Contacts the user when a SIP invitation is receives.
E. Returns a response on behalf of the user to the invitation originator.

Correct Answer: ABC Section: (none) Explanation
Explanation/Reference:
Actualtests.com – The Power of Knowing 642-436
Explanation:
1.4.4 SIP Invitation A successful SIP invitation consists of two requests, INVITE followed by ACK. The INVITE (Section 4.2.1) request asks the callee to join a particular conference or establish a two-party conversation. After the callee has agreed to participate in the call, the caller confirms that it has received that response by sending an ACK (Section 4.2.2) request. If the caller no longer wants to participate in the call, it sends a BYE request instead of an ACK.

QUESTION 238

Which characteristic is true about SIP protocol messages?
A. Binary
B. Text-based
C. Numeric
D. Encrypted

Correct Answer: B Section: (none) Explanation
Explanation/Reference:
Explanation:
Format
All SIP messages are either requests from a server or client or responses to a request. The messages are
formatted according to RFC 822,
“Standard for the format of ARPA internet text messages.” For all messages, the general format is:

1.
A start line

2.
One or more header fields

Actualtests.com – The Power of Knowing 642-436

3.
An empty line
4.
A message body (optional)
Each line must end with a carriage return-line feed (CRLF).

QUESTION 239
Upon which protocol model is the SIP protocol based?
A. HTML
B. H.323
C. Q.931
D. MGCP
E. HTPP/WWW

Correct Answer: E Section: (none) Explanation
QUESTION 240
With regard to SIP and SDP, which of the following statements is true?
A. SIP is similar to RAS and SDP is similar to RTP
B. SIP is similar to RTP and SDP is similar to RAS
C. SIP is similar to H.225 and SDP is similar to H.245
D. SIP is similar to H.245 and SDP is similar to H.323
E. SIP is similar to H.323 and SDP is similar to H.225

Correct Answer: C Section: (none) Explanation
QUESTION 241
You are the network engineer at Certkiller .com. You are configuring a connection to a SIP proxy server. Which command would you use to specify the IP address of the server?
A. sip-ua sip-server ipv4:1.2.3.4
B. sip-ua sip-server target:1.2.3.4
C. dial-peer voice 1 voip session target sip:1.2.3.4
D. dial-peer voice 1 voip session target sip-server:1.2.3.4

Correct Answer: A Section: (none) Explanation
Explanation/Reference:
Actualtests.com – The Power of Knowing 642-436
QUESTION 242
Which of the following call control models are based on decentralized call control? (Choose two.)
A. SIP
B. CAS
C. H.323
D. Q.931
E. MGCP

Correct Answer: AC Section: (none) Explanation
QUESTION 243
You are meeting with a customer that has deployed IP telephony at their headquarters location. They would like to roll out IP telephony to their regional office as well. They are now using the G.711 codec at headquarters. They want to be able to maximize the number of calls carried without impacting voice quality or forcing a WAN upgrade. Which codec would be appropriate for their WAN?
A. G.726
B. G.723.1
C. G.711
D. G.729B

Correct Answer: D Section: (none) Explanation
QUESTION 244
Examine the output. ccm-manager mgcp ! mgcp 5036 ! voice-port 1/0/0 ! voice-port 1/0/1 ! dial-peer voice 1 pots application MGCPAPP port 1/0/0 ! dial-peer voice 2 ports application MGCPAPP
Actualtests.com – The Power of Knowing 642-436
port 1/0/1 ! Your customer has sent you their MGCP gateway configuration. They are unable to get the gateway to communicate with the call agent. What command needs to be inserted to resolve the problem?
A. ccm-manager mgcp 172.16.1.1
B. mgcp call-agent 172.16.1.1
C. application MGCPAPP 172.16.1.1
D. mgcp 5036 172.16.1.1

Correct Answer: B Section: (none) Explanation
QUESTION 245
You are the Voice technician at Certkiller .com. Your newly appointed Certkiller trainee wants to know what
request method initiates a SIP call setup.
What will your reply be?

A. ACK
B. INVITE
C. OPTIONS
D. REGISTER
E. DISCOVER

Correct Answer: B Section: (none) Explanation
QUESTION 246

hostname CK1 ! interface serial 0/0 ip address 172.16.1.1 255.255.255.248
!
Actualtests.com – The Power of Knowing 642-436
controller t1 framing esp clock source line linecode b8zs ds0-group 1timeslots 1-24 type e&m-wink-start ! voice-port 1/0:1 ! dial-peer voice 1 voip destination-pattern 404555…. session-target ipv4:172.16.1.6 ! dial-peer voice 2 ports destination-pattern 201555…. port 1/0:1 hostname CK2 ! interface serial 0/0 ip address 172.16.1.6 255.255.255.248 ! controller t1 framing esp clock source line linecode b8zs ds0-group 1timeslots 1-24 type e&m-wink-start ! voice-port 1/0:1 ! dial-peer voice 1 voip destination-pattern 201555…. session-target ipv4:172.16.1.1 ! dial-peer voice 2 ports destination-pattern 404555…. port 1/0:1 Use the figure above to answer this question. When extension 201-555-1000 dials 404-555-1200, how are digits manipulated in R1 so they are presented correctly at CK2 ?
A. When extension 201-555-1000 dials 404-555-1200, the digits 404-555 are stripped off prior to matching the outbound POTS dial peer.
B. When extension 202-555-1000 dials 404-555-1200, the digits 404-555 are stripped off by the connection trunk and CK2 receives only 1200.
C. When extension 201-555-1000 dials 404-555-1200, the outbound VoIP dial peer is matched and all digits are sent.
D. When extension 201-555 1000 dials 404-555-1200, CK1 collects the 1200 and Actualtests.com – The Power of Knowing 642-436 prepends the tie-line digits 404555. That number is matched to a VoIP dial peer and sent to the appropriate address.

Correct Answer: D Section: (none) Explanation
QUESTION 247
How is CAS different on E1 and T1?
A. T1 has more signaling channels.
B. E1 CAS signaling is out-of-band while T1 is in-band.
C. E1 uses robbed-bit signaling.
D. T1 uses the D channel for CAS signaling.

Correct Answer: B Section: (none) Explanation
QUESTION 248
When impendence is mismatched in a two-wire to four-wire circuit, what is the common result?
A. glare
B. jitter
C. echo
D. clipping

Correct Answer: C Section: (none) Explanation
QUESTION 249
In the connection between a Cisco router and an E&M port on a PBX, which side is generally the Cisco side?
A. loop start
B. trunk circuit
C. switch port
D. signaling unit

Correct Answer: D Section: (none) Explanation
Explanation/Reference:
Explanation: Analog trunk circuits connect automated systems, such as a private branch exchange (PBX) and the network, such as a central office (CO). The most common form of analog trunking is the E&M interface. E&M Signaling is commonly refer to as “ear & mouth” or “recEive and transMit”, but its origin comes from the term earth and magnet. Earth represents electrical ground and magnet represents the electromagnet used to generate
Actualtests.com – The Power of Knowing 642-436
tone. E&M signaling defines a trunk circuit side and a signaling unit side for each connection similar to the data circuit-terminating equipment (DCE) and data terminal equipment (DTE) reference type. Usually the PBX is the trunk circuit side and the telco, CO, channel-bank, or Cisco voice enabled platform is the signaling unit side. Note:Cisco’s analog E&M interface functions as the signaling unit side, so it expects the other side to be a trunk circuit.
QUESTION 250
You are the network technician at Certkiller .com. Your newly appointed Certkiller trainee wants to know
which signal types are used by E&M.
What will your reply be?

A. wink start, delay start, and loop start
B. wink start, loop start, and immediate start
C. wink start, delay start, and immediate start
D. delay start, and loop start, and immediate start

Correct Answer: C Section: (none) Explanation
QUESTION 251

In an effort to consume less bandwidth across the WAN, the decision was made at Certkiller to change the voice packet size. They changed from two voice frames per packet to one voice frame per packet. What effect did this have on Certkiller ‘s voice traffic?
A. Per call bandwidth consumption decreased and end-to-end delay increased.
B. Per call bandwidth consumption increased and end-to-end delay decreased.
C. Per call bandwidth consumption decreased and end-to-end delay decreased.
D. Per call bandwidth consumption increased and end-to-end delay also increased.
E. There was no effect on voice traffic.

Correct Answer: B Section: (none) Explanation
Explanation/Reference:
Actualtests.com – The Power of Knowing 642-436
QUESTION 252
You have been forwarded some questions by a prospective VoIP customer who would like to know the
Cisco default sample size for the G.729 codec.
What is it?

A. 40 ms
B. 30 ms
C. 20 ms
D. 10 ms

Correct Answer: D Section: (none) Explanation
Explanation/Reference:
Explanation: Codec Sample Interval (ms) This is the sample interval at which the codec operates. For example, the
G.729 coder operates on sample intervals of 10 ms, corresponding to 10 bytes (80 bits) per sample at a bit rate of 8 Kbps. (codec bit rate = codec sample size / codec sample interval).
QUESTION 253
What component can be used to compensate for jitter?
A. FIFO queuing
B. Ethernet hubs
C. DSP algorithms
D. Playout delay buffer
E. Transmission medium

Correct Answer: D Section: (none) Explanation
QUESTION 254
You are the network engineer at Certkiller .com. Certkiller has its headquarters in New York and a branch
office in Delaware. Users at headquarters must be able to call users at the branch office and users at the
branch office must be able to call headquarters.
How many dial peers must you configure to meet these requirements?

A. 1
B. 2
C. 3
D. 4
E. none Actualtests.com – The Power of Knowing 642-436

Correct Answer: D Section: (none) Explanation

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QUESTION 113
What from the list below combines voice mail, e-mail, and fax into a single application suite where a single application can be used to store and retrieve entire suite of message types?
A. PBSX Listing
B. Name Resolution IPTC
C. Call Manager 3.01
D. Cat 4000 STP v3
E. Unified messaging

Correct Answer: E Section: (none) Explanation
QUESTION 114
In a distributed call processing model, which three are located at each site? (Choose three.)
A. gatekeeper
B. voice messaging
C. media resources
D. Cisco CallManager cluster

Correct Answer: BCD Section: (none) Explanation
Explanation/Reference:
Actualtests.com – The Power of Knowing 642-436
QUESTION 115
What can be used not only to restrict dialing, but also to identify a subset of a subset of a wildcard pattern (when using the @ wildcard in the North American Dialing Plan)?
A. An IS sheet
B. A ACL
C. A Route Filter
D. A DN top

Correct Answer: C Section: (none) Explanation QUESTION 116
What does the Digit Discard Instruction of PreDot do to the pattern 9.2148134444?
A. prefix a 9 before the “-” if none is dialled
B. discard 2148134444 and send the 9 access code
C. only collect the first four digits counting right to left.
D. change it to 2148134444 before presenting it to the PSTN

Correct Answer: D Section: (none) Explanation
QUESTION 117
What is the key element in call admission control when interconnecting CallManager sites via the IP WAN?
A. gatekeepers
B. voice messaging
C. media resources
D. call processing agents

Correct Answer: A Section: (none) Explanation
QUESTION 118
Certkiller has its headquarters in New York and branch offices in Delaware, Delhi and Dakar.
Headquarters and the Delaware branch office has IP Phones. The other two offices have analog phones
that are connected to FXS port on the router in the site′s administration building. Users at these offices
complain that they are unable to call out in the PSTN or to each other.
You receive the following output:
2611#s voice port 1/0/0

Actualtests.com – The Power of Knowing
642-436

Foreign Exchange Station 1/0/0 Slot is 1, Sub-unit is 0, Port is 0
Type of VoicePort is FXS
Operation State is DORMANT
Administrative State is UP
No Interface Down Failure
Description is not set
Noise Regeneration is enabled
Non Linear Processing is enabled
Non Linear Mute is disabled
Non Linear Threshold is -21 dB
Music On Hold Threshold is Set to 38 dBm
In Gain is Set to 0 dB
Out Attention is Set to 3 dB
Echo Cancellation is enabled
Echo Cancellation NLP mute is disabled
Echo Cancellation NLP threshold is -21 dB
Echo Cancel Coverage is set to default
Playout-delay Mode is set to default
Playout-delay Nominal is set to 60 ms
Playout-delay Maximal is set to 200 ms
Playout-delay Minimum mode is set to default, value 40 ms Playout-delay Fax is set to 300 ms
Connection Mode is normal
Connection Number is not set

Initial Time Out is set to 10 s Interdigit Time Out is set to 10 s Call Disconnect Time Out is set to 60 s Ringing Time Out is set to 180 s Wait Release Time Out is set to 30 s Companding Type is u-law Region Tone is set for US Analog Info Follows: Currently processing none Maintenance Mode Set to None (not in mtc mode) Number of signaling protocol errors are 0 Impedance is set to 600r Ohm Station name None, Station number None Voice card specific Info Follows: Signal Type is groundStart Ring Frequency is 25 Hz Hook Status is On Hook Ring Active Status is inactive Ring Ground Status is inactive Tip Ground Status is inactive Digit Duration Status is inactive
Actualtests.com – The Power of Knowing 642-436
Digit Duration Timing is set to 100 ms InterDigit Duration Timing is set to 100 ms No disconnect acknowledge Ring Cadence is defined by CPTone Selection Ring Cadance are [20 40] * 100 msec 2611# What is the cause of this problem?
A. The cptone is incorrect
B. The dial-type is incorrect
C. The signal type is incorrect
D. The playout-delay is incorrect
E. The disconnect-ack is incorrect

Correct Answer: C Section: (none) Explanation
QUESTION 119
You are the Voice technician at Certkiller .com. Your newly appointed Certkiller trainee wants to know on
what type of port you would set impedance.
What will your reply be?

A. T1
B. E1
C. FXS
D. FXO
E. E&M

Correct Answer: E Section: (none) Explanation
Explanation/Reference:
Source Cisco CVOICE book – page 3-48 VoicePortTuning Parameters E&M voice port parameters
-input-gain
-no echo-cancel enable
-impedance FXO voice port parameters
-echo-cancel coverage – output-attenuation
QUESTION 120
Which type of delay is caused by the line speed of the interface?
A. Queuing delay
B. Serialization delay
C. Propagation delay Actualtests.com – The Power of Knowing 642-436
D. Packetiziation delay

Correct Answer: B Section: (none) Explanation
Explanation/Reference:
Explanation: Serialization Delay Serialization delay (?n) is the fixed delay required to clock a voice or data frame onto the network interface, and It is directly related to the clock rate on the trunk. Remember that at low clock speeds and small frame sizes the extra flag needed to separate frames is significant. Queuing/Buffering Delay After the compressed voice payload is built, a header is added and the frame is queued for transmission on the network connection. Because voice should have absolute priority in the router/gateway, a voice frame must only wait for either a data frame already playing out, or for other voice frames ahead of it. Essentially the voice frame is waiting for the serialization delay of any preceding frames in the output queue. Queuing delay (.n) is a variable delay and is dependent on the trunk speed and the state of the queue. Clearly there are random elements associated with the queuing delay. PacketizationDelay Packetization delay(?n) is the time taken to fill a packet payload with encoded/compressed speech. This delay is a function of the sample block size required by the vocoder and the number of blocks placed in a single frame. Packetization delay may also be called Accumulation delay, as the voice samples accumulate in a buffer before being released.
QUESTION 121
Certkiller has its headquarters in New York and branch offices in Delaware, Detroit and Denver.Each office has an analog phone at each location. These phones are connected to an FXS port on the on-site router. The Finance department at the Denver office is unable to make any phone class from these analog phones. You receive the following output: 2611#s voice port 1/0/0 Foreign Exchange Station 1/0/0 Slot is 1, Sub-unit is 0, Port is 0 Type of VoicePort is FXS Operation State is DORMANT Administrative State is UP No Interface Down Failure Description is not set Noise Regeneration is enabled Non Linear Processing is enabled Non Linear Mute is disabled Non Linear Threshold is -21 dB Music On Hold Threshold is Set to 38 dBm In Gain is Set to 0 dB
Actualtests.com – The Power of Knowing
642-436

Out Attention is Set to 3 dB Echo Cancellation is enabled Echo Cancellation NLP mute is disabled Echo Cancellation NLP threshold is -21 dB Echo Cancel Coverage is set to default Playout-delay Mode is set to default Playout-delay Nominal is set to 60 ms Playout-delay Maximal is set to 200 ms Playout-delay Minimum mode is set to default, value 40 ms Playout-delay Fax is set to 300 ms Connection Mode is normal Connection Number is not set Initial Time Out is set to 10 s Interdigit Time Out is set to 10 s Call Disconnect Time Out is set to 60 s Ringing Time Out is set to 180 Wait Release Time Out is set to 30 s Companding Type is u-law Region Tone is set for US Analog Info Follows: Currently processing none Maintenance Mode Set to None (not in mtc mode) Number of signaling protocol errors are 0 Impedance is set to 600r Ohm Station name None, Station number None Voice card specific Info Follows: Signal Type is groundStart Ring Frequency is 25 Hz Hook Status is On Hook Ring Active Status is inactive Ring Ground Status is inactive Tip Ground Status is inactive Digit Duration Status is inactive Digit Duration Timing is set to 100 ms InterDigit Duration Timing is set to 100 ms No disconnect acknowledge Ring Cadence is defined by CPTone Selection Ring Cadance are [20 40] * 100 msec 2611# What is the cause of this problem?
A. The cptone is incorrect
B. The dial-type is incorrect
C. The signal type is incorrect
D. The playout-delay is incorrect
E. The disconnect-ack is incorrect Actualtests.com – The Power of Knowing 642-436

Correct Answer: C Section: (none) Explanation
QUESTION 122
You are the Voice technician at Certkiller .com. Your newly appointed Certkiller trainee wants to know what
types of trunks Cisco support with the connection trunk command.
What will your reply be? (Choose three)

A. FXS to FXS trunks, FXS to FXO trunks, and FXS to E&M trunks
B. FXS to FXS trunks, FXS to FXO trunks, and E&M to E&M trunks
C. FXS to FXS trunks, FXO to FXO trunks, and E&M to E&M trunks
D. FXO to FXS trunks, FXO to FXO trunks, and E&M to E&M trunks
E. FXS to FXS trunks, FXS to E&M trunks, and E&M to E&M trunks

Correct Answer: B Section: (none) Explanation
QUESTION 123
You are the voice technician at Certkiller .com. Certkiller has its offices in Great Britain. You need to install
a Cisco router to support IP Telephony services with direct-connected analog phones. You need to
emulate the local PSTN provider.
What FXS port parameter do you need to change?

A. Pulse
B. Signal
C. Cptone
D. Busyout
E. Description

Correct Answer: C Section: (none) Explanation
QUESTION 124
You are the network technician at Certkiller .com. Your newly appointed Certkiller trainee wants to know
when glare occurs.
What will your reply be?

A. When echo cancellers fail to synchronize.
B. When two phones go off-hook at the same time.
C. When two optical wavelengths collide in the same fiber.
D. When both ends of a telephone line or trunk experience echo.
E. When both ends of a telephone line or trunk are seized by different users.

Correct Answer: E Section: (none) Explanation
Explanation/Reference:
Actualtests.com – The Power of Knowing 642-436
QUESTION 125
You are the Voice technician at Certkiller .com. The Certkiller network uses VoIP. Your newly appointed
Certkiller trainee wants to know what the modes of the playout delay buffer are.
What will your reply be?

A. Percent and Unit.
B. Nominal and Full.
C. Dynamic and Static.
D. Smooth and Serrated.
E. Minimum and Maximum.
Correct Answer: C Section: (none) Explanation

QUESTION 126
You are the network engineer at Certkiller .com. Certkiller has its headquarters in New York and a branch office in Delaware. In the branch office, one VoIP dial-peer has been configured to point to headquarters over a low speed serial link. You want to limit the maximum number of concurrent calls to 3. Which command would you use?
A. interface serial 3/3 ip rsvp bandwidth 3
B. interface serial 3/3 max-con 3
C. dial-peer voice 1000 voip max-conn 3
D. dial-peer voice 1000 voip max-concurrent 3
E. dial-peer voice 1000 voip ip rsvp neighbor 3

Correct Answer: C Section: (none) Explanation
QUESTION 127
You are the network engineer at Certkiller .com. Certkiller has its headquarters in New York and branch
offices in Delaware, Detroit and Denver. You have deployed VoIP over the Certkiller WAN. Certkiller user
at headquarters complain that early in the day, the quality of calls between headquarters and the branch
offices is very good, but as the day progresses and more calls are placed to the branch offices, the quality
degrades.
The Certkiller network is using RSVP. The WAN bandwidth to the branch offices

Actualtests.com – The Power of Knowing
642-436

allows 4 calls to the Delaware office, 6 calls to the Detroit office, and 8 calls to the Denver office. You want
to verify the configuration of Call Admission Control on the headquarters router.
What command should you use?

A. show call cac conf
B. show call rsvp-sync logs
C. show call rsvp-sync conf
D. show call rsvp-sync stats
E. show call rsvp-sync events

Correct Answer: C Section: (none) Explanation
QUESTION 128

Use the exhibit to answer the following questions.
When a call is placed from extension 1001 to 555-2212, which outbound dial peer is matched?

A. dial-peer voice 5 voip destination-pattern55[1-5]5[01][0-4].
B. dial-peer voice 1 voip destination-pattern 55[0-1]0[1-3]..
C. dial-peer voice 2 voip destination-pattern .!5551978
D. dial-peer voice 4 voip destination-pattern55[153][19]…[19][19][1] Actualtests.com – The Power of Knowing 642-436
E. dial-peer voice 3 voip destination-pattern .T

Correct Answer: E Section: (none) Explanation
QUESTION 129
What will be the outcome of an incoming VoIP call arriving at CK2 from CK1 , given the following router configurations? CK1 Configuration dial-peer voice 1 pots destination-pattern 1111 port 1/0/0 CK2 Configuration dial-peer voice 1 pots destination-pattern 2222 port 1/0/0 ! dial-peer voice 2 voip destination-pattern 1111 session-target ipv4:172.16.1.1
A. The call setup will proceed by matching dial-peer 1 pots, but will have one-way audio.
B. The call setup will fail.
C. The call setup will proceed and audio path will be established by matching the inbound call to the default dial peer.
D. The call setup will proceed, but will have no audio path.

Correct Answer: B Section: (none) Explanation
Explanation/Reference:
Based on the configuration shown the correct answer concern a VoIP call arriving at CK2 from CK1 should be “The call will fail”. The reason is that CK1 does not have a dial-peer statement defining a session target with the “session target ipv4:” command. The telephone on CK1 has no route defined on how to reach CK2 . Reference page 4-24 of CVoice version 4.1 class books. If the call was originating from CK2 to CK1 the correct answer would be “C” OR if the configurations were reversed then the answer would be “C”.
QUESTION 130
Actualtests.com – The Power of Knowing 642-436

In router CK2 which dial peer statement will match only the four extensions?
A. dial-peer voice 1 pots destination pattern 5552[5-6].
B. dial-peer voice 1 pots destination-pattern 5552[5-6][05]0
C. dial-peer voice 1 pots destination-pattern 5552.[0-5]0
D. dial-peer voice 1 pots destination-pattern 555[2-5][56]0

Correct Answer: C Section: (none) Explanation Explanation/Reference:
Note: “C” is a correct answer but “B” would also work based upon the statements here.
QUESTION 131

hostname CK1 ! interface serial0/0 ip address 172.16.1.1 255.255.255.248 ! controller t1 framing esp clock source line
Actualtests.com – The Power of Knowing 642-436
linecode b8zs ds0-group 1timeslots 1-24 type e&m-wink-start ! voice port 1/0:1 ! dial-peer voice 1 voip destination-pattern 404555….. session-target ipv4:172.16.1.6 ! dial-peer voice 2 pots destination-pattern 201555….. port 1/0:1 hostname CK2 ! interface serial0/0 ip address 172.16.1.6 255.255.255.248 ! controller t1 framing esp clock source line linecode b8zs ds0-group 1timeslots 1-24 type e&m-wink-start ! voice port 1/0:1 ! dial-peer voice 1 voip destination-pattern 201555….. session-target ipv4:172.16.1.1 ! dial-peer voice 2 pots destination-pattern 404555….. port 1/0:1
Your customer has forwarded this diagram and configuration. The customer wishes to have a connection
between its PBXs, a connection that is created and dropped as required. There is one configuration
statement missing from each router.
What are the two missing statements? (Choose two)

A. connection trunk 20155510004555… .
B. connection trunk 4045551200
C. connection tie-line 4045551200
D. connection tie-line 404555… .
E. connection tie-line 2015551000

Correct Answer: CE Section: (none) Explanation
Explanation/Reference:
Actualtests.com – The Power of Knowing 642-436
QUESTION 132

The following are the original dial peer configurations for routers CK1 and CK2 : CK1 : dial-peer voice 20 voip destination-pattern 408……. session target ipv4: 192.168.2.254 ! CK2 dial-peer voice 21 ports destination-pattern 4085554321 port 1/0/1 ! Which phones can call to the other?
A. Only Phone A can call Phone B.
B. Only Phone B can call Phone A.
C. Both phones can call each other.
D. Neither phone can call the other.

Correct Answer: A Section: (none) Explanation
QUESTION 133
How are inbound and outbound call legs handled from the perspective of the source router?
A. Only the inbound call leg is established by the source router.
B. Only the outbound call leg is established by the source router.
C. The inbound call leg and outbouond call leg are matched to the same dial peer.
D. The outbound call leg is matched first. Then, once the source is known, an inbound call leg is established.
E. The inbound call leg is matched first. Then, once the destination is known, an outbound call leg is established. Actualtests.com – The Power of Knowing 642-436

Correct Answer: E Section: (none) Explanation
QUESTION 134
You are the Voice technician at Certkiller .com. The Certkiller network uses VoIP. Your newly appointed
Certkiller trainee wants to know what the modes of the playout delay buffer are.
What will your reply be?

A. Percent and Unit.
B. Nominal and Full.
C. Dynamic and Static.
D. Smooth and Serrated.
E. Minimum and Maximum.

Correct Answer: C Section: (none) Explanation
QUESTION 135
You are the network technician at Certkiller .com. Your newly appointed Certkiller trainee wants to know
what factors affects audio quality.
What will your reply be?

A. Echo and delay variation
B. Infidelity and delay variation
C. Echo and playout delay buffer
D. Infidelity and transmission medium

Correct Answer: A Section: (none) Explanation
QUESTION 136
When a call is placed, it is routed toward the destination. Which call legs are created on that router for the call?
A. long legs
B. short legs
C. inbound call legs only
D. outbound call legs only
E. inbound and outbound call legs
Correct Answer: E Section: (none) Explanation

QUESTION 137
Actualtests.com – The Power of Knowing 642-436
Which of the following parameter is checked first when matching inbound dial peers?
A. called number (DNIS) with voice-port
B. calling number (ANI) with answer-address
C. calling number (ANI) with destination pattern
D. calling number (ANI) with incoming called-number
E. called number (DNIS) with incoming called-number

Correct Answer: E Section: (none) Explanation
QUESTION 138
What is used to translate called (DNIS) and calling automatic number identification (ANI) numbers before routing the call?
A. IR IP internetworking
B. Transitional Pattern
C. PIM Routing
D. Translation Pattern

Correct Answer: D Section: (none) Explanation
QUESTION 139
Choose all functions available to you with the IP SoftPhone. (Choose all that apply.)
A. Automatic IPX blocking ASICs
B. Displays caller name
C. Displays the caller address
D. Resets all calls every 1 hour
E. Logs calls to the call log
F. Displays caller phone number

Correct Answer: BCEF Section: (none) Explanation
QUESTION 140
What could happen if the playout delay buffer size is configured too large?
A. The overall echo on the connection may rise to unacceptable levels.
B. The overall delay on the connection may rise to unacceptable levels.
C. The overall stress on the connection may rise to unacceptable levels.
D. The overall volume on the connection may rise to unacceptable levels.
Correct Answer: B Section: (none) Explanation

Explanation/Reference:
Actualtests.com – The Power of Knowing 642-436
QUESTION 141
You are the network engineer at Certkiller .com. You have configured dial peers in a hunt group for a
Support team that answers when the number 5952215 is dialled. The Support team consists of one senior
agent and three junior agents. You want the senior agent to receive the incoming call first.
Which dial peer should you configure to point to the senior agent?

A. dial-peer voice 1 pots destination-pattern 5952215 port 1/0/0 preference 1
B. dial-peer voice 2 pots destination-pattern 5952215 port 1/0/1 preference 0
C. dial-peer voice 3 pots destination-pattern 5952215 port 1/1/0 preference 9
D. dial-peer voice 4 pots destination-pattern 5952215 port 1/1/1 preference 0

Correct Answer: D Section: (none) Explanation
Explanation/Reference:
Note: “D” is a valid answer but based on the configuration statements shown “B” would work. Both have the preference set to 0 and all other statements in each answer are correct.
QUESTION 142
You are the network engineer at Certkiller .com. Certkiller has its headquarters in New York and a branch office in Delaware. In the branch office, one VoIP dial-peer has been configured to point to headquarters over a low speed serial link. You want to limit the maximum number of concurrent calls to 3. Which command would you use?
A. interface serial 3/3 ip rsvp bandwidth 3
B. interface serial 3/3 max-con 3
C. dial-peer voice 1000 voip max-conn 3
D. dial-peer voice 1000 voip Actualtests.com – The Power of Knowing 642-436 max-concurrent 3
E. dial-peer voice 1000 voip ip rsvp neighbor 3

Correct Answer: C Section: (none) Explanation QUESTION 143
With Cisco CallManager Release 3.0, the term “route point” is replaced with which term from the list below?
A. Call list
B. Phone/list
C. Route list
D. IP_PHONE_SET
E. Phone-all
F. Route Print

Correct Answer: C Section: (none) Explanation
QUESTION 144
For very low-speed links (those with a link speed of less than 768 K), it is necessary to use techniques that
provide link fragmentation and interleaving of packets. This prevents voice traffic from being delayed
behind large data frames and hence bounds jitter.
What are two techniques that exist for this?

A. Ipng for DSL links
B. LECS for ATM links
C. Multilink PPP (MLP) for serial links
D. FRF.12 for Frame Relay

Correct Answer: CD Section: (none) Explanation
QUESTION 145
You are the network engineer at Certkiller .com. Certkiller has been using the following dial peer codec
command:
Codec g729r8
You reconfigure the dial peers with the following command:
Codec g729ar8 bytes 10
How will this reconfiguration affect the voice network bandwidth and delay characteristics? (Choose two.)

A. There will be no change. Actualtests.com – The Power of Knowing 642-436
B. Delay will increase on a per call basis.
C. Delay will decrease on a per call basis.
D. Bandwidth consumption will decrease on a per call basis.
E. Bandwidth consumption will increase on a per call basis.

Correct Answer: CE Section: (none) Explanation
QUESTION 146
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 147
If a PC connected to an IP Phone is having trouble obtaining an IP address, which setting on the phone might help resolve the problem?
A. Admin VLAN
B. Spanning Tree
C. Default Gateway
D. Forwarding Delay

Correct Answer: D Section: (none) Explanation
QUESTION 148
What are two characteristics of a distributed call processing model? (Choose two.)
A. sites connected via the PSTN
B. sites connected via the IP WAN
C. call processing agent at one site
D. call processing agent at each site

Correct Answer: BD Section: (none) Explanation
QUESTION 149
You have all ten digits being sent to your CM from the PSTN (via a gateway). If you have four digit extensions, how do you make sure that the call gets routed?
Actualtests.com – The Power of Knowing 642-436
A. update the Phone Calling Parity mask
B. -change the Route Group configuration
C. configure the GW to only collect four digits
D. change the Network Side/User Side Parameter on the gateway

Correct Answer: C Section: (none) Explanation
QUESTION 150
Cisco is making every effort to ensure that the gateways, applications, and client produced integrate and operate seamlessly with third party products. From the list below, select which protocols are being used to ensure this effort.
A. H.323
B. Session Initiation Protocol (SIP)
C. Media Gateway Control Protocol (MGCP)
D. Simple Gateway Control Protocol (SGCP)
E. All choices are correct.

Correct Answer: E Section: (none) Explanation
QUESTION 151
Which of the following statements is correct when discussing how the Cisco CallManager works with IP Phone registration? (Choose all that apply.)
A. On initial configuration, an IP phone is assigned a DSNP listing, which it loses when moved.
B. On initial configuration, an IP phone is assigned a directory number (DN), which it loses when moved
C. On initial configuration, an IP phone is assigned a DSNP, which it maintains wherever it resides
D. On initial configuration, an IP phone is assigned a directory number (DN)k, which it maintains wherever it resides.

Correct Answer: D Section: (none) Explanation
QUESTION 152
When discussing Route Groups, we know that they control specific devices such as gateways. On which protocols can gateways be based?
A. H.323
B. MGCP
C. IPNCP
D. Skinny Gateway Protocol Actualtests.com – The Power of Knowing 642-436
E. SNA
F. SAA
G. DecLat

Correct Answer: ABD Section: (none) Explanation
QUESTION 153
Which statement is true about VoIP packet loss?
A. Lost packets are simply retransmitted.
B. Even minimal packet loss causes echo.
C. IP phones can reconstruct up to three consecutive loss packets.
D. Codec algorithms can overcome minimal packet loss.

Correct Answer: D Section: (none) Explanation
QUESTION 154
Which is the best way to achieve a scalable dial plan?
A. Group numbers for a particular area.
B. Variable number of extension digits.
C. Single number prefixing.
D. Hunt groups.

Correct Answer: A Section: (none) Explanation
QUESTION 155
Which channel carries Q.931 signals in a T1 connection from a PBX to a Cisco gateway?
A. 0
B. 16
C. 24
D. 31

Correct Answer: C Section: (none) Explanation
QUESTION 156
Actualtests.com – The Power of Knowing 642-436

At what point does the MGCP call agent turn over to the residential gateways the setup of the call path?
A. After the call agent has been notified that an event has occurred at the source residential gateway.
B. After the call agent has been notified of an event and has instructed the source residential gateway to create a connection.
C. The call agent is never out of the call path setup.
D. After the call agent has sent a connection requests to both the source and destination and has relayed a modify-connection request to the source so that the source and destination can set up the call path.
E. After the call agent has forwarded session description protocol information to the destination from the source and has sent a modify connection to the destination and a create-connection request to the source.

Correct Answer: D Section: (none) Explanation QUESTION 157
What is true about H.323 endpoint call setup?
A. Endpoints always do their own call setup.
B. Endpoints require a gatekeeper to do call setup.
C. Endpoints can either do their own setup or be assisted by a gatekeeper.
D. Endpoints require a proxy server to do call setup.

Correct Answer: C Section: (none) Explanation
Explanation/Reference:
Explanation: A gatekeeper is an H.323 entity on the network that provides services such as address translation and network access control for H.323 terminals, gateways, and MCUs. Also, they can provide other services such as bandwidth management, accounting, and dial plans that can be centralized to provide salability. Gatekeepers are logically separated from H.323 endpoints such as terminals and
Actualtests.com – The Power of Knowing 642-436
gateways. They are optional in an H.323 network, but if a gatekeeper is present, endpoints must use the services provided.
QUESTION 158
Examine the example output hostname GW1 ! interface Ethernet 0/0 ip address 172.16.2.1 255.255.255.0 h323-gateway voip interface h323-gateway voip id GK1-zone1 .abc.com abc.com ipaddr 172.16.2.2 h323-gateway voip h323-id GW1 h323-gateway voip bind srcaddr 172.16.2.1 ! dial-peer voice 1 voip destination-pattern 12.12… … . session-target ras ! dial-peer voice 2 pots destination-pattern 2125551212 no register e164 ! end Choose the command that will restore communication with gatekeeper functionality to this device.
A. h323-gateway voip h323-id GK1
B. gateway
C. h323-gateway voip bind srcaddr 172.16.2.2
D. h323-gateway voip GW1-zone2.abc.com abc.com ipaddr 172.16.2.1

Correct Answer: B Section: (none) Explanation

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