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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
!
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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 191
In a VoIP environment when speech samples are framed every 20 ms. a payload of 20 bytes is generated. Assuming a total packet length of 60 bytes, what is the length of the packet header if cRTP is deploued without redundancy checks?
Actualtests.com – The Power of Knowing 642-436
A. 1 byte
B. 2 bytes
C. 3 bytes
D. 4 bytes
E. 20 bytes
F. 40 bytes

Correct Answer: B Section: (none) Explanation
QUESTION 192
What does the PBX use to determine the destination of a call?
A. An ISDN ANI packet
B. A blocked/permitted call list
C. An analysis of the dialled digits
D. Historic requests from the specific phone extension

Correct Answer: C Section: (none) Explanation
QUESTION 193
Which of the following are CS-ACELP coding schemes? (Choose two)
A. G.711
B. G.728
C. G.729
D. Q.931
E. G-729A

Correct Answer: CE Section: (none) Explanation
QUESTION 194
Which of the following is the worst-case compression delay for CD-ACELP?
A. 2.5 ms
B. 5 ms
C. 7.5ms
D. 10 ms
E. 20 ms

Correct Answer: E Section: (none) Explanation
QUESTION 195
Actualtests.com – The Power of Knowing 642-436
What type of connection is considered a call leg?
A. A digital connection
B. A virtual connection
C. A logical connection
D. A physical connection
E. A hardwired connection

Correct Answer: C Section: (none) Explanation
QUESTION 196
To which layer of the OSI model does Q.921 signaling equates to in ISDN?
A. Session
B. Network
C. Transport
D. Data-Link
E. Application

Correct Answer: D Section: (none) Explanation
QUESTION 197
Certkiller has a PBX at corporate HQ and one at a branch office. You to replace the PBX-to-PXB TDM trunk connection with IP connectivity. The PBXs use proprietary signalling method. The following is a partial configuration of the HQ router that connect to the PBX: controller t1 1/0 ds0-group 1 timeslots 1-24 type ext-sig dial-peer voice 1 voip destination-pattern 1001 session target ipv4:10.10.0.1 dial-peer voice 2 pots destination-pattern 2001 port 1/0:1 connection trunk 1001 Which command is missing from the above configuration?
A. transparent-ccs in the voice port configuration
B. signal wink-start in the controller t1 configuration
C. auto-cut-through in the pots dial peer configuration
D. codec clear-channel in the voip dial peer configuration

Correct Answer: D Section: (none) Explanation
Explanation/Reference:
Actualtests.com – The Power of Knowing 642-436
QUESTION 198
You are the network engineer at Certkiller .com. The Certkiller ISDN network has two PBX systems from
different manufactures.
Which protocol allows functionality between these two PBX systems?

A. QSIG
B. Q.921
C. Q.931
D. T-CCS

Correct Answer: A Section: (none) Explanation
QUESTION 199
You are the network technician at Certkiller .com. Your newly appointed Certkiller trainee wants to know
which application conveys fax using T.37 fax relay.
What will your reply be?

A. IVR
B. TCL
C. TIFF
D. SNMP
E. SMTP

Correct Answer: E Section: (none) Explanation
QUESTION 200
What will happen when a network link is oversubscribed?
A. The link goes down.
B. All voice calls suffer.
C. Voice packets are fragmented.
D. Excess voice calls are dropped.
E. Data packets are given priority.

Correct Answer: B Section: (none) Explanation
QUESTION 201
Certkiller sells managed IP Phone service to businesses in multi-tenant units. Certkiller has POPs in many
cities, so all of their dial peer patterns are based on 10 digit numbers. Users dial 9 for local calls, followed
by the 7 digital local number. The following dial peer has been configured in a New York POP:
dial-peer voice 595 pots

Actualtests.com – The Power of Knowing
642-436

destination-pattern 595
port 1/0:24
A user dials a local number, 9-638-4422.
What command must be configured in the gateway to allow the call to complete?

A. prefix 595
B. forward-digits 7
C. rule 1 9…….595…….

D. forward 9…….595…….

E. num-exp 9…….595…….
Correct Answer: E Section: (none) Explanation
QUESTION 202
IP Telephony uses which protocol that does not accommodate re-transmission?
A. RIP (Routing Information Protocol)
B. IP (Internet Protocol)
C. RTP (real time protocol)
D. TCP (Transmission Control Protocol)

Correct Answer: C Section: (none) Explanation
QUESTION 203
When placing a call from an IP Phone to another IP Phone, how is ringback generated??
A. CallManager generates an RTP stream to play ringback on the originated phone.
B. CallManager sends a command to the originating IP Phone to play ringback locally.
C. The originating IP Phone plays ringback locally until the RTP stream has been established.
D. The phone is connected to an audio file server that generates the inband ringback tones.

Correct Answer: B Section: (none) Explanation
QUESTION 204
Actualtests.com – The Power of Knowing
642-436
In the VoIP network above, which protocol provides the necessary sequence numbers so voice packets originating at CK1 are played in the correct order to CK5 ?
A. UDP
B. TCP
C. RTCP
D. RTP
E. CRTP

Correct Answer: D Section: (none) Explanation
QUESTION 205
What is the most probable cause of jitter?
A. Variable delay
B. Dropped packets
C. Impedance mismatch
D. Excessive delay

Correct Answer: A Section: (none) Explanation
Explanation/Reference:
Explanation: Jitter in Packet Voice Networks Jitter is defined as a variation in the delay of received packets. At the sending side, packets are sent in a continuous stream with the packets spaced evenly apart. Due to network congestion, improper queuing, or configuration errors, this steady stream can
Actualtests.com – The Power of Knowing 642-436
become lumpy, or the delay between each packet can vary instead of remaining constant. This diagram
illustrates how a steady stream of packets is handled.

When a router receives a Real-Time Protocol (RTP) audio stream for Voice over IP (VoIP), it must compensate for the jitter that is encountered. The mechanism that handles this function is the playout delay buffer. The playout delay buffer must buffer these packets and then play them out in a steady stream to the digital signal processors (DSPs) to be converted back to an analog audio stream. The playout delay buffer is also sometimes referred to as the de-jitter buffer.
QUESTION 206
When an IP phone says “Configuration CM List”, what is it doing?
A. downloading a .cnf.xml file via TFTP
B. retrieving the OS79XX.txt files from TFTP
C. downloading the application load from the TFTP server
D. attempting to register with the first two CallManagers onits list of configure CallManagers

Correct Answer: A Section: (none) Explanation
QUESTION 207
Name two sensitivities that Voice traffic has that data traffic is not necessarily affected by.
A. TPI
B. RFI
C. Delay
D. EMI
E. Jitter
F. Noise

Correct Answer: CE Section: (none) Explanation
QUESTION 208
Actualtests.com – The Power of Knowing 642-436 Your customer would like to investigate converging voice and data on their existing T1 Frame Relay WAN link between New York and Atlanta. The following applications are consuming no more bandwidth than what is in the list on this segment of the network. T1 link 1536 Kbps e-mail 75 Kbps Internet 200 Kbps Oracle 500 Kbps FTP 250 Kbps Total 1025 Kbps The customer has allocated 25% of the WAN link for routing updated and other overhead. Assuming 6 bytes overhead for Frame Relay, no cRTP and using the

A. 729 codec, how many calls could be placed on this link?
B. 2 calls
C. 3 calls
D. 4 calls
E. 5 calls
F. 6 calls

Correct Answer: C Section: (none) Explanation
Explanation/Reference:
Based upon a total bandwidth of 1536 Kbps and 1025 Kbps being used by other applications you can only have 4 calls not 5. The reason is that of the 1536 Kbps of bandwidth only 75% of it is available (or 1152 Kbps). 1152 minus 1025 leaves just 127 Kbps available for voice traffic. Assuming that you are using FRF.12, G.729 (stated in this scenario), and no cRTP (also stated in this scenario) then you will need approximately 28.14 Kbps per call with 5% overhead included (26.8 Kbps without overhead). 26.8 x 5 = 134 Kbps and 28.14 x 5 = 140.7 Kbps. Both exceed the 127 Kbps available for voice. To calculate the required bandwidth reference the “Voice Codec Bandwidth Calculator” available on Cisco’s web site (requires a CCO sign-on to access the calculator).
QUESTION 209
You have set up Call Admission Control for a customer between their headquarters and manufacturing facility over their Frame Relay WAN. You are using the
A. 726r16 codec with a 40 byte sample, CRTP without CRC, and 90 kbps configured as the maximum bandwidth for CAC to use. What will happen when 7 calls try to call the remote office?
B. All the calls will go through without any quality issues. Actualtests.com – The Power of Knowing 642-436
C. Only 4 calls will go through and the remainder will get a reorder tone.
D. Six calls will go through, and the seventh call will be placed on hold until bandwidth is available.
E. Three calls will cross the Frame Relay WAN link, and four will use the PSTN with AAR.

Correct Answer: B Section: (none) Explanation QUESTION 210
You have designed a complex dial plan using digit manipulation. Given the following snippet of your configuration file, what action would you expect to result when a call beginning with the digits “5501” is received? dial-peer voice 1 pots destination-pattern 5501… … prefix port 1/0/0
A. A nine digit number beginning with 5501 will be forwarded.
B. A ten digit number beginning with 5501 will be forwarded.
C. A nine digit number beginning with 5501612 will be forwarded.
D. A ten digit number beginning with 5501612 will be forwarded.

Correct Answer: B Section: (none) Explanation
Explanation/Reference:
Explanation: Destination Pattern 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.
QUESTION 211
What transport layer protocol does RTP utilize?
A. TCP Actualtests.com – The Power of Knowing 642-436
B. UDP
C. IP
D. ICMP

Correct Answer: B Section: (none) Explanation
Explanation/Reference:
Explanation: RTP typically runs on top of UDP to utilize its multiplexing and checksum services. Other transport protocols besides UDP can carry RTP as well. Real-Time Transport Protocol, an Internet protocol for transmitting real-time data such as audio and video. RTP itself does not guarantee real-time delivery of data, but it does provide mechanisms for the sending and receiving applications to support streaming data. Typically, RTP runs on top of the UDP protocol, although the specification is general enough to support other transport protocols.
QUESTION 212
You are the network technician at Certkiller .com. VoIP is implemented on the Certkiller network. Your newly appointed Certkiller trainee wants to know what is used to carry VoIP voice packets on this network. What will your reply be?
A. ICMP/IP
B. RTP/TCP
C. RTP/UDP
D. STP/UDP
E. RTP/RCMP

Correct Answer: C Section: (none) Explanation
QUESTION 213
Which lower layer protocol does the Real-Time Protocol (RTP) use?
A. TCP
B. UDP
C. WDP
D. HTTP
E. RTCP

Correct Answer: B Section: (none) Explanation
QUESTION 214
You are the network technician at Certkiller .com. Your newly appointed Certkiller trainee wants to know what TCP’s reliable deliver service provides.
Actualtests.com – The Power of Knowing 642-436
What will your reply be?
A. Connectionless service, flow control, sequenced delivery, and automatic error recovery
B. Flow control, sequenced delivery, automatic error recovery, and transmission window management
C. Unregulated send rate, automatic error recovery, and transmission window management
D. Connectionless service, unregulated send rate, automatic error recovery, and transmission window management

Correct Answer: B Section: (none) Explanation
QUESTION 215
You are the Voice technician at Certkiller , Inc. You want to deploy an IP telephony solution for the
company. The Certkiller network is currently a traditional LAN/WAN based on Frame Relay.
Your CEO has read about the issues of converging both data and voice traffic onto a single network. She
is concerned about the quality of their calls that need to cross the WAN in particularly.
What would you need to implement to ensure QoS for VoIP over Frame Relay?

A. Traffic shaping, priority queuing, Call Admission Control, and Class Based Weighted Fair Queuing
B. Traffic shaping, priority queuing, Call Admission Control, and Weighted Random Early Detection
C. Fragmentation, traffic shaping, priority queuing, Low Latency Queuing, and link efficiency with cRTP.
D. Fragmentation, traffic shaping, priority queuing, Call Admission Control, and Weighted Random Early Detection

Correct Answer: C Section: (none) Explanation
QUESTION 216
On what is system capacity planning based?
A. On calculations and measurements of packet length distributions.
B. On calculations and measurements of busy hour call volume/estimates.
C. On calculations and measurements of the phone costs from phone bills.
D. On calculations and measurements of the total number of calls placed during a month.

Correct Answer: B Section: (none) Explanation
Explanation/Reference:
Actualtests.com – The Power of Knowing 642-436
QUESTION 217
You have a customer that is interested in determining the number of VoIP calls their Frame Relay WAN links can support. Each of their Frame Relay WAN links has 54 kbps of bandwidth available outside all other applications and overhead. How many G.726 calls using the 32 kbps codec and 80 byte sample size can be supported?
A. 1
B. 2
C. 3
D. 4

Correct Answer: A Section: (none) Explanation
QUESTION 218
You are the network engineer at Certkiller .com. Your newly appointed Certkiller trainee wants to know
which functions use UDP as their transport mechanism.
What will your reply be? (Choose two)

A. RTP
B. RAS control function
C. call signaling function
D. H.245 control function

Correct Answer: AB Section: (none) Explanation
QUESTION 219
What does gateway require to function as a translating gateway?
A. The capacity to translate the audio.
B. The ability to recognize the call control procedures of both connecting endpoints.
C. The ability to establish separate RTP sessions with the originating and terminating endpoints.
D. The ability to recognize the call control procedures for at least one of the connecting endpoints.

Correct Answer: B Section: (none) Explanation
QUESTION 220
You are the Voice engineer at Certkiller .com. Your newly appointed Certkiller trainee wants to know what
compressed RTP does.
What will your reply be?

Actualtests.com – The Power of Knowing
642-436

A. It significantly reduce packet delay
B. It significantly reduce total bandwidth
C. It significantly reduce Frame Relay overhead
D. It significantly reduce the total number of packets

Correct Answer: B Section: (none) Explanation
QUESTION 221
You are the network engineer at Certkiller .com. You are implementing Frame Relay traffic shaping on the
Certkiller network. Your newly appointed Certkiller trainee wants to know why Frame Relay traffic shaping
is important.
What will your reply be?

A. It ensures that excess traffic above the CIR on the link is dropped.
B. It ensures that voice packets are not trapped behind large data packets.
C. It ensures that the priority of the voice packet is higher than the data packets.
D. It ensures that the RTP headed is reduced in size to reduce the overall size of the voice packet.
E. It ensures that excess traffic above the CIR on the link is not dropped, but is buffered and sent when there is capacity on the link.

Correct Answer: E Section: (none) Explanation
QUESTION 222
You are the network engineer at Certkiller .com. Certkiller has its headquarters in New York and a branch
office in Delaware. The branch office is using a 128 kbps Frame Relay link to connect to headquarters.
You want to ensure good voice quality on this link.
Which two QoS mechanisms should you implement on the Frame Relay interface? (Choose two.)

A. CIR
B. LLQ
C. WFQ
D. WRED
E. Fragmentation

Correct Answer: BE Section: (none) Explanation
QUESTION 223
You are the Voice technician at Certkiller .com. The Certkiller network uses RTCP. Your newly appointed Certkiller trainee wants to know what RTCP does.
Actualtests.com – The Power of Knowing 642-436
What will your reply be?
A. It provides independent services irrespective of RTP.
B. It provides compression techniques to save bandwidth.
C. It provides in-band control information for an RTP flow.
D. It provides out-of-band control information for an RTP flow.

Correct Answer: D Section: (none) Explanation
Explanation/Reference:
Explanation: RTCP provides out-of-band control information for an RTP flow.
QUESTION 224
Which statement is true about the MGCP call agent?
A. Acts only as a recorder of call details.
B. Provides only call signaling and call setup.
C. Manages all aspects of the call and voice stream.
D. Monitors the quality of each call after setup.

Correct Answer: B Section: (none) Explanation
Explanation/Reference:
Explanation:
In the MGCP model, the gateways focus on the audio signal translation function, while the Call Agent
handles the signaling and call processing functions.

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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
QUESTION 159
What does a gateway router match to a dialed number when setting up a VoIP call?
A. IP route
B. Destination pattern
C. Call leg
D. Session target

Correct Answer: B Section: (none) Explanation
Explanation/Reference:
Explanation:
Actualtests.com – The Power of Knowing 642-436
The router selects a dial peer for a call leg by matching the string that is defined by using the answer-address, destination-pattern, or incoming called-number command in the dial peer configuration.
QUESTION 160
What is used in the Cisco implementation of T.37?
A. Special gateways configured as IVRs
B. Special gateways configured as TIFFs
C. Special gateways configured as on-ramps and off ramps
D. Special gateways configured as MTA, MDN, and DSN parameters

Correct Answer: C Section: (none) Explanation
QUESTION 161
You are the network engineer at Certkiller .com. Your newly appointed Certkiller trainee wants to know
how an endpoint determines the address of the gatekeeper.
What will your reply be? (Choose two.)

A. The endpoint issues a GCP.
B. The endpoint issues a GRQ.
C. The endpoint queries the registrar server.
D. The endpoint is preconfigured to recognize the domain name or IP address of its gatekeeper.

Correct Answer: BD Section: (none) Explanation
QUESTION 162
You are the Voice engineer at Certkiller .com. Certkiller has an H.323 gatekeeper. Your newly appointed
Certkiller trainee wants to know what functions are supported by this gatekeeper.
What will your reply be? (Choose four.)

A. It provides services to registered endpoints.
B. It converts an alias address to an IP address.
C. It responds to bandwidth requests and modifications.
D. It provides translation between audio, video, and data formats.
E. It provides conversion between call setup signals and procedures.
F. It limits access to network resources based on call bandwidth restrictions.
G. It provides conversion between communication control signals and procedures.
Correct Answer: ABCF Section: (none) Explanation

Explanation/Reference:
Actualtests.com – The Power of Knowing 642-436
QUESTION 163
You are the network engineer at Certkiller .com. The Certkiller network is shown in the following exhibit:

If the show gatekeeper calls command shows a total of five active calls on the gatekeeper, how many call legs would the show call active voice command display on Gateway A?
A. 2
B. 5
C. 6
D. 10
E. 15

Correct Answer: D Section: (none) Explanation
QUESTION 164
You are the Voice technician at Certkiller .com. Your newly appointed Certkiller trainee wants to know what
makes it possible for gatekeepers to communicate with each other.
What will your reply be?

A. RTP
B. RAS channel
C. call signaling channel
D. H.245 control channel
E. Q.931 control channel

Correct Answer: B Section: (none) Explanation
QUESTION 165
Your newly appointed Certkiller trainee wants to know what protocol negotiates the codec type for H.323
sessions.
What will your reply be?

Actualtests.com – The Power of Knowing
642-436

A. H.225
B. H.245
C. Q.931
D. Q.932
E. H.320

Correct Answer: B Section: (none) Explanation
QUESTION 166
You are the network engineer at Certkiller .com. Certkiller has its offices in London.
You are installing a voice gateway.
What do you need to verify? (Choose two.)

A. The PSTN standards in England.
B. Encryption capabilities legalities.
C. The service provider installing the gateway.
D. Supplementary service including fax and modem.

Correct Answer: AB Section: (none) Explanation
QUESTION 167
You are the network engineer at Certkiller .com. Your newly appointed Certkiller trainee wants to know
what a voice gateway is.
What will your reply be?

A. It is a device that connects two dissimilar networks.
B. It is a device that transports voice and restricts data.
C. It is a device that can support only a distributed call processing model.
D. It is a device that cannot be connected to the traditional PSTN network.

Correct Answer: B Section: (none) Explanation
QUESTION 168
What would Receiving an Alarm Indication Signal of Blue indicate on your T1 connection where your voice traffic is going over?
A. Blue means there is an alarm occurring in the building, it is part of your disaster plan.
B. Blue means there is an alarm occurring on the line downstream from the equipment that is connected to the port
C. There is no blue alarm, only red and yellow.
D. Blue means there is an alarm occurring on the line upstream from the equipment that is connected to the port Actualtests.com – The Power of Knowing 642-436

Correct Answer: D Section: (none) Explanation
QUESTION 169
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
QUESTION 170

One voice packet is lost between Phone A and Phone B. What will be the result to the listener?
A. The call is terminated.
B. The listener will experience a gap in the received audio stream.
C. The listener will hear the audio normally. Packet loss concealment will make the loss inaudible.
D. The listener will hear the audio out of order when the lost packet is retransmitted.

Correct Answer: C Section: (none) Explanation
Explanation/Reference:
Explanation: The listener will hear the audio normally. Packet loss concealment will make the loss inaudible. Cisco Systems’ VoIP implementation enables the voice router to respond to periodic packet loss. If a voice packet is not received when expected (the expected time is variable), it is assumed to be lost and the last packet received is replayed, as shown in the figure below. Because the packet lost is only 20 ms of speech, the average listener does not notice the difference in voice quality.
Actualtests.com – The Power of Knowing 642-436

Figure. Packet Loss with G.729
Using Cisco’s G.729 implementation for VoIP, let’s say that each of the lines in the figure represents a packet. Packets 1, 2, and 3 reach the destination, but packet 4 is lost somewhere in transmission. The receiving station waits for a period of time (per its jitter buffer) and then runs a concealment strategy. This concealment strategy replays the last packet received (in this case, packet 3), so the listener does not hear gaps of silence. Because the lost speech is only 20 ms, the listener most likely does not hear the difference. You can accomplish this concealment strategy only if one packet is lost. If multiple consecutive packets are lost, the concealment strategy is run only once until another packet is received. Because of the concealment strategy of G.729, as a rule of thumb G.729 is tolerant to about five percent packet loss averaged across an entire call.
QUESTION 171
What will happen when a network link is oversubscribed?
A. The link goes down.
B. All voice calls suffer.
C. Voice packets are fragmented.
D. Excess voice calls are dropped.
E. Data packets are given priority.

Correct Answer: B Section: (none) Explanation
QUESTION 172
Your newly appointed Certkiller trainee wants to know what CAC applies to. What will your reply be?
A. Latency
B. Data traffic
C. Voice traffic
D. TCP networks
E. Voice and data traffic

Correct Answer: C Section: (none) Explanation
QUESTION 173
You are the network engineer at Certkiller .com. Certkiller has its headquarters in New York and a branch office in New Hamshire. You want to configure a permanent connection between the PBX at headquarters and the PBX at the branch
Actualtests.com – The Power of Knowing 642-436
office. The following configuration is used at the New York site: dial-peer voice 20 pots destination-pattern 20 port 1.0:1 dial-peer voice 41 voip destination-pattern 41 session target ipv4:10.2.0.20 The following configuration is used at the New Hamshire site: dial-peervoice 40 pots destination-pattern 41 port 1/0:1 dial-peer voice 20 voip destination-pattern 20
session target ipv4:10.4.1.41
What must be added to the voice port configuration at the New York site?

A. connection trunk 20
B. connection trunk 41
C. connection tie-line 20
D. connection tie-line 41

Correct Answer: B Section: (none) Explanation
Explanation/Reference:
Explanation: You must specify the same number in the connection trunk voice port command as in the appropriate dial peer destination-pattern command in order to create a permanent trunk.
QUESTION 174
You are the Voice technician at Certkiller .com. Your newly appointed Certkiller trainee wants to know what
configuration would define a destination pattern for all of the 1000 and 2000 range of extensions starting
with the numbers 555.
What will your reply be?

A. 5551…
B. 5552…
C. 555[1-2]…
D. 555[100-200]…
E. 555[1000-2000]…

Correct Answer: C Section: (none) Explanation
QUESTION 175
Certkiller distributes computer components and has warehouses in New York and
Actualtests.com – The Power of Knowing 642-436
Chicago. Headquarters is located in Washington, DC. To keep costs low, all inside sales associates are located at headquarters. Your want to provide a direct analog telephone connection to the inside sales teams from the pick-up counters at the warehouses. This connection should not require the inside sales teams to dial any digits. One of the warehouses is having a problem with their sales phone. You receive the following output: altwhse#show voice port 1/0:1 Foreign Exchange Office Type of VoicePort is E&M Operation State is DORMANT Administrative State is UP The Last Interface Down Failure Cause is Administrative Shutdown Description is not set Noise Regeneration is enabled Non Linear Processing is enabled Music On Hold Threshold is Set to -38 dBm In Gain is Set to 0 dB Out Attenuation is Set to 0 dB Echo Cancellation is enabled Echo Cancel Coverage is set to 8 ms Connection Mode is plar Connection Number is 2000 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 Region Tone is set for US What is the cause of the problem?
A. VoicePort type is incorrect.
B. Echo cancellation is enabled.
C. Connection Number is not required.
D. Interdigit Time Out is set to 10 seconds.

Correct Answer: A Section: (none) Explanation
QUESTION 176
You are the network engineer at Certkiller .com. Your newly appointed Certkiller trainee wants to know
which features render VAD ineffective.
What will your reply be? (Choose two.)

A. Fax
B. CNG
C. Call waiting Actualtests.com – The Power of Knowing 642-436
D. Music on hold
E. Call forwarding

Correct Answer: AD Section: (none) Explanation
QUESTION 177
What is a logical grouping of directory numbers (DN) and route patterns with similar reachability characteristics when working with IP Telephony?
A. Call Manager
B. CiscoWorks IP set
C. A DSN
D. A Partition

Correct Answer: D Section: (none) Explanation
QUESTION 178
What unlocks the 7960 configuration menu?
A. **3
B. **#
C. **4
D. **#*

Correct Answer: B Section: (none) Explanation QUESTION 179
Name two standards that are being adopted from by the telecommunicates industry that are used to communicate between applications such as the Cisco CallManager providing IP PBX functionality and unified products such as the GateServer products acquired through the acquisition of Amteva. (Select two.)
A. The Java Telephone Application Programmable Interface (JTAPI)
B. The IP Telephone Call protocol (IPTC)
C. The Telephony Application programmable Interface (TAPI)
D. The System Architecture Voice Telephony Architecture (SAVTA)

Correct Answer: AC Section: (none) Explanation
QUESTION 180
Which network protocols does an IP Phone use to communicate?
Actualtests.com – The Power of Knowing 642-436
A. TCP/IP for both skinny signalling and RTP voice streams
B. UDP/IP for both skinny signalling and RTP voice streams
C. TCP/IP for skinny signalling and UDP/IP for RTP voice streams.
D. TCP/IP for skinny signalling and TCP/IP for RTP voice streams.

Correct Answer: C Section: (none) Explanation
QUESTION 181
There are six major steps for WAN deployment when preparing IP telephony. From the list below, please
select which of the following are valid pre deployment choices.
(Choose all that apply.)

A. Choosing Wiring Closets carefully
B. Determining Voice Bandwidht Requirements
C. Assessing Results
D. Selecting the right handset for the IP SoftPhone
E. Analyzing Upgrade Requirements
F. Collecting Information on the Current WAN Environment

Correct Answer: BCEF Section: (none) Explanation
QUESTION 182
Before voice and video can be placed on a network, it is necessary to ensure that adequate bandwidth exists for all required applications. To begin, the minimum bandwidth requirements for each major application (for example, the voice media streams, video streams, voice control protocols, and all data traffic) should be summed. This sum represents the minimum bandwidth requirement for any given link, and it should consume no more than what percentage of the total bandwidth available on that link?
A. 25%
B. 50%
C. 100%
D. 75%

Correct Answer: D Section: (none) Explanation
QUESTION 183
You need to prefix any outbound number dialled by a user with a 9. Where can you do this? (Choose two.)
A. in a Route Filter
B. on a Route Pattern Actualtests.com – The Power of Knowing 642-436
C. on a Translation Pattern
D. on the phone configuration mask

Correct Answer: BC Section: (none) Explanation
QUESTION 184
Your Manager asks you as the Lead Network Designer to give a status on the VOIP integration project. Your Manager specifically asks what you need to replace the PBX. From the list below, what are you going to need to replace the PBX to roll out the VOIP solution? (Choose all that apply.)
A. TCP/IP
B. Cisco CallManager
C. IPX/SPX compatible
D. IP Telephones
E. All of the answers
F. Cat 4000’s

Correct Answer: ABDF Section: (none) Explanation
QUESTION 185
What is the major advantage of designing and placing VoIP and Internet telephony in a clients organization?
A. It is cheap but you still need a PBX regardless
B. The PSTN is doomed to be EOL in 5 years and this is the replacement.
C. It avoids the tolls charged by ordinary telephone service
D. Even without QoS it is much clearer that PSTN technology.

Correct Answer: C Section: (none) Explanation
QUESTION 186
You are the network technician at Certkiller .com. Your newly appointed Certkiller trainee wants to know
what function T-CCS performs.
What will your reply be?

A. It allows a PBX to pass signalling to the PSTN switch.
B. It allows a PBX to pass analog signalling to the router
C. It allows a PBX to pass signalling to the router for compression and processing
D. It allows a PBX to pass proprietary signalling to another PBX across the IP network.

Correct Answer: D Section: (none) Explanation
Explanation/Reference:
Actualtests.com – The Power of Knowing 642-436
QUESTION 187
You are the network engineer at Certkiller .com. You to connect a Cisco voice gateway to a PBX or the
PSTN via ISDN (PRI, QSIG, BRI).
What are two attributes of the PBX/PSTN switch that must be known to understand which features to
configure on the voice gateway to connect successfully to it? (Choose two)

A. Whether Q.921 or Q.931 is supported by the PBX/PSTN switch.
B. Whether Symmetric mode is supported by the PBX/PSTN switch.
C. Which PRI/BRI switch-type is supported by the PBX/PSTN switch.
D. Whether network or user side is supported by the PBX/PSTN switch.
E. Whether wink, delay dial, or immediate dial is supported by the PBX/PSTN switch.

Correct Answer: CD Section: (none) Explanation
QUESTION 188
You are working with a potential customer that would like to integrate its existing

Actualtests.com – The Power of Knowing
642-436

PBX telephone system into its IP network. The accompanying figure shows that the customer has two
offices that need to be connected to the IP network so that the customer can exchange telephone calls
without using the PSTN. Both PBXs are currently connected to T1 ISDN circuits.
Which signaling type will allow you to support your customer?

A. QSIG
B. CCS
C. CAS
D. T-CCS
E. E&M
F. FXO

Correct Answer: A Section: (none) Explanation
QUESTION 189
Which statement is an example of in-band signaling?
A. Uses a single channel for synchronization and hook status.
B. Transports synchronization signals within the voice channel.
C. Carries hook status in a dedicated signaling channel.
D. Robs bits from some frames to provide signaling states.

Correct Answer: D Section: (none) Explanation
QUESTION 190
You are the network technician at Certkiller .com. VoIP is implemented on the Certkiller network. Your
newly appointed Certkiller trainee wants to know what this implementation uses to carry the payload
across the network.
What will your reply be?

A. Only RTP
B. Only UDP
C. UDP inside RTP
D. RTP inside UDP

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

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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?

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: B Section: (none) Explanation

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Exam A
QUESTION 1
When setting up a VoIP call, what is the first thing a gateway router tries to match to a dialed number?
A. session target
B. call leg
C. destination pattern
D. IP route
Correct Answer: C Section: (none) Explanation
QUESTION 2
In North America, which E&M signaling type is used most often for geographically separated equipment?
A. Type III
B. Type II
C. Type V
D. Type I
E. Type IV
Correct Answer: B Section: (none) Explanation
QUESTION 3
When using CUBE, which two statements describe how media flow-through differs from media flow-around? (Choose two.)
A. Media flow-through terminates the signaling channel and the RTP streams flow directly between endpoints.
B. Media flow-through terminates the RTP streams but allows signaling to flow directly between endpoints.
C. Media flow-around and media flow-through function in a similar manner, but media flow-around supports NAT traversal.
D. Media flow-around provides address hiding by terminating both signaling and RTP streams.
E. Media flow-through provides address hiding by terminating both signaling and RTP streams.
F. Media flow-around terminates the signaling stream and allows RTP streams to flow directly between endpoints.
Correct Answer: EF Section: (none) Explanation
Explanation/Reference:
Actualtests.com – The Power of Knowing 642-436
QUESTION 4
Which two are types of Call Admission Control? (Choose two.)
A. gateway zone bandwidth
B. topology-based
C. resource-based
D. gatekeeper-controlled RSVP
E. local
F. QoS-based
Correct Answer: CE Section: (none) Explanation
QUESTION 5
Site A uses three-digit internal numbers and remote Site B uses four-digit internal numbers. All calls to the PSTN are routed through Site B. What dial plan below best represents provision simplicity, assuming the NANP numbering plan?
A. Translate all called numbers within Site A to four digits.
B. Translate all called numbers within Site B to three digits.
C. Translate all called numbers at either site to ten digits.
D. Translate all called numbers leaving Site A to ten digits.
Correct Answer: D Section: (none) Explanation
QUESTION 6
Which three are supervisory signals? (Choose three.)
A. call waiting
B. on hook
C. ring
D. busy
E. off hook
Correct Answer: BCE Section: (none) Explanation
QUESTION 7
Which command parameter specifies that the router should not attempt to initiate a trunk connection but should wait for an incoming call before establishing the trunk?
A. connection trunk answer-mode 408555
B. connection trunk 408555….
C. connection-trunk 404555…. answer-mode Actualtests.com – The Power of Knowing 642-436
D. ds0-group timeslots 1-23 type ext-sig
E. voice-port 1/0:1
Correct Answer: C Section: (none) Explanation
QUESTION 8
What does a gatekeeper do when it matches a technology prefix?
A. strips off the zone prefix and forwards the technology prefix to the remote gatekeeper
B. strips off the technology prefix and sends the matching zone prefix to the remote gatekeeper
C. sends both the technology prefix and zone prefix to the remote gatekeeper
D. strips off both the technology prefix and zone prefix and forwards the remaining destination number

Correct Answer: C Section: (none) Explanation
QUESTION 9
At what point does the MGCP call agent release the setup of the call path to the residential gateways?
A. 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
B. after the call agent has been notified of an event and has instructed the source residential gateway to create a connection
C. does not release call path setup
D. after the call agent has been notified that an event occurred at the source residential gateway
E. after the call agent has sent a connection request 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
Correct Answer: E Section: (none) Explanation
QUESTION 10
What is the best description of an MGCP endpoint?
A. the gatekeepers in a VoIP network
B. IP phones
C. any analog telephony device (PBX, switch, etc.)
D. the interconnection between packet and traditional telephone networks Actualtests.com – The Power of Knowing 642-436
Correct Answer: D Section: (none) Explanation
QUESTION 11
Using Cisco Unified Communications Manager Express, what four steps are necessary to implement COR? (Choose four.)
A. Define COR labels.
B. Configure COR lists on voice ports.
C. Configure dial peers and assign COR lists.
D. Assign COR list to ephone-DN.
E. Configure SRST.
F. Configure COR lists.
Correct Answer: ACDF Section: (none) Explanation QUESTION 12
A telemarketing firm needs to use number translation for incoming and outgoing calls. They have defined
two translation profiles, one for incoming and one for outgoing calls.
What can be used to simplify this task?

A. dial peer
B. source IP group
C. trunk group
D. voice port
E. hunt group
Correct Answer: C Section: (none) Explanation
QUESTION 13
The SJ local zone contains a gatekeeper that controls two gateways, SJ1 and SJ2. Both gateways provide access to area code 408. Which two command strings should be entered into the gatekeeper to give the SJ2 gateway priority over the SJ1 gateway? (Choose two.)
A. zone prefix SJ 408 gw-priority 0 SJ2, 10 SJ1
B. zone prefix SJ 408 gw-priority 10 SJ1
C. zone prefix SJ 408 gw-priority 6 SJ1
D. zone prefix SJ 408 gw-priority 6 SJ2
E. zone prefix SJ 408 gw-priority 6 SJ1, 10 SJ2
F. zone prefix SJ 408 gw-priority 10 SJ2
Correct Answer: CF Section: (none) Explanation
Explanation/Reference:
Actualtests.com – The Power of Knowing 642-436
QUESTION 14
What is the most common E&M type used outside North America?
A. Type III
B. Type I
C. Type II
D. Type V
E. Type IV
Correct Answer: D Section: (none) Explanation
QUESTION 15
Which statement is true about MGCP?
A. Call completion is always shared, with some intelligence on the endpoint, some on the call agent.
B. Endpoints may act alone or cooperate with call agent to complete calls.
C. Endpoints always take all actions to complete calls.
D. Call agents order and direct each step of call completion for the endpoints.
Correct Answer: D Section: (none) Explanation
QUESTION 16
Which dial-peer command can set the parameters that search through a series of dial peers for a destination that is not in use?
A. distribute
B. request
C. circulate
D. rotary
E. hunt
F. query
Correct Answer: E Section: (none) Explanation
QUESTION 17
What is the E.164 standard?
A. national numbering plan
B. dial plan
C. private numbering plan Actualtests.com – The Power of Knowing 642-436
D. international public telecommunications numbering plan
Correct Answer: D Section: (none) Explanation
QUESTION 18
Which option is true concerning the MGCP call agent?
A. manages all aspects of the call and voice stream
B. acts only as a recorder of call details
C. provides only call signaling and call setup
D. monitors the quality of each call after setup
Correct Answer: C Section: (none) Explanation
QUESTION 19
Which best defines an ACD?
A. a telephone system that switches calls between users on local lines
B. a telephone system that responds to a caller with a voice menu and helps to appropriately connect the call
C. a telephone system that is connected to the exchange to provide conventional voice services to several subscribers
D. a local company that provides phone capability and distribution from the phone company’s central office
Correct Answer: B Section: (none) Explanation
QUESTION 20
Which two statements describe the purpose of the technology prefix? (Choose two.)
A. Technology prefixes are prepended to the destination address by the gateway.
B. Technology prefixes must always be configured on gateways.
C. Technology prefixes are configured on gateways to indicate to the gatekeeper whether they support voice or video.
D. Technology prefixes have to be unique on each gateway.
E. Technology prefixes are used to identify different types or classes of gateways.
Correct Answer: AE Section: (none) Explanation
QUESTION 21
You have set up a complex dial plan using translation rules. The following translation rule has been configured. What output would correspond to the test translation-rule
Actualtests.com – The Power of Knowing 642-436
command? translation-rule 1 rule 0 ^0.. 215550210 rule 1 ^1.. 215550211 rule 2 ^2.. 215550212 rule 3 ^3.. 215550213 rule 4 ^4.. 215550214 rule 5 ^5.. 215550215 rule 6 ^6.. 215550216 rule 7 ^7.. 215550217 rule 8 ^8.. 215550218 rule 9 ^9.. 215550210
A. test translation-rule 1 555 The replaced number: 55521555021
B. test translation-rule 1 617 The replaced number: 61721555021
C. test translation-rule 1 910 The replaced number: 21555021910
D. test translation-rule 1 512 The replaced number: 21555021512
Correct Answer: D Section: (none) Explanation
QUESTION 22
A customer wants to roll out IP telephony to the regional office. They are currently using the G.711 codec at headquarters. Which codec will support voice activity detection and comfort noise generation?
A. G.723.1
B. G.726
C. G.711
D. G.729B
Correct Answer: D Section: (none) Explanation
QUESTION 23
Which three services are supported by CUBE when supporting H323-to-SIP calls? (Choose three.)
A. codec transparent support
B. media flow-through
C. Transport Layer Security
D. H.261, H.263, and H.264 video codecs Actualtests.com – The Power of Knowing 642-436
E. SIP cause codes
F. media flow-around
Correct Answer: ABC Section: (none) Explanation
QUESTION 24
Which statement is true about only out-of-band signaling?
A. Signaling bits are sent in a special order in a dedicated signaling frame.
B. All voice packets carry their own signaling.
C. All signaling is directly associated with its corresponding voice frame.
D. A signaling bit is robbed from each frame.
Correct Answer: A Section: (none) Explanation
QUESTION 25
A customer needs to configure a CAS E & M circuit that will support inbound and outbound DNIS and inbound ANI. Which configuration will accomplish this task?
A. ds0-group 0 timeslots 1-24 type fgd-eana
B. ds0-group 0 timeslots 1-24 type e&m-fgd
C. pri-group timeslots 1-24
D. ds0-group 0 timeslots 1-24 type none
E. ds0-group 0 timeslots 1-31 type r2-digital r2-compelled ani
Correct Answer: B Section: (none) Explanation
QUESTION 26
The D channel in ISDN is an example of which two signaling methods? (Choose two.)
A. in-band
B. gateway
C. CAS
D. CCS
E. out-of-band

Correct Answer: DE Section: (none) Explanation
QUESTION 27
Examine the example output. hostname GW1 !
Actualtests.com – The Power of Knowing 642-436
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 1212. 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 GW1-zone2.abc.com abc.com ipaddr 172.16.2.1
B. h323-gateway voip bind srcaddr 172.16.2.2
C. h323-gateway voip h323-id GK1
D. gateway
Correct Answer: D Section: (none) Explanation
QUESTION 28
Which statement best describes gatekeeper operation when the technology prefix is matched and the gatekeeper is using the technology prefix with hopoff?
A. The gatekeeper attempts to forward the call to the hopoff zone, but if this fails, it will forward the call to the zone specified in the zone prefix command.
B. The gatekeeper only forwards the call to the hopoff zone if the zone prefix does not match.
C. The gatekeeper attempts to forward the call to the zone specified in the zone prefix command first, but if this fails, it will forward the call to the zone specified in the hopoff command.
D. The gatekeeper always forwards the call to the zone specified in the hopoff command.
Correct Answer: D Section: (none) Explanation QUESTION 29
A new business in Great Britain needs to have a PSTN connection that will handle a maximum of 30 inbound and outbound calls at any given time. The customer only has
Actualtests.com – The Power of Knowing 642-436
one slot available on the designated PSTN router. Which digital line type should be recommended?
A. QSIG
B. ISDN E1 PRI
C. ISDN BRI
D. ISDN T1 PRI
Correct Answer: B Section: (none) Explanation
QUESTION 30
Where would you assign COR lists in Cisco Unified Communications Manager Express?
A. ephone
B. voice register dn
C. ephone-dn
D. voice register pool

Correct Answer: C Section: (none) Explanation
QUESTION 31
Which two codes together make up the number that follows the E.164 recommendation numbering scheme? (Choose two.)
A. subscriber code
B. national destination code
C. country code
D. provider code

Correct Answer: AC Section: (none) Explanation
QUESTION 32
Which process changes an internal extension into a fully qualified external PSTN number before matching to a dial peer?
A. number expansion
B. digit masking
C. prefix extension
D. forward digits
Correct Answer: A Section: (none) Explanation

Explanation/Reference:
Actualtests.com – The Power of Knowing 642-436
QUESTION 33
Using a standalone IOS gateway, which three steps are necessary to implement COR? (Choose three.)
A. Configure COR lists.
B. Configure dial peers and assign COR lists.
C. Define COR labels.
D. Assign COR list to ephone-DN.
E. Configure COR lists on voice ports.
F. Configure SRST.

Correct Answer: ABC Section: (none) Explanation
QUESTION 34
Which dial plan characteristic shows the most obvious improvement by dropping a number translation step?
A. post-dial delay
B. availability
C. scalability
D. hierarchical design

Correct Answer: A Section: (none) Explanation
QUESTION 35
Exhibit:

You work as a network administrator for Certkiller .com. You study the exhibit carefully. All IP phones use SCCP. Fax machine F calls fax machine J. Which call setup signaling
Actualtests.com – The Power of Knowing 642-436
statement is correct?
A. Gateway A signals the call agent. The call agent processes the call and signals gateway B.
B. Fax F signals Fax J directly. Call setup is handled by the fax machines.
C. Gateway A processes the call and signals gateway B. Gateway B processes the call. During the setup, the gateways query the gatekeeper for address resolution and call setup permission.
D. Gateway A processes the call and signals gateway B. Gateway B processes the request.
E. Gateway A signals the gatekeeper. The gatekeeper processes the call and signals gateway B.

Correct Answer: C Section: (none) Explanation
QUESTION 36
Exhibit:
Actualtests.com – The Power of Knowing 642-436

You work as a network administrator for Certkiller .com. You study the exhibit carefully. You have a client
that is testing a directory gatekeeper in the lab to provide address resolution between two different zones.
Two of the commands in the running-config output are incorrect.
Which two changes will correct the configuration? (Choose two.)

A. replace zone prefix GK-B 404 . . . . with zone prefix GK-B 404 . . . . . . .
B. replace zone prefix GK-A 770 . . . . . . . with zone prefix GK-A 770 . . . .
C. replace zone remote GK-B acme.com 172.16.14.99 1719 with Actualtests.com – The Power of Knowing 642-436 zone local GK-B acme.com 172.16.14.99 1719
D. replace zone local DGK acme.com with zone remote DGK acme.com
E. replace zone local GK-A acme.com 172.16.14.44 1719 with zone remote GK-A acme.com 172.16.14.44 1719

Correct Answer: BE Section: (none) Explanation
QUESTION 37
Exhibit:

You work as a network administrator for Certkiller .com. You study the exhibit carefully. Certkiller .com uses H.323 to place calls to their supplier RR Industries. Certkiller .com also has a voice connection to an ITSP for long distance over a SIP network. Which configuration should Certkiller use to deploy the CUBE?
A. voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323
B. service voice voip allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to sip
allow-connections sip to h323

C. service voice voip allow-connections h323 to h323 allow-connections h323 to sip
D. voice service voip allow-connections h323 to h323 Actualtests.com – The Power of Knowing 642-436 allow-connections h323 to sip

Correct Answer: A Section: (none) Explanation
QUESTION 38
Exhibit:

You work as a network administrator for Certkiller .com. You study the exhibit carefully. Which protocol provides the necessary sequence numbers so that voice packets originating at R1 are played in the correct order at R5?
A. UDP
B. RTP
C. RTCP
D. LFI
E. CRTP

Correct Answer: B Section: (none) Explanation QUESTION 39
Exhibit:
Actualtests.com – The Power of Knowing 642-436

You work as a network administrator for Certkiller .com. You study the exhibit carefully. What is the minimum WAN bandwidth required to support three simultaneous VoIP calls in this network?
A. 51,600 bps
B. 19,200 bps
C. 247,200 bps
D. 79,200 bps

Correct Answer: D Section: (none) Explanation
QUESTION 40
Exhibit:

You work as a network administrator for Certkiller .com. You study the exhibit carefully. Which configuration option will allow communication between a voice-enabled router and a PBX?
A. voice port 1/0/0 signaling wink-start operation 4-wire auto-cut-through type 1
B. voice port 1/0/0 signaling immediate-start operation 4-wire type 5
C. voice port 1/0/0 signaling delay-start auto-cut-through operation 4-wire type 3 Actualtests.com – The Power of Knowing
642-436
D. voice port 1/0/0 signaling wink-start operation 4-wire type 4

Correct Answer: A Section: (none) Explanation
QUESTION 41
Exhibit:

You work as a network administrator for Certkiller .com. You study the exhibit carefully. You have been asked to configure a dial peer on Certkiller 2 that will match only the extensions of the four telephones attached. Which dial-peer statement will you use?
A. dial-peer voice 1 pots destination-pattern 5552.[0-5]0
B. dial-peer voice 1 pots destination pattern 5552[5-6].0
C. dial-peer voice 1 pots destination-pattern 5552[5-6][05]0
D. dial-peer voice 1 pots destination-pattern 555[2-5][56]

Correct Answer: C Section: (none) Explanation
QUESTION 42
Exhibit:
Actualtests.com – The Power of Knowing
642-436
You work as a network administrator for Certkiller .com. You study the exhibit carefully. Users are not able to complete a call from 678-555-1212 to 770-555-1111. What is the correct diagnosis for the problem?
A. incorrect destination-pattern in router 1
B. incorrect port statement in router 1 pots dial peer
C. missing no digit-strip on the voip dial peer in router 1
D. incorrect POTS dial-peer statement in router 2
E. incorrect session-target statement in router 2

Correct Answer: A Section: (none) Explanation
QUESTION 43
Exhibit:
Actualtests.com – The Power of Knowing 642-436 You work as a network administrator for Certkiller .com. You study the exhibit carefully. You have configured a gatekeeper and an IP-IP gateway on the same router. When you look at the output from the show gatekeeper endpoint command, the IP-IP gateway is not registered with the gatekeeper. What needs to be configured to resolve this issue?

A. You need to add a VoIP dial peer to the configuration.
B. You need to stop and restart the gateway.
C. The h323-gateway voip id command has an incorrect IP address.
D. The h323-gateway voip id command has an incorrect gatekeeper ID and IP address.

Correct Answer: A Section: (none) Explanation
QUESTION 44
Exhibit:
Actualtests.com – The Power of Knowing 642-436 Refer to the output from the debug h225 asn1 command in the exhibit. You have configured a gatekeeper with two local zones, hq and br. You want the gateway at the branch location to register with zone BR. What needs to be corrected in the branch gateway to resolve the issue?

A. Change the gatekeeper-id in the h323-gateway voip id command.
B. Change the IP address in the h323-gateway voip id command.
C. Add a zone remote for zone BR so the gateway can register with the correct zone.
D. Change the gatekeeper-id and the IP address in the h323-gateway voip id command. Actualtests.com – The Power of Knowing 642-436

Correct Answer: A Section: (none) Explanation
QUESTION 45
Exhibit:

To hide its identity when initiating calls, Phone B requests that Server Certkiller B place its calls for it. What kind of device is Server Certkiller B?
A. user agent client
B. redirect
C. registrar
D. proxy
E. user agent server

Correct Answer: D Section: (none) Explanation
QUESTION 46
Exhibit:

Refer to the IOS configuration in the exhibit. How will the next incoming call be routed?
Actualtests.com – The Power of Knowing 642-436
A. The call will be routed to a random available channel.
B. The call will be routed to the longest idle channel.
C. The call will be routed to the next available channel, starting from channel 1, hunting up toward channel
24.
D. The call will be routed to the least used channel.
E. The call will be routed to the next available channel, starting from channel 24, hunting down toward channel 1.

Correct Answer: E Section: (none) Explanation
QUESTION 47
Exhibit:

You work as a network administrator for Certkiller .com. You study the exhibit carefully. Enzo’s Bikes manufactures high end bicycle frames. Until recently they sold only to bicycle shops; however, now they are starting to sell to end users. They need a way to add two additional sales staff and ensure that the senior sales technician always gets the first call. Drew is the senior sales technician. Bob is the newest sales technician. Bob’s phone should always be the last one chosen for incoming sales calls, after Drew and James. Bob’s phone should be chosen first only when Drew and James are busy on calls. Select the correct dial-peer command set for Bob’s phone.
A. dial-peer voice 3 pots destination-pattern 5555110 preference 2
B. dial-peer voice 3 pots destination-pattern 5555110 preference firstlast
C. dial-peer voice 3 pots destination-pattern 5555110 preference 3 huntstop
D. dial-peer voice 3 pots destination-pattern 5555110 Actualtests.com – The Power of Knowing
642-436
preference high

E. dial-peer voice 3 pots destination-pattern 5555110 preference 0

Correct Answer: A Section: (none) Explanation
QUESTION 48
Exhibit:

You work as a network administrator for Certkiller .com. You study the exhibit carefully. Three department managers share the directory number 3000. The Marketing manager’s phone is attached to port 1/1. The Engineering manager’s phone is attached to port 1/2. The Shipping manager’s phone is attached to port 1/3. In which situation would an incoming call ring on the Shipping manager’s phone?
A. None of the managers are on the phone.
B. The Engineering manager is on the phone.
C. The Marketing manager is on the phone.
D. The Shipping manager and Marketing manager are on the phone.
E. The Engineering manager and Marketing manager are on the phone.

Correct Answer: E Section: (none) Explanation
QUESTION 49
Exhibit:
Actualtests.com – The Power of Knowing
642-436
You work as a network administrator for Certkiller .com. You study the exhibit carefully. All IP phones use SCCP. Phone D calls phone G. Which statement is true about the audio path?
A. The first call leg terminates at gateway
B. The second call leg is from gateway A to phone G.
C. The first call leg terminates at gateway B. The second call leg is from gateway B to phone G.
D. The voice packets are routed through the gatekeeper.
E. The first call leg terminates at gateway
F. The second call leg is from gateway A to its termination at gateway B. The third call leg is from gateway B to phone G.
G. The voice packets are routed through the call agent.
H. The voice packets travel directly from phone to phone.

Correct Answer: F Section: (none) Explanation
QUESTION 50

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Exam A
QUESTION 1
Which two codes together make up the number that follows the E.164 recommendation numbering scheme? (Choose two.)
A. country code
B. subscriber code
C. national destination code
D. provider code
Correct Answer: AB Section: Voice Fundamentals Explanation
Explanation/Reference:
E.164 is an international numbering plan created by the International Telecommunication Union (ITU).
Each number in the E.164 numbering plan
contains the following components:
Country code (CC)
National destination code (NDC – optional)
Subscriber number (SN)

The CC consists of one, two or three digits. It is what we add in order to access different countries and often prefixed with a +
The NDC is the code we often call the area code.
The SN is for telephone numbering. It is given by your phone operator.
E.164 numbers are limited to a maximum length of 15 digits.
For example, the North American Numbering Plan E.164 is as follows:
1-602-555-1212
+
1: Country code

+
602555: National destination code (for North American Numbering Plan, 602 is called the area code while 555 is called Central Office Code)

+
1212: Subscribe Number
Answer C is also correct but just optional. E.164 Numbering Plan must have Country Code and Subscriber Code so A & B are the correct answers.
QUESTION 2
Which statement is true about only out-of-band signaling?
A. A signaling bit is robbed from each frame.
B. Signaling bits are sent in a special order in a dedicated signaling frame.
C. All signaling is directly associated with its corresponding voice frame.
D. All voice packets carry their own signaling.
Correct Answer: B Section: Voice Fundamentals Explanation
Explanation/Reference:
Out-of-Band signaling is telecommunication signaling exchange of information in order to control a telephone call. Out-of-Band signaling uses common channel signaling (CCS), that means signaling information is transmitted using a separate, dedicated signaling channel.
Answers A C D are characteristics of Channel associated signaling (CAS) so they are not correct.
QUESTION 3
The D channel in ISDN is an example of which two signaling methods? (Choose two.)
A. CCS signaling
B. out-of-band signaling
C. in-band signaling
D. CAS signaling
Correct Answer: AB Section: Voice Fundamentals Explanation
Explanation/Reference:
QUESTION 4
In North America, which E&M signaling type is used most often for geographically separated equipment?
A. Type I
B. Type II
C. Type III
D. Type IV
E. Type V
Correct Answer: B Section: Voice Fundamentals Explanation
Explanation/Reference:
This information is quoted from http://www.cisco.com/en/US/tech/tk1077/
technologies_tech_note09186a0080093f60.shtml

E&M Type I – This is the most common interface in North America.
Type I uses two leads for supervisor signaling: E, and M.
During inactivity, the E-lead is open and the M-lead is connected to the ground.
The PBX (that acts as trunk circuit side) connects the M-lead to the battery in order to indicate the off-hook
condition.
The Cisco router/gateway (signaling unit) connects the E-lead to the ground in order to indicate the off-
hook condition.

E&M Type II – Two signaling nodes can be connected back-to-back.
Type II uses four leads for supervision signaling: E, M, SB, and SG.
During inactivity both the E-lead and M-lead are open.
The PBX (that acts as trunk circuit side) connects the M-lead to the signal battery (SB) lead connected to
the battery of the signaling side in order to indicate the off-hook condition.
The Cisco router / gateway (signaling unit) connects the E-lead to the signal ground (SG) lead connected
to the ground of the trunk circuit side in order to indicate the off-hook condition.

E&M Type III – This is not commonly used in modern systems.
Type III uses four leads for supervision signaling: E, M, SB, and SG.
During inactivity, the E-lead is open and the M-lead is set to the ground connected to the SG lead of the
signaling side.
The PBX (that acts as trunk circuit side) disconnects the M-lead from the SG lead and connects it to the
SB lead of the signaling side in order to indicate the off-hook condition.
The Cisco router / gateway (signaling unit) connects the E-lead to the ground in order to indicate the off-
hook condition.

E&M Type IV – This is not supported by Cisco routers / gateways.

E&M Type V – Type V is symmetrical and allows two signaling nodes to be connected back-to-back. This is
the most common interface type used outside of North America.
Type V uses two leads for supervisor signaling: E, and M.
During inactivity the E-lead and M-lead are open.
The PBX ( that acts as trunk circuit side) connects the M-lead to the ground in order to indicate the off-
hook condition.
The Cisco router / gateway (signaling unit) connects the E-lead to the ground in order to indicate off-hook
condition.

Although above information specifies E&M Type 1 is the most commonly used interface in North America
but this type generates significant delay in the signaling operation when transmitting between
geographically separated equipment and affects voice signal quality (because of significant inductance and
capacitance of the long wires) so Type 2 is often used instead.

QUESTION 5
Which three are supervisory signals? (Choose three.)
A. busy
B. on hook
C. off hook
D. call waiting
E. ring
Correct Answer: BCE Section: Voice Fundamentals Explanation
Explanation/Reference:
QUESTION 6
What is the approximate frequency range of human speech?
A. 20 Hz to 20,000 Hz
B. 40 Hz to 15,000 Hz
C. 200 Hz to 9000 Hz
D. 600 Hz to 5400 Hz
Correct Answer: C Section: Voice Fundamentals Explanation
Explanation/Reference:
QUESTION 7
What is the process of assigning audio amplitude to a unique digital code word?
A. linear prediction
B. encoding
C. sampling
D. quantization
Correct Answer: D Section: Voice Fundamentals Explanation
Explanation/Reference: QUESTION 8
What is the E.164 standard?
A. private numbering plan
B. national numbering plan
C. dial plan
D. international public telecommunications numbering plan
Correct Answer: D Section: Voice Fundamentals Explanation
Explanation/Reference:
QUESTION 9
For the following items, which is the most common E&M type used outside North America?
A. Type IV
B. Type I
C. Type II
D. Type III
E. Type V
Correct Answer: E Section: Voice Fundamentals Explanation
Explanation/Reference:
QUESTION 10
A new business in Great Britain needs to have a PSTN connection that will handle a maximum of 30 inbound and outbound calls at any given time. The customer only has one slot available on the designated PSTN router. Which digital line type should be recommended?
A. QSIG
B. ISDN BRI
C. ISDN E1 PRI
D. ISDN T1 PRI
Correct Answer: C Section: Voice Fundamentals Explanation
Explanation/Reference: Exam B
QUESTION 1
Refer to the exhibit for IP addresses and telephone numbers. You are working with a customer opening a small sales office in Atlanta. You want the user in Atlanta to be able to dial into the PBX in New York over the IP WAN. The New York PBX uses ground start, a two-wire operation, and DTMF dialing. Choose the correct FXO port configuration commands for New York.
Exhibit:

A. voice-port 1/0/0signal ground-startoperation 2-wiredial-type dtmf
B. voice-port 1/1/1destination 2015551212 signal ground-startoperation 2-wiretype 1dial-type dtmf
C. voice port 1/0/0session target ipv4:172.16.1.1destination 2015551212 signal ground-startoperation 2-wiredial-type dtmf
D. voice port 1/0/0session target ipv4:172.16.1.1source 2015551212 signal wink-startoperation 2-wiredial-type dtmf
Correct Answer: A Section: Analog Voice Port Explanation
Explanation/Reference:
QUESTION 2
Refer to the exhibit. Which configuration option will allow communication between a voice-enabled router and a PBX?
Exhibit:

A. voice port 1/0/0signaling wink-startoperation 4-wireauto-cut-throughtype 1
B. voice port 1/0/0signaling immediate-startoperation 4-wiretype 5
C. voice port 1/0/0signaling delay-startauto-cut-throughoperation 4-wiretype 3
D. voice port 1/0/0signaling wink-startoperation 4-wiretype 4
Correct Answer: A Section: Analog Voice Port Explanation
Explanation/Reference:
QUESTION 3
Examine the following PBX system parameters:
The calling side seizes the line by going off-hook on its E-lead and sends information as DTMF digits.

The voice path is 4-wires, and the voice enabled router is in another building from the PBX.
Select the correct set of commands to allow communication between a voice enabled router and a PBX.
A. voice port 1/0/0signal immediate-startoperation 4-wiretype 2
B. voice-port 1/0/0signal delay-dialoperation 4-wiretype 1
C. voice port 1/0/0signal wink-startoperation 4-wiretype 3
D. voice port 1/0/0signal immediate-startoperation 4-wiretype 4
Correct Answer: A Section: Analog Voice Port Explanation
Explanation/Reference: Exam C
QUESTION 1
Which option is true concerning the MGCP call agent?
A. acts only as a recorder of call details
B. provides only call signaling and call setup
C. manages all aspects of the call and voice stream
D. monitors the quality of each call after setup
Correct Answer: B Section: Call Signaling Explanation
Explanation/Reference:
MGCP Call Agent is a central control component to remotely control various devices. When the MGCP call agent exists in the network, calls are routed via route patterns on the Call Agent (Cisco Unified Communications Manager), not by dial peers on the gateway.

The messages sent between the voice gateway and the MGCP Call Agent are just used for call signaling and call setup only. In summary, the Call Agent will instruct the gateways what to do in each stage: receive dialed digits, find the destination gateway, send connection request… Finally, the Call Agent will allow gateways to establish RTP Streams with each other. Notice that the voice streams only flow between the two voice gateways, not to the Call Agent.
At the conversation finishs (one of the endpoints goes on-hook), that gateway notifies the Call Agent and the Call Agent sends Delete Connection (DLCX) Requests for both gateways.
QUESTION 2
At what point does the MGCP call agent release the setup of the call path to the residential gateways?
A. after the call agent has been notified that an event 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. does not release call path setup
D. after the call agent has sent a connection request 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: Call Signaling Explanation

The MGCP call agent releases the setup of the call path to the residential gateways when the conversation begins. After sending the Modify Connection (MDCX), the two gateways have enough information to start the conversation so the duty of the Call Agent finishs.
QUESTION 3
Which three services are supported by CUBE when supporting H323-to-SIP calls? (Choose three.)
A. SIP cause codes
B. media flow-around
C. media flow-through
D. codec transparent support
E. Transport Layer Security
F. H.261, H.263, and H.264 video codecs
Correct Answer: CDE Section: Call Signaling Explanation Explanation/Reference:
QUESTION 4
Which two are attributes of SCCP? (Choose two.)
A. It is Cisco proprietary.
B. It is a supervisory signaling protocol.
C. It is classified as client/server architecture.
D. SCCP devices are considered intelligent endpoints.
Correct Answer: AC Section: Call Signaling Explanation
Explanation/Reference:
QUESTION 5
Refer to the exhibit. All IP phones are SCCP phones. Phone D makes an internal call to phone G. Which call setup signaling statement is true?
Exhibit:

A. Phone D signals phone G directly. Call setup is handled by the phones.
B. Phone D signals gateway A, which processes the call and signals phone G.
C. Phone D signals gateway B, which processes the call and signals phone G.
D. Phone D signals gatekeeper. The gatekeeper processes the call and signals phone G.
E. Phone D signals the call agent. The call agent processes the call and signals phone G.
Correct Answer: E Section: Call Signaling
Explanation Explanation/Reference:
QUESTION 6
Which statement is true about MGCP?
A. Call completion is always shared, with some intelligence on the endpoint, some on the call agent.
B. Endpoints always take all actions to complete calls.
C. Endpoints may act alone or cooperate with call agent to complete calls.
D. Call agents order and direct each step of call completion for the endpoints.
Correct Answer: D Section: Call Signaling Explanation
Explanation/Reference: Exam D
QUESTION 1
You work as a network technician , study the exhibit carefully. The Acme Corp. uses H.323 to place calls to their supplier RR Industries. Acme also has a voice connection to an ITSP for long distance over a SIP network. Which configuration should Acme use to deploy the CUBE?
Hot Area:

Correct Answer: Section: Internet Telephony Service Provider Explanation
Explanation/Reference:
The Acme Corp connects to the ITSP via SIP Trunk and connects to RR industries via H.323. The Acme
Corp itself uses H.323 so we have to enable protocol interworking with allow-connections commands:

allow-connections h323 to h323: allow Acme Corp to communicate with RR industries (in both ways)
allow-connections h323 to sip: allow Acme Corp to talk with ITSP (Acme Corp can talk and ITSP can hear
but not vice versa)
allow-connections sip to h323: allow ITSP to talk with Acme Corp (Acme Corp can hear and ITSP can talk
but not vice versa)

Notice that the configuration for H.323 and SIP interworking is unidirectional, thus if bidirectional
interworking is required, you need to configure the mirror-matching statement as well.

Acme Corp doesn’t use SIP so we don’t need to configure “allow-connections sip to sip”.

QUESTION 2
H.323 is an umbrella Recommendation from the ITU Telecommunication Standardization Sector (ITU-T) that defines the protocols to provide audio-visual communication sessions on any packet network. Which CUBE configuration will support H.323 protocol interworking and address hiding?
A. voice services voip
h323 interworking
media flow-around

B. voice services h323 to h323 h323 interworking media flow-through
C. voice services voip allow-connections h323 to h323 media flow-around
D. voice service voip allow-connections h323 to h323
Correct Answer: D Section: Internet Telephony Service Provider Explanation
Explanation/Reference:
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Exam A
QUESTION 1
Refer to the exhibit. You have been asked to configure a dial peer on R2 that will match only the extensions of the four telephones attached. Which dial-peer statement will you use?

A. dial-peer voice 1 potsdestination-pattern 5552.[0-5]0
B. dial-peer voice 1 potsdestination pattern 5552[5-6].0
C. dial-peer voice 1 potsdestination-pattern 555[2-5][56]
D. dial-peer voice 1 potsdestination-pattern 5552[5-6][05]0
Correct Answer: D Section: Multiple Choice Explanation
Explanation/Reference:
QUESTION 2
Refer to the exhibit. When extension 201-555-1000 dials 404-555-1200, how are the digits manipulated in R1 so that they are presented correctly at R2?

A. The outbound VoIP dial peer is matched and all digits are sent.
B. The digits 404-555 are stripped off before matching the outbound POTS dial peer.
C. The digits 404-555 are stripped off by the connection trunk and R2 receives only 1200.
D. R1 collects the 1200 and prepends the tie-line digits 404555. That number is matched to a VoIP dial peer and sent to the appropriate address.
Correct Answer: A Section: Multiple Choice Explanation
Explanation/Reference:
QUESTION 3
Refer to the exhibit. Your customer wants to converge the existing PBX network with the IP network. The three remote offices have various types of PBXs. The customer is using a combination of tie-lines and trunks to connect the PBXs today. Which kind of connection should be implemented to allow calls to be placed from 201-555-1000 to 727-555-1000 so that when the call is completed, network resources are returned for other uses?

A. PLAR
B. trunk
C. tie-line
D. answer-mode
Correct Answer: C Section: Multiple Choice Explanation
Explanation/Reference:
QUESTION 4
Which dial plan characteristic shows the most obvious improvement by dropping a number translation step?
A. availability
B. post-dial delay
C. scalability
D. hierarchical design
Correct Answer: B Section: Multiple Choice Explanation
Explanation/Reference:
QUESTION 5
Which option is true concerning the MGCP call agent?
A. acts only as a recorder of call details
B. provides only call signaling and call setup
C. manages all aspects of the call and voice stream
D. monitors the quality of each call after setup
Correct Answer: B Section: Multiple Choice Explanation
Explanation/Reference: QUESTION 6
At what point does the MGCP call agent release the setup of the call path to the residential gateways?
A. after the call agent has been notified that an event 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. does not release call path setup
D. after the call agent has sent a connection request 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: Multiple Choice Explanation
Explanation/Reference:
QUESTION 7
Refer to the exhibit for IP addresses and telephone numbers. You are working with a customer opening a small sales office in Atlanta. You want the user in Atlanta to be able to dial into the PBX in New York over the IP WAN. The New York PBX uses ground start, a two-wire operation, and DTMF dialing. Choose the correct FXO port configuration commands for New York.

A. voice-port 1/0/0signal ground-startoperation 2-wiredial-type dtmf
B. voice-port 1/1/1destination 2015551212 signal ground-startoperation 2-wiretype 1dial-type dtmf
C. voice port 1/0/0session target ipv4:172.16.1.1destination 2015551212 signal ground-startoperation 2-wiredial-type dtmf
D. voice port 1/0/0session target ipv4:172.16.1.1source 2015551212 signal wink-startoperation 2-wire
dial-type dtmf
Correct Answer: A Section: Multiple Choice Explanation
Explanation/Reference:
QUESTION 8
Refer to the exhibit.

Lighthorse Equine Management would like to investigate converging voice and data on their existing T1 Frame Relay WAN link between New York and Atlanta. Currently the following list of applications are consuming no more bandwidth than what is listed on this segment of the network.
T1 link 1536 kbps e-mail 75 kbps internet 200 kbps Oracle 500 kbps FTP 250 kbps Total 1025 kbps
The customer has allocated 25% of the WAN link for routing updates and other overhead. They would like to increase the number of samples encapsulated in each PDU to 40 ms. You have calculated 6 bytes of overhead for Frame Relay, no cRTP, and the use of the G.711 codec. How many simultaneous calls could be placed on this link?
A. 0 calls
B. 1 call
C. 2 calls
D. no more than 5 calls
E. no more than 10 calls
F. no more than 20 calls
Correct Answer: B Section: Multiple Choice Explanation
Explanation/Reference:
QUESTION 9
Refer to the exhibit. A QoS strategy has already been deployed on the LAN. Choose three WAN QoS best practices that should be used over the WAN link. (Choose three.)

A. Implement NBAR.
B. Implement admission control.
C. Mark voice traffic as EF in DSCP.
D. Mark voice traffic highest priority in 802.1p.
E. Use cRTP to maximize bandwidth utilization.
F. Configure access switches to trust traffic from IP phones.
Correct Answer: BCE Section: Multiple Choice Explanation
Explanation/Reference:
QUESTION 10
When setting up a VoIP call, what is the first thing a gateway router tries to match to a dialed number?
A. call leg
B. IP route
C. session target
D. destination pattern
Correct Answer: D Section: Multiple Choice Explanation
Explanation/Reference:
QUESTION 11
Refer to the exhibit. Users are not able to complete a call from 678-555-1212 to 770-555-1111. What is the correct diagnosis for the problem?

A. incorrect destination-pattern in router 1
B. incorrect POTS dial-peer statement in router 2
C. incorrect session-target statement in router 2
D. incorrect port statement in router 1 pots dial peer
E. missing no digit-strip on the voip dial peer in router 1
Correct Answer: A Section: Multiple Choice Explanation
Explanation/Reference:
QUESTION 12
The SJ local zone contains a gatekeeper that controls two gateways, SJ1 and SJ2. Both gateways provide access to area code 408. Which two command strings should be entered into the gatekeeper to give the SJ2 gateway priority over the SJ1 gateway? (Choose two.)
A. zone prefix SJ 408 gw-priority 6 SJ1
B. zone prefix SJ 408 gw-priority 6 SJ2
C. zone prefix SJ 408 gw-priority 10 SJ1
D. zone prefix SJ 408 gw-priority 10 SJ2
E. zone prefix SJ 408 gw-priority 0 SJ2, 10 SJ1
F. zone prefix SJ 408 gw-priority 6 SJ1, 10 SJ2
Correct Answer: AD Section: Multiple Choice Explanation
Explanation/Reference:
QUESTION 13
Refer to the exhibit. Highland Park Property Development is integrating a Cisco Unified Communications Manager Express system with the existing PBX via an E1 QSIG trunk. After the initial configuration, no
calls can be placed from IP phones to PBX phones. How can this problem be resolved?

A. Increase the ISDN T302 timer to allow more time for call setup.
B. Add the command isdn negotiate-bchan to the serial interface.
C. Add the command isdn contiguous-bchan to the serial interface.
D. Change the channel selection order from descending to ascending.
Correct Answer: B Section: Multiple Choice Explanation
Explanation/Reference:
QUESTION 14
Refer to the exhibit. The Carmichael caller dials the site access code for Merrimack (6) followed by the four-digit extension number of the destination phone (0124). If the call is going to go across the IP WAN, which action will have to be taken?

A. Translate 50124 to 5125550124.
B. Strip the site access code and send four digits.
C. Strip the site access code and prepend 1512555.
D. Do nothing because the site access code matches the last five digits of the target number.
E. Strip the site access code, send four digits, then prepend the access code when it reaches the Merrimack gateway.
Correct Answer: B Section: Multiple Choice Explanation
Explanation/Reference:
QUESTION 15
Which path selection mechanism lets you choose either the even or odd channels first?
A. hunt groups
B. trunk groups
C. tailend hopoff
D. Call Admission Control
Correct Answer: B Section: Multiple Choice Explanation
Explanation/Reference:
QUESTION 16
Which mechanism do you use to implement calling privileges on Cisco Unified Communications Manager Express?
A. CoS
B. QoS

C. CAC
D. COR
E. SRST

Correct Answer: D Section: Multiple Choice Explanation
Explanation/Reference:
QUESTION 17

Refer to the H.323 message in the exhibit. What is the gateway doing with the gatekeeper?

A. initial registration
B. full registration

C. lightweight registration
D. registration retry

Correct Answer: C Section: Multiple Choice Explanation
Explanation/Reference:
QUESTION 18

In which three RAS messages is the technology prefix sent? (Choose three.)
A. GRQ
B. RRQ
C. RCF
D. IRR
E. IRQ
Correct Answer: ABE Section: Multiple Choice Explanation
Explanation/Reference:
QUESTION 19
Refer to the output from the debug h225 asn1command in the exhibit. You have configured a gatekeeper with two local zones, hq and br. You want the gateway at the branch location to register with zone BR. What needs to be corrected in the branch gateway to resolve the issue?
A. Change the IP address in the h323-gateway voip id command.
B. Change the gatekeeper-id in the h323-gateway voip id command.
C. Add a zone remote for zone BR so the gateway can register with the correct zone.
D. Change the gatekeeper-id and the IP address in the h323-gateway voip id command.
Correct Answer: B Section: Multiple Choice Explanation
Explanation/Reference:
QUESTION 20
Which three services are supported by CUBE when supporting H323-to-SIP calls? (Choose three.)
A. SIP cause codes
B. media flow-around
C. media flow-through
D. codec transparent support
E. Transport Layer Security
F. H.261, H.263, and H.264 video codecs
Correct Answer: CDE Section: Multiple Choice Explanation
Explanation/Reference:
QUESTION 21
You have been asked to deploy a gatekeeper to support CUBE that will connect your organizational domain to the domain of an Internet Telephony Service Provider so that callers can reach the 407 area code. Which configuration will support this function?
Correct Answer: D Section: Multiple Choice Explanation
Explanation/Reference:
QUESTION 22
When using CUBE, which two statements describe how media flow-through differs from media flow-around? (Choose two.)
A. Media flow-around provides address hiding by terminating both signaling and RTP streams.
B. Media flow-through terminates the signaling channel and the RTP streams flow directly between endpoints.
C. Media flow-around and media flow-through function in a similar manner, but media flow-around supports NAT traversal.
D. Media flow-through terminates the RTP streams but allows signaling to flow directly between endpoints.
E. Media flow-around terminates the signaling stream and allows RTP streams to flow directly between endpoints.
F. Media flow-through provides address hiding by terminating both signaling and RTP streams.
Correct Answer: EF Section: Multiple Choice Explanation
Explanation/Reference:
QUESTION 23
Which CUBE configuration will support H.323 protocol interworking and address hiding?
Correct Answer: D Section: Multiple Choice Explanation
Explanation/Reference:
QUESTION 24
Refer to the exhibit. The Acme Corp. is deploying a CUBE. As a component of protocol interworking between RR Industries and the ITSP, they need to configure at least two dial peers. When the IP WAN is functional, Acme Corp. wants to use 5-digit dialing to RR Industries. Which three dial peers will complete the configuration for Acme Corp.? (Choose three.)
Correct Answer: BCF Section: Multiple Choice Explanation
Explanation/Reference:
QUESTION 25
Refer to the exhibit. You have configured a gatekeeper and an IP-IP gateway on the same router. When you look at the output from the show gatekeeper endpoint command, the IP-IP gateway is not registered with the gatekeeper. What needs to be configured to resolve this issue?

A. You need to stop and restart the gateway.
B. You need to add a VoIP dial peer to the configuration.
C. The h323-gateway voip id command has an incorrect IP address.
D. The h323-gateway voip id command has an incorrect gatekeeper ID and IP address.
Correct Answer: B Section: Multiple Choice Explanation
Explanation/Reference:
QUESTION 26
A new business in Great Britain needs to have a PSTN connection that will handle a maximum of 30 inbound and outbound calls at any given time. The customer only has one slot available on the designated PSTN router. Which digital line type should be recommended?
A. QSIG
B. ISDN BRI
C. ISDN E1 PRI

D. ISDN T1 PRI
Correct Answer: C Section: Multiple Choice Explanation
Explanation/Reference:
QUESTION 27
A customer needs to configure a CAS E&M circuit that will support inbound and outbound DNIS and inbound ANI. Which configuration will accomplish this task?
A. pri-group timeslots 1-24
B. ds0-group 0 timeslots 1-24 type none
C. ds0-group 0 timeslots 1-24 type e&m-fgd
D. ds0-group 0 timeslots 1-24 type fgd-eana
E. ds0-group 0 timeslots 1-31 type r2-digital r2-compelled ani
Correct Answer: C Section: Multiple Choice Explanation
Explanation/Reference:
QUESTION 28
Refer to the IOS configuration in the exhibit. How will the next incoming call be routed?

A. The call will be routed to the longest idle channel.
B. The call will be routed to the least used channel.
C. The call will be routed to a random available channel.
D. The call will be routed to the next available channel, starting from channel 1, hunting up toward channel
E. The call will be routed to the next available channel, starting from channel 24, hunting down toward channel 1.
Correct Answer: E Section: Multiple Choice Explanation
Explanation/Reference:
QUESTION 29
A telemarketing firm needs to use number translation for incoming and outgoing calls. They have defined two translation profiles, one for incoming and one for outgoing calls. What can be used to simplify this task?
A. dial peer
B. voice port
C. hunt group
D. trunk group
E. source IP group
Correct Answer: D Section: Multiple Choice Explanation
Explanation/Reference:
QUESTION 30
Refer to the exhibit. Three department managers share the directory number 3000. The Marketing manager’s phone is attached to port 1/1. The Engineering manager’s phone is attached to port 1/2. The Shipping manager’s phone is attached to port 1/3. In which situation would an incoming call ring on the Shipping manager’s phone?

A. The Marketing manager is on the phone.
B. None of the managers are on the phone.
C. The Engineering manager is on the phone.
D. The Shipping manager and Marketing manager are on the phone.
E. The Engineering manager and Marketing manager are on the phone.
Correct Answer: E Section: Multiple Choice Explanation
Explanation/Reference:

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