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Exam A
QUESTION 1
Which three Cisco IOS commands are required to configure a voice gateway as a DHCP server to support a data subnet with the IP address of 10.1.30.0/24 and a default gateway of 10.1.30.1/24? (Choose three.)
A. ip dhcp pool
B. subnet 10.1.30.1 255.255.255.0
C. ip dhcp pool data
D. network 10.1.30.1/24
E. network 10.1.30.0 255.255.255.0
F. default-gw 10.1.30.1/24
G. default-router 10.1.30.1
Correct Answer: CEG Section: (none) Explanation
Explanation/Reference:
1) To configure the DHCP address pool name and enter DHCP pool configuration mode, use the following command in global configuration mode:
Router(config)# ip dhcp pool name – Creates a name for the DHCP Server address pool and places you in DHCP pool configuration mode
2) To configure a subnet and mask for the newly created DHCP address pool, which contains the range of available IP addresses that the DHCP Server may assign to clients, use the following command in DHCP pool configuration mode:
Router(dhcp-config)# network network-number [mask | /prefix-length] – Specifies the subnet network number and mask of the DHCP address pool. The prefix length specifies the number of bits that comprise the address prefix. The prefix is an alternative way of specifying the network mask of the client. The prefix length must be preceded by a forward slash (/).
3) After a DHCP client has booted, the client begins sending packets to its default router. The IP address of the default router should be on the same subnet as the client. To specify a default router for a DHCP client, use the following command in DHCP pool configuration mode:
Router(dhcp-config)# default-router address [address2 … address8] – Specifies the IP address of the default router for a DHCP client. One IP address is required; however, you can specify up to eight addresses in one command line.
http://www.cisco.com/en/US/docs/ios/12_2/ip/configuration/guide/1cfdhcp.html#wp1000999
QUESTION 2
Which four Cisco IOS commands are required to configure a DHCP server on a voice gateway to support a voice subnet so that both IP addresses and the IP address of the TFTP server are provided? The voice subnet has an address of 10.1.130.0/24, the default gateway is 10.1.130.1/24, and the TFTP server is located at 10.1.5.2. (Choose four.)
A. subnet 10.1.130.1/24
B. ip dhcp pool voice
C. default-router 10.1.130.1
D. option 150 10.1.5.2
E. network 10.1.130.0 255.255.255.0
F. dhcp pool voice
G. option 150 ip 10.1.5.2
H. default-gw 10.1.130.1
Correct Answer: BCEG Section: (none) Explanation
Explanation/Reference:
QUESTION 3
The router with the IP address of 10.1.120.1 needs to be configured to use the device 10.1.140.1 as the clock source. Which configuration command will accomplish this task?
A. clock source 10.1.140.1
B. ntp server 10.1.140.1
C. clock set 10.1.140.1
D. ntp source ip addr 10.1.140.1
E. ntp client 10.1.120.1 server 10.1.140.1
Correct Answer: B Section: (none) Explanation
Explanation/Reference:
: To configure your routers to use a NTP server for time synchronization, the command ntp server, followed by the IP address or hostname of the NTP server, is used. To specify additional timeservers for redundancy, simply repeat the ntp server command with the IP address of each additional server. http://www.cisco.com/en/US/products/hw/switches/ps700/products_tech_note09186a008010e97e. shtml
QUESTION 4
Which four types of ephone-dns are supported by SCCP in Cisco Unified Communications Manager Express? (Choose four.)
A. single-line
B. dual-line
C. shared-line, nonexclusive
D. two directory numbers with one telephone number
E. dual-number
F. octo-line
Correct Answer: ABEF Section: (none) Explanation
Explanation/Reference:
QUESTION 5
In which situation would an administrator configure telephony services, but not configure any individual ephones?
A. Phones that are controlled by Cisco Unified Communications Manager Express
B. Cisco Unified Communications Manager SRST fallback
C. Cisco Unified Communications Manager Express with HSRP
D. Remotely located phones that are controlled by a third-party PBX
E. This is not a valid scenario. Ephones are always required.
Correct Answer: B Section: (none) Explanation Explanation/Reference:
: When a phone registers for SRST service with a Cisco Router and the router discovers that the phone was configured with a specific extension number, the router searches for an existing prebuilt ephone-dn with that extension number and then assigns that ephone-dn number to the phone. If there is no prebuilt ephone-dn with that extension number, the system automatically creates one. In this way, extensions without prebuilt configurations are automatically populated with extension numbers and features as the numbers and features are “learned” by the Cisco router in SRST mode when the phone registers to the router after a WAN link fails.
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmesrst.html
QUESTION 6
Refer to the exhibit. Which type of ephone-dn is configured for the two ephones that are shown?
A. single-line-octo
B. hunt line
C. shared-line, nonexclusive
D. two directory numbers with one telephone number
E. shared-line, overlay
F. octo-line
Correct Answer: E Section: (none) Explanation
Explanation/Reference:
: The above exhibit shows the configuration for a simple shared-line overlay set. The primary ephone-dn that is configured for each phone is unique while the remaining ephone-dns 10, 11, and 12 are shared in the overlay set on both phones. The primary ephone-dn in a shared- line overlay set is configured unique to the phone to guarantee that the phone has a line available for outgoing calls, and to ensure that the phone user can obtain dial-tone even when there are no idle lines available in the rest of the shared-line overlay set. Using a unique ephone-dn also provides a unique calling party identity on outbound calls made by the phone so that the called user can see which specific phone is calling. http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmecover.htm l#wp1099687
QUESTION 7
Refer to the exhibit.
A new Cisco Unified Communications Manager Express system has been deployed and the technician is trying to add the first new IP phone to the system. The phone powers up, but it does not register with the system. The technician has verified that the phone is getting the proper VLAN information from Cisco Discovery Protocol. The phone is also getting the correct IP address and TFTP server address from DHCP. The phone has been assigned to an ephone and the correct MAC address is configured. With the information provided, which two of the following does the administrator need to verify to resolve this situation? (Choose two.)
A. Verify that the ip helper-address is correctly configured.
B. Verify that telephony-service has been configured.
C. Verify that the ephone has a button assigned.
D. Verify that the tftp-server path has been configured.
E. Verify that the Cisco Unified Communications Manager Express service is running.
F. Verify that the correct phone type files are in the tftp-server path.
Correct Answer: DF Section: (none) Explanation
Explanation/Reference:
: Since the phone is getting the correct TFTP address, the next thing that needs to be verified is the TFTP Server path and IP Reachablity for the IP Phone to the TFTP Server. Once the TFTP settings has been verified, check if the files mentioned in the termxx.defaults.loads file is available in the TFTP Server for the phone to download. http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/7960g_7940g/7_0/sip/ english/administra tion/guide/7960trbS.html
QUESTION 8
The administrator has added a new ephone-dn and a new ephone to the Cisco Unified Communications Manager Express system, but the new phone will not register with the system. If other phones are operating properly, which of the following should the administrator do first to try to resolve the issue?
A. Reboot the router.
B. Remove the ephone, then re-add the ephone.
C. Verify that the url authentication is configured for the correct authentication URL.
D. Verify that the url services is configured to the correct URL for services.
E. Enter the command no telephony-service, then enter telephony service in global configuration mode.
F. Enter the command no create cnf-files, then enter create cnf-files under the telephony-service configuration.
Correct Answer: F Section: (none) Explanation
Explanation/Reference:
QUESTION 9
Refer to the exhibit.
Cisco Unified Communications Manager Express has been partially configured to support 6 IP phones and 12 directory numbers. The Cisco Unified Communications Manager Express will use the IP address 10.1.130.1/24. Which two elements of the configuration are missing from the command output and need to be added so that phones do not auto-register, but can manually register with Cisco Unified Communications Manager Express? (Choose two.)
A. ip address 10.1.130.1
B. no reg-ephone
C. create profile
D. ip source-address 10.1.130.1
E. create cnf-files
F. no auto-reg-ephone
Correct Answer: DF Section: (none) Explanation
Explanation/Reference:
: To identify the IP address and port through which IP phones communicate with a CiscoUnifiedCME router, use the ip source-address command in telephony-service or group configuration mode. This command enables a router to receive messages from CiscoUnifiedIPphones through the specified IP address and port. The CiscoUnifiedCME router cannot communicate with CiscoUnifiedCME phones if the IP address of the port to which they are attached is not configured.
Normally when you configure basic telephony-service parameters, then phone can register with CME although no DN will be assigned to them. You can disable this by using the no auto-reg- ephone command. After this command the phone which will try to register will receive message “Registration Rejected: No configuration entry…..”.. When automatic registration is blocked, CiscoUnifiedCME records the MAC addresses of phones that attempt to register but cannot because they are blocked. http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/command/reference/cme_a1ht.html#wp 1031242
QUESTION 10
Which three functions are associated with MGCP? (Choose three.)
A. Control is implemented by a series of plain-text commands that are sent over UDP port 2427 between Cisco Unified Communications Manager and the gateway.
B. A PRI backhaul channel forwards PRI Layer 2 (Q.921) signaling information via a TCP connection from the gateway to the call agent.
C. MGCP uses a separate channel for backhauling signaling information between the call agent and the gateway.
D. The gateway maintains a separate dial plan for redundancy in case the call agent fails.
E. Users query the call agent to determine the location of the call recipient.
F. A call agent uses control messages to direct its gateways and their operational behavior.
Correct Answer: ACF Section: (none) Explanation
Explanation/Reference:
: MGCP is a plain-text protocol used by call-control devices to manage IP Telephony gateways. MGCP is a master/slave protocol that allows a call control device to take control of a specific port on a gateway. With this protocol, the Cisco CallManager knows and controls the state of each individual port on the gateway. It allows complete control of the dial plan from Cisco CallManager, and gives CallManager per-port control of connections to the PSTN, legacy PBX, voice mail systems, POTS phones and so forth. This is implemented with the use of a series of plain-text commands sent over User Datagram Protocol (UDP) port 2427 between the Cisco CallManager and the gateway.
Another concept relevant to the MGCP implementation with Cisco CallManager is PRI Backhaul. This occurs when Cisco CallManager takes control of the Q.931 signaling data used on an ISDN PRI. The one thing that distinguishes a PRI from other interfaces is the fact that the data that is received from the PSTN on the D-channel and needs to be carried in its raw form back to the Cisco CallManager to be processed. The gateway does not process or change this signaling data, it simply passes it onto the Cisco CallManager through TCP port 2428. The gateway is still responsible for the termination of the Layer 2 data. That means that all the Q.921 data-link layer connection protocols are terminated on the gateway, but everything above that (Q.931 network layer data and beyond) is passed onto the Cisco CallManager. This also means that the gateway does not bring up the D-channel unless it can communicate with Cisco CallManager to backhaul the Q.931 messages contained in the D-channel. http://www.cisco.com/en/US/tech/tk1077/technologies_tech_note09186a00801da84e.shtml
QUESTION 11
Refer to the exhibit. An administrator is migrating a PBX telephony system to an IP Phone solution using a fixed numbering plan. The extension numbers and PSTN DIDs cannot be changed. Which of the following methods can be used in order to reach the individual extensions at Site B when called via the PSTN?
A. The administrator can add a 1 to the DID for Site B to become 300-555-31xxx.
B. The administrator needs to map the last four digits in the DID to the extension numbers and prefix a site code.
C. The administrator needs to map the last four digits in the DID to the extension numbers and prefix an intersite code.
D. The administrator needs to map the last four digits in the DID to the extension numbers using translation rules.
E. No changes are necessary because PSTN calls are preceded with access code 9.
Correct Answer: D Section: (none) Explanation
Explanation/Reference:
: Since the extension and PSTN DID is one and the same for the customer, no manipulation is required the Route Plan to reach individual extensions from PSTN DID
QUESTION 12
Which of the following best describes the implementation challenges that are associated with variable-length numbering plans?
A. the variable number of extensions that need to be implemented
B. the number of trunks that need to be assigned
C. the mapping between IP addresses and extension numbers
D. the identification of the number of digits that need to be dialed before the call is routed
E. the degree in which the dial plan varies
Correct Answer: D Section: (none) Explanation Explanation/Reference:
QUESTION 13
Refer to the exhibit. An administrator is migrating a PBX telephony system to a VoIP solution using a fixed numbering plan. The extension numbers and PSTN DIDs cannot be changed. Which of the following methods can be used in order to reach the individual extensions at Site B when called via the PSTN?
A. The administrator can replace the last three digits of the DID with xxx to cover the individual extensions.
B. The administrator can replace the last three digits of the DID with xxx and use translation rules to map the individual extensions.
C. The administrator needs to implement an auto-attendant solution where individual extensions can be dialed.
D. The administrator needs to map the last four digits in the DID to the extension numbers using translation rules.
Correct Answer: D Section: (none) Explanation
Explanation/Reference:
QUESTION 14
Which two statements are true regarding SCCP? (Choose two.)
A. SCCP requires each endpoint or gateway event to be communicated to Cisco Unified Communications Manager.
B. Endpoints can operate autonomously if communication with Cisco Unified Communications Manager is lost.
C. SCCP may interoperate with H.323 endpoints if it is implemented with Cisco Unified Communications Manager.
D. Endpoints and gateways maintain the dial plan.
E. SCCP uses hex messages for communication.
Correct Answer: AC Section: (none) Explanation
Explanation/Reference:
: The Skinny client (i.e. an Ethernet Phone) uses TCP/IP to transmit and receive calls. Skinny messages are carried above TCP and use port 2000. Cisco IP Phones that use SCCP can coexist in an H.323 environment. When used with CUCM, the SCCP client can interoperate with H.323-compliant terminals. The client communicates with the CUCM using TCP/IP-based communication to establish a call with another H.323-compliant end station. Once the CUCM has established the call, the two H.323 end stations use connectionless UDP/IP-based communication for audio transmissions. The CUCM acts as a proxy by processing all H.323 and SIP transactions. This allows the IP Phone to process the VoIP RTP data stream. http://www.cisco.com/en/US/docs/voice_ip_comm/cata/186_188/2_15_ms/english/administration/g uide/ sccp/sccpaaph.pdf
QUESTION 15
You are configuring a network to support voice to the PSTN. One important aspect to the configuration is to be able to determine the individual slot, subunit, and port number from the gateway endpoint identifier. Which signaling protocol is appropriate for this situation?
A. H.323
B. SIP
C. SCCP
D. MGCP
Correct Answer: D Section: (none) Explanation
Explanation/Reference:
: Endpoints are any of the voice ports on the designated gateway. These voice ports provide connectivity to both analog ports and digital trunks to the PSTN. Ports on gateways are identified by endpoints in very specific ways. It is important to note that gateways can have multiple endpoints dependent on the number of ports it contains, and that the endpoints are case insensitive. A sample MGCP endpoint addressing scheme is provided below.
http://www.cisco.com/en/US/tech/tk1077/technologies_tech_note09186a00801da84e.shtml
QUESTION 16
Which two functions are associated with a voice gateway? (Choose two.)
A. switches voice channels between connected analog and digital voice circuits
B. provides voice-messaging services to connected analog and digital voice circuits
C. interconnects two logically separate VoIP networks
D. negotiates endpoint capabilities
E. controls opening and closing of logical channels that are used to carry media streams
Correct Answer: AE Section: (none) Explanation Explanation/Reference:
: The basic function of a gateway is to translate between different types of networks. In a VoIP environment, voice gateways are the interface between a VoIP network and the public switched telephone network (PSTN), a private branch exchange (PBX), or analog devices such as fax machines. In its simplest form, a voice gateway has an IP interface and a legacy telephone interface, and it handles the many tasks involved in translating between transmission formats and protocols. The gateway allows communication between the two networks by performing tasks such as Interfacing with the IP network and the PSTN or PBX, Supporting IP call control protocols, Performing call setup and teardown for calls between the VoIP and PSTN networks by terminating and reoriginating the call media and signaling, Providing supplementary services, such as call hold and transfer, Relaying dual tone multifrequency (DTMF) tones, Supporting analog fax and modems over the IP network. http://www.cisco.com/en/US/prod/collateral/routers/ps5854/product_data_sheet0900aecd8016981 2.pdf
QUESTION 17
Which type of voice port supports immediate-start, wink-start, and delay-start followed by pulse or DTMF tones?
A. FXS
B. FXS-DID
C. FXO
D. E&M
Correct Answer: D Section: (none) Explanation
Explanation/Reference:
QUESTION 18
Which types of voice ports allow a small office to provide outbound DNIS and inbound DID?
A. FXS and FXO
B. FXO and E&M
C. FXS and FXS-DID
D. FXS and E&M
E. FXS-DID and FXO
Correct Answer: E Section: (none) Explanation
Explanation/Reference:
: An FXO trunk is one of the simplest analog trunks available. Because Dialed Number Information Service (DNIS) information can only be sent out to the PSTN, no direct inward dialing (DID) is possible. ANI is supported for inbound calls. Two signaling types exist, loopstart and groundstart, with groundstart being the preferred method. An FXS DID trunk can receive only inbound calls, thus a combination of FXS DID, and FXO ports is required for inbound and outbound calls http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/srnd/design/guide/gatewy.html#wp1052
QUESTION 19
In a voice gateway, the configured codec complexity of the DSPs on a voice card can be changed. What is the impact on the DSPs if high codec complexity is configured?
A. The codec complexity affects call density, which is the number of calls that are reconciled on the DSPs. This results in lower call density when high complexity is configured.
B. With higher codec complexity, more calls can be processed.
C. Lower codec complexity supports the fewest number of voice channels, provided that the lower complexity is compatible with the particular codecs that are in use.
D. The DSP will process codecs that support high complexity transparently and shift to flex mode for those codecs that are not high complexity.
Correct Answer: A Section: (none) Explanation
Explanation/Reference:
: The difference between medium and high complexity codecs is the amount of CPU utilization necessary to process the codec algorithm, and therefore, the number of voice channels that can be supported by a single DSP. For this reason, all the medium complexity codecs can also be run in high complexity mode, but fewer (usually half) of the channels are available per DSP.
http://www.cisco.com/en/US/tech/tk1077/technologies_tech_note09186a00800b6710.shtml#code_ com
QUESTION 20
Which codec complexity type will offer the greatest number of voice channels, provided that the complexity type is compatible with the particular codecs that are in use?
A. low complexity
B. medium complexity
C. high complexity
D. flex complexity
Correct Answer: D Section: (none) Explanation
Explanation/Reference:
QUESTION 21
Your PSTN carrier sends digits to your T1 PRI circuit in a digit-by-digit format. How must the T1 PRI circuit be configured to support this capability?
A. The T1 PRI controller supports either en-bloc or digit-by-digit formats natively.
B. The serial interface that is associated with the T1 controller needs to include the isdn incoming- voice command.
C. The T1 controller needs to include the isdn overlap-receiving command.
D. The serial interface that is associated with the T1/E1 controller needs to include the isdn overlap-receiving command.
Correct Answer: D Section: (none) Explanation
Explanation/Reference:
: Configuring Overlap-receiving on the D-channel changes the way routers behave when receiving ISDN calls. Overlap receiving allows the matching of dial peers as the digits are being received. The router responds to the setup message with a SETUP ACK. This informs the network that it is ready to receive further information messages containing additional call routing elements. http://www.cisco.com/en/US/tech/tk801/tk133/technologies_tech_note09186a00800b48cb.shtml
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