642-437 642-437 Implementing Cisco Unified Communications Voice over IP and QoS v8.0 (CVOICE v8.0) Version 14.

5 Ensure Success With Money Back Guarantee

642-437 QUESTION NO: 1 What is the function of class-based marking? A. Marking packets based on CoS value, IP precedence value, or DSCP value allows Layer 3 frames to be identified and distinguished from other packets. B. Marking frames based only on CoS value or IP precedence value allows Layer 2 frames to be identified and distinguished from other frames. C. Marking packets or frames sets information in the Layer 2 and Layer 3 headers of a packet so that the packet or frame can be identified and distinguished from other packets or frames. D. Marking frames only sets information in the Layer 2 headers of a frame so that the frame can be identified and distinguished from other packets or frames. Answer: A QUESTION NO: 2 Voice packets are arriving at a destination with a variance of between 20 and 50 ms. If the jitter buffer has a capacity of 30 ms, what is the impact on the audio at the receiving IP phone? A. The jitter buffer will replay the previous voice packets to replace those packets that exceed 30 msto avoid speech gaps. B. The audio stream at the receiving IP phone will be delayed and garbled. C. The DSP will automatically increase the jitter buffer size after sampling the range of incoming voice packets to accommodate the wider range in variation of voice packet arrival times to avoid voice gaps. D. The IP phone will negotiate, in mid-call, a lower bandwidth codec to reduce the delay in the arrival of voice packets to avoid voice gaps. Answer: B QUESTION NO: 3 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 Ensure Success With Money Back Guarantee

642-437 D. two directory numbers with one telephone number E. shared-line, overlay F. Octo-line Answer: E QUESTION NO: 4 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 Answer: D, F QUESTION NO: 5 How does LLO ensure that voice traffic is always expedited? A. LLO adds a strict priority class to CBWFO. This class allows delay-sensitive data such as voice to be dequeued and sent first. B. LLO uses CBWFO to prioritize voice traffic and dequeue the voice packets so that they can be handled first. C. The strict priority queue has a higher weight than the queues in CBWFO. This weight allows the delay-sensitive data such as voice to be dequeued and sent first. D. The LLO strict priority queue allows delay-sensitive data such as voice to be dequeued and sent first (before packets in other queues are dequeued), giving delay-sensitive data preferential treatment over other traffic. Answer: A Reference: http://docstore.mik.ua/cisco/CCNP-CCDP/AVVID-NI/AVVID-NI-35.html Ensure Success With Money Back Guarantee

642-437 QUESTION NO: 6 Your PSTN carrier sends digits to your TI 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 digitby-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. Answer: D Reference: http://www.cisco.com/en/US/tech/tk801/tk133/technologies_tech_note09186a00800 b48c b.shtml QUESTION NO: 7 You have a Cisco Unified Border Element configured to provide H.323 to SIP interworking. Which command will verify that you have a single H.323 and a single SIP call leg when the call is placed? A. show call active voice B. debug voip ipipgw C. show dialpeer voice D. debug voice dialpeer Answer: A QUESTION NO: 8 Which QoS technology provides a strict priority queuing scheme that allows delaysensitive data such as voice to be dequeued and sent before packets in other queues are dequeued, and also works with WFQ and CBWFQ. A. header compression B. IP RTP Priority and Frame Relay IP RTP Priority C. RSVP D. low latency queuing E. FRF.12 Answer: D QUESTION NO: 9 What is the decimal equivalent of the DSCP value AF21? A. 16 B. 17 C. 18 D. 21 Ensure Success With Money Back Guarantee

642-437 Answer: C Reference: http://www.cisco.com/en/US/tech/tk543/tk757/technologies_tech_note09186a00800 949f 2.shtml QUESTION NO: 10 DRAG DROP Answer: Ensure Success With Money Back Guarantee

642-437 QUESTION NO: 11 DRAG DROP Answer: Ensure Success With Money Back Guarantee

642-437 QUESTION NO: 12 A new Cisco 7965 IP phone is installed on a Cisco Unified Communications Manager Express system. When the phone requests the .loads file from the TFTP server, it sees that the versions are different. What does the IP phone do to resolve this issue? A. The IP phone requests the SEP<mac>.cfg file and reboots. B. The IP phone attempts to obtain the new firmware file image from the TFTP server. C. The IP phone boot requests the XMLDefault.cnf.xml file and boots up. D. The IP phone does not boot up and will require manual intervention to factory reset the phone before a new firmware image can be downloaded. Answer: B QUESTION NO: 13 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 Answer: D QUESTION NO: 14 DRAG DROP Ensure Success With Money Back Guarantee

642-437 Answer: QUESTION NO: 15 How many IP phone calls can be sent across a 64-kb/s Frame Relay link that uses the G.729 codec? The sampling rate is 50 times a second, with 20 bytes per sample. There are 8 bytes of Frame Relay header overhead with no checksum, and header compression is used. A. 3 B. 4 C. 5 D. 7 Answer: C QUESTION NO: 16 When Cisco Unified Border Element is configured to support RSVP-based CAC, at which point during call setup are the RSVP path and reservation messages sent and received? A. The path message is sent immediately after the call setup message is received and the reservation message is received after H.245 capabilities negotiation is completed. Ensure Success With Money Back Guarantee

642-437 B. The reservation message is sent immediately after the call setup message is received and the path message is received after H.225 call setup messages have been sent. C. The path and reservation messages are sent and received after the H.245 capabilities negotiation is completed. D. The path and reservation messages are sent and received immediately after the call setup message is received. Answer: D QUESTION NO: 17 Which command should be included in order to trust the DSCP-marked traffic from the distribution layer? A. mis qos trust cos B. mis trust dscp-cos C. mis qos trust dscp D. mis qos trust dscp-cos Answer: C QUESTION NO: 18 What are the PHBs that DiffServ use? A. resource reservation and admission control B. default, AF, and EF PHBs C. AF, EF, and CS PHBs D. AF and EF PHBs E. default, AF, EF, and CS PHBs Answer: E QUESTION NO: 19 DRAG DROP Ensure Success With Money Back Guarantee

642-437 Answer: QUESTION NO: 20 Ensure Success With Money Back Guarantee

642-437 Refer to the exhibit. Your companyas QoS policy states that all traffic that is arriving at access layer switches from IP phones should be marked with a DSCP value of 46 and that all untagged traffic that is arriving from a PC that is attached to an IP phone should be marked with a CoS value of 1. Which two options will satisfy the requirements for the CoS-to-DSCP map and are the correct QoS commands? (Choose two.) A. mis qos 1 B. mis qos map cosdscp 0 10 18 26 34 46 48 56 C. mis qos cos 1 D.mis qos map dscp 0 8 16 26 32 40 48 56 E. mis qos map cos 0 8 18 26 40 48 50 56 F. mis qos dscp 1 Answer: B, C QUESTION NO: 21 What is the reason that an outgoing call succeeds when there is no COR list that is applied to the incoming dial peer and a COR list is applied to the outgoing dial peer? A. The COR list for incoming calls on the incoming dial peer is a superset of COR lists for outgoing calls on the outgoing dial peer. B. COR lists for incoming calls on the incoming dial peer are not a superset of COR lists for outgoing calls on the outgoing dial peer. C. The outgoing dial peer, by default, has the lowest priority. D. The incoming dial peer, by default, has the highest COR priority when no COR is applied. Answer: D QUESTION NO: 22 Ensure Success With Money Back Guarantee

642-437 Refer to the exhibit. When an inbound PSTN call from 4087071222 arrives at the ISDN port that is shown in the exhibit, which dial peer will be matched for the inbound leg? A. Dial-peer 123, because destination-pattern takes precedence over answer-address. B. Dial-peer 2123, because answer-address takes precedence over destination-pattern. C. The matching inbound dial peer will be selected at random. D. Although dial-peer 2123 takes precedence, it will not be matched because the command direct-inward-dial is missing. E. Dial-peer 123 will be matched because dial-peer 2123 will strip all the digits. Answer: B QUESTION NO: 23 Calls are failing to egress the local PSTN gateway that uses an E1 PRI circuit. Which debug command would be most useful in determining which dialed digits are being sent to the PSTN? A. debug voice dial-peer B. debug isdn q921 C. debug isdn q931 D. ccapi inout Answer: C QUESTION NO: 24 Ensure Success With Money Back Guarantee

642-437 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 300-55531xxx. 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 Answer: D. QUESTION NO: 25 DRAG DROP Ensure Success With Money Back Guarantee

642-437 Answer: QUESTION NO: 26 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 wi 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 Answer: B QUESTION NO: 27 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 Ensure Success With Money Back Guarantee

642-437 Answer: A, E QUESTION NO: 28 Refer to the exhibit. When an inbound PSTN call to 4087071222 is received by the router that is shown in the exhibit, what is the resulting called number? A. 14087071222 B. 11222 C. 14081222 D. 1222 E. 4Q87071222 Answer: D QUESTION NO: 29 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 (0.921) 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. Answer: A, C, F QUESTION NO: 30 DRAG DROP Ensure Success With Money Back Guarantee

642-437 Answer: QUESTION NO: 31 Ensure Success With Money Back Guarantee

642-437 Refer to the exhibit. Callers dial 0 to reach an outside line. When they try to place calls to directory services (322) or services (422), they hear the reorder tone. What needs to be edited in the dial peer to allow these calls to complete successfully? A. The destination pattern is incorrect. It needs to start with a 9. B. A "prefix 1" statement needs to be added to the dialpeer configuration. C. The forward-dig its all command needs to be applied to the dial peer. D. The destination pattern needs to be edited so that the first digit that is matched is a 0. E. The destination pattern needs to be edited so that the first digit that is matched is a 0 and the forward-digits all command needs to be added to the dial peer. F. The destination pattern needs to be edited so that the first digit that is matched is a 1 and the forward-digits all command needs to be added to the dial peer. G. The destination pattern needs to be edited so that the first digit that is matched is a 0 and the forward-digits 3 command needs to be added to the dial peer. Answer: G QUESTION NO: 32 In which way does SIP Early Media differ from SIP Delayed Offer? A. In SIP Early Media mode, the SDP media capabilities are exchanged in the INVITE and the 200 OK messages. B. SIP Early Media uses session indicator 183. C. The INVITE message includes an SDP-formatted list of media capabilities (including codecs) that are supported by the originating UA. D. The 200 OK message carries an SDP message with the final media description that has been selected by the terminating UA based on the received list and the locally supported options. Answer: B QUESTION NO: 33 Ensure Success With Money Back Guarantee

642-437 Refer to the exhibit. Which class is always present even though it is not in the configuration snip? A. class best-effort B. class class-default C. default class D. best-effort class E. class class-scavenger Answer: B QUESTION NO: 34 A small office needs to provide outbound dialing and inbound DID without the cost of a T1 circuit. All signaling is loop-start. Which analog port configuration will support these requirements? A. voice-port 0/0/0/ description fxs-did signal did loop-start Ensure Success With Money Back Guarantee

642-437 i voice-port 0/1/0 description fxo signal loop-start i dial-peer voice 1 pots incoming called-number. direct-inward-dial port 0/0/0 i dial-peer voice 90 pots destination-pattern 9T port 0/1/0 voice-port 0/0/0 signal loop-start i voice-port 0/1/0 signal loop-start i dial-peer voice 1 pots incoming callednumberT direct-inward-dial ! dial-peer voice 90 pots destination-pattern 9T port 0/1/0 voice-port 0/1/0 signal did loop-start !" dial-peer voice 1 pots incoming called-number ! dial-peer voice 90 pot destination-pattern 9T port 0/1/0 voice-port 0/0/0 signal did loop-start !" dial-peer voice 1 pots incoming called-number direct-inward-dial i dial-peer voice 90 pots destination-pattern 9T port 0/0/0 B. C. D. Answer: B Ensure Success With Money Back Guarantee

642-437 QUESTION NO: 35 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 SRSTfallback C. Cisco Unified Communications Manager Express with HSRP D. Remotely located phones that are controlled by a thirdparty PBX E. This is not a valid scenario. Ephones are always required. Answer: B QUESTION NO: 36 DRAG DROP Answer: Ensure Success With Money Back Guarantee

642-437 QUESTION NO: 37 When configuring Cisco AutoQoS VoIP on a Cisco catalyst switch, how is the configuration performed? A. The auto qos voip command is applied to each interface. B. The auto qos voip command is applied globally in the switch. C. Each interface will need either the auto qos voip ciscophone or auto qos voip trust on each interface, depending on the upstream device D. Each interface will need either the auto qos voip trust cisco-phone or auto qos voip trust trust on each interface, depending on the upstream device. Answer: C QUESTION NO: 38 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 Answer: E Explanation: DID is supported on E&M and FXS-DID ports. E&M supports Inbound and Outbound DNIS and DID. Ensure Success With Money Back Guarantee

642-437 An FXO trunk is one of the simplest analog trunks Number Information Service (DNIS) information can PSTN, no direct inward dialing (DID) is possible. receive only inbound calls, thus a combination of required for inbound and outbound calls QUESTION NO: 39 DRAG DROP Answer: Ensure Success With Money Back Guarantee

available. Because Dialed only be sent out to the An FXS DID trunk can FXS DID, and FXO ports is

642-437 QUESTION NO: 40 When a Cisco 7965 IP phone downloads an SCCP firmware file package, which file is downloaded first by the IP phone from the TFTP server to describe the files it should request from the TFTP server? A. term45.default.loads B. cnu45.8-4-0-79.sbn C. term65.default.loads D. dsp45.8-4-0-79.sbn E. apps45.9-0-2ES2.sbn F. SCCP45.9-0-2SR1S.loads G. dsp45.8-4-0-79.sbn H. jar45sccp.8-4-0-79.sbn I. cvm45sccp.8-4-0-79.sbn Answer: A QUESTION NO: 41 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. Answer: A, C QUESTION NO: 42 DRAG DROP Ensure Success With Money Back Guarantee

642-437 Answer: QUESTION NO: 43 Ensure Success With Money Back Guarantee

642-437 Refer to the exhibit. What will the class map do if a packet arrives that is marked with a CoS of 6 and a DSCP value of EF? A. The class map will match the packet and forward it to the policy map to be marked. B. The class map will not map the packet and no OoS will be applied. C. The class map will wait for the next packet in the stream to see if it has a CoS marking of 5 and then forward both packets to the policy map D. For the packet to be forwarded to the policy map, it must have a CoS of 5 and a DSCP value of EF. Answer: B QUESTION NO: 44 How many VoIP G.729 calls can be made simultaneously over a 128-kb/s Frame Relay circuit (Layer 3) if 50 percent of the circuit is dedicated to voice and 50 percent is dedicated to data? A. 1 B. 2 C. 3 D. 4 E. 5 Answer: B QUESTION NO: 45 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. Ensure Success With Money Back Guarantee

642-437 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 Answer: A QUESTION NO: 46 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 Answer: D, F QUESTION NO: 47 How does Packet Loss Concealment improve voice quality? A. Cisco Packet Loss Concealment technology decreases the voice sampling rate to 10 ms of the voice payload to smooth gaps in the voice stream. B. Packet Loss Concealment intelligently analyzes missing packets and generates a reasonable replacement packet to improve the voice quality C. Packet Loss Concealment will buffer 20 to 50 ms of a voice stream to minimize lost or out-of-order voice packets. D. Packet Loss Concealment will compensate for packet loss rates between 1 and 5 percent by generating a reasonable replacement packet to improve the voice quality. Answer: B QUESTION NO: 48 Which three methodologies are specific types of QoS used on a 512-kb/s point-to-point IP WAN link? (Choose three.) Ensure Success With Money Back Guarantee

642-437 A. header compression B. FRF.12 C. LLQ D. MultilinkPPP E. RSVP Answer: A, B, D QUESTION NO: 49 DRAG DROP Answer: Ensure Success With Money Back Guarantee

642-437 QUESTION NO: 50 Which three of the following methods are used by a Cisco Unified Border Element to provide network hiding? (Choose three.) A. back-toback user agent, replacing all H.323-embedded IP addressing B. IP network security boundary C. media flow-through D. RSVP E. IP network privacy F. intelligent IP address translation for call media and RTP flows Answer: B, E, F Reference: http://www.cisco.com/en/US/prod/collateral/voicesw/ps6790/gatecont/ps5640/pro duct_da ta_sheet09186a00801da698.html QUESTION NO: 51 DRAG DROP Ensure Success With Money Back Guarantee

642-437 Answer: DSP delay, Packetization delay, Serialization delay & Dejitter Buffer delay are Fixed delay types. Queuing and Buffering delay & Network delay are Variable Delay types. QUESTION NO: 52 Refer to the exhibit. Consider an outgoing call that is being placed in all three scenarios that are shown in the exhibit. What is the result of the call, going down the table from top to bottom? Ensure Success With Money Back Guarantee

642-437 A. success,success,success B. success, success, fail C. success, fail, success D. success, fail, fail E. fail, success, success F. fail, success, fail Answer: A Reference: http://www.cisco.com/en/US/tech/tk652/tk90/technologies_configuration_example 09186 a008019d649.shtml QUESTION NO: 53 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 10.1.30.1J24 G. default-router 10.1.30.1 Answer: C, E, G QUESTION NO: 54 Ensure Success With Money Back Guarantee

642-437 Refer to the exhibit. When an international call to 90114989531212001 is placed from extension 2001, which of the following statements is true? A. The call will fail because no incoming COR list is applied. B. The call will succeed because the incoming COR list is a superset of the outgoing COR list. C. The call will fail because the incoming COR list is not a superset of the outgoing COR list. D. The call will succeed because the incoming COR list has the highest priority, by default, when no incoming COR list is applied. Answer: D QUESTION NO: 55 In which situation would the trust boundary be located at the access layer? A. if the endpoints, both IP phones and PCs, are incapable of marking traffic properly B. if PCs are switched through an IP phone and the IP phone traffic can be trusted to mark both traffic streams properly C. if the access layer switch cannot trust or re-mark incoming traffic from endpoints properly D. if there are endpoints that cannot be trusted and connect directly to the distribution layer Answer: B QUESTION NO: 56 Ensure Success With Money Back Guarantee

642-437 An access layer switch is configured to extend priority to an IP phone. Cisco Discovery Protocol is enabled on all ports. What are the three possible ways that an IP phone can be instructed to treat the Layer 2 CoS priority value of the attached PC? (Choose three.) A. trusted IEEE 802.1Q B. configured DSCP level C. configured CoS level D. trusted E. configured IEEE 802.1Q F. untrusted Answer: A, C, D QUESTION NO: 57 Refer to the exhibit. When 5551234 is being matched with the outgoing dial peer that is shown in the exhibit, which of the following called numbers will be sent to the PSTN? A. 5551234 B. 1234 C. 555 D. Null E. 5 F. 51234 Answer: F QUESTION NO: 58 DRAG DROP Ensure Success With Money Back Guarantee

642-437 Answer: INVITE => 1. <= 100 Trying 2. <= 180 Ringing 3. <= 200 OK 4. ACK => QUESTION NO: 59 DRAG DROP Ensure Success With Money Back Guarantee

642-437 Answer: 1. Distribution of dial plan logic 2. Provide summarization 3. Simplicity in provisioning 4. Reduces post-dial delay 5. Availability and fault tolerance QUESTION NO: 60 Refer to the exhibit. 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? Ensure Success With Money Back Guarantee

642-437 A. service voice voip allow-connections h323 to h323 allow-connections h323 to sip B. voice service voip allow-connections h323 to h323 allow-connections h323 to sip C. service voice voip allow-connections h323 to h323 allowconnections h323 to sip allow-connections sip to sip allow-connections sip to h323 D. voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 Answer: D Explanation: 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: Ensure Success With Money Back Guarantee

642-437 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 doesnt use SIP so we dont need to configure allowconnections sip to sip. QUESTION NO: 61 Refer to the exhibit. Your customer has connected an existing PBX to the IP network. The PBX users can make calls to other extensions on the PBX but are unable to call the test extension 2201. All other applications on the IP network are working correctly. Compare the PBX system requirements to the configuration for R1 in the exhibit. Which configuration change will resolve the problem? A. configure operation 2-wire and type 5 on voice-port 1/1/0 B. configure operation 4-wire and type 5 on voice-port 1/1/0 C. configure forward digits all in dial-peer 1 POTS D. configure wink-start signaling on voice-port 1/1/0 Ensure Success With Money Back Guarantee

642-437 Answer: B QUESTION NO: 62 Where would you assign COR lists in Cisco Unified Communications Manager Express? A. ephone-dn B. voice register dn C. voice register pool D. ephone Answer: A Explanation: For Cisco Unified Communications Manager Express, the COR list is directly assigned to the appropriate Ethernet phone-dn (ephone-directory number) QUESTION NO: 63 Refer to the exhibit. Choose the correct configuration command set that will allow the gateway in zone BR to register with the gatekeeper at HQ so that the gateway is placed in zone BR. Ensure Success With Money Back Guarantee

642-437 Ensure Success With Money Back Guarantee

642-437 A. A B. B C. C Ensure Success With Money Back Guarantee

642-437 D. D Answer: C Explanation: Notice that the router at zone Br is functioned as both gateway and gatekeeper and it uses the IP address of 10.2.120.1 as the zone local BR. Therefore if we want the gateway in zone BR to register with the gatekeeper in the same zone we must use 10.2.120.1 in the command: h323-gateway voip id BR ipaddr 10.2.120.1 In which, BR is the zone name defined in the zone local BR command (of the gatekeeper) and the is the IP address of an interface of the gatekeeper and it should be 10.2.120.1-> B, D and E are not correct. A can be correct but it is not as clear as answer C. Notice that last command h323-gateway voip h323-id BRgw specifies the BRgw is the name of the gateway to communicate with the gatekeeper. QUESTION NO: 64 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 at either site to ten digits. B. Translate all called numbers within Site B to three digits. C. Translate all called numbers within Site A to four digits. D. Translate all called numbers leaving Site A to ten digits. Answer: D Explanation: North American Numbering Plan (NANP) is designed around a 10-digit numbering plan: Ensure Success With Money Back Guarantee

642-437 (Sometimes you will see it as NXX NXXX XXXX, which means that the first and fourth digits cant be zero or one) It consists of 3-digit area codes and 7-digit telephone. For telephone numbers that are located within an area code, the PSTN uses a seven-digit dial plan numbers. Notice that Site B uses four-digit internal numbers means we need ten digits to access site B from an outside PSTN. Therefore, if people from Site A want to call people at site B and sometimes they just press 4 digits then the administrators should translate the called numbers to ten digits before leaving Site A. QUESTION NO: 65 Which CUBE configuration will support H.323 protocol interworking and address hiding? A. A B. B C. C Ensure Success With Money Back Guarantee

642-437 D. D Answer: D Explanation: Address hiding is a security feature of the CUBE which will hide the IP address of the originating gateway. This feature is turn on by default so we dont need to set it. A and B are not correct because the command h323 interworking doesnt exist (moreover A uses media flow-around feature which will turn off the address hiding feature). C is not correct because it uses media flow-around feature too. QUESTION NO: 66 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 Answer: C Explanation: Human speech ranges from 200 Hz to 9000 Hz but Nyquist cut the sampling frequency range to 4000 Hz to save bandwidth although this cut down the quality of voice too. QUESTION NO: 67 What is the process of assigning audio amplitude to a unique digital code word? A. linear prediction B. encoding C. sampling D. quantization Answer: D Explanation: Quantization is the process of assigning a value from the voltage range based on the amplitude of each audio sample. Notice that the Voltage values are not evenly spaced. The spaces near the horizontal line are much closer than the ones at the two ends. This helps our ears distinguish common sounds more easily. Ensure Success With Money Back Guarantee

642-437 QUESTION NO: 68 Refer to the graphic for IP addresses and telephone numbers. You are working with a customer that is opening a small sales office in R2. You would like to be able to have the user in R2 be able to dial into the PBX in R1 over the IP WAN. The R1 PBX uses loop start, a two-wire operation, and DTMF dialing. Please choose the correct FXO port configuration for R1. A. voice-port 1/0/0 signal ground-start Ensure Success With Money Back Guarantee

642-437 operation 2-wire dial-type dtmf B. voice-port 1/1/1 destination 2015551212 signal ground-start operation 2wire type 1 dial-type dtmf C. voice port 1/0/0 session target ipv4:172.16.1.1 destination 2015551212 signal ground-start operation 2-wire dial-type dtmf D. voice port 1/0/0 session target ipv4:172.16.1.1 source 2015551212 signal wink-start operation 2-wire dial-type dtmf Answer: A QUESTION NO: 69 Examine the following PBX system parameters: The calling side seizes the line by going off-hook on its Elead 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 1/0/0 signal immediate-start operation 4-wire type 2 B. voice port 1/0/0 Ensure Success With Money Back Guarantee

642-437 signal delay-dial operation 4-wire type 1 C. voice port 1/0/0 signal winkstart operation 4-wire type 3 D. voice port 1/0/0 signal immediate-start operation 4-wire type 4 Answer: A QUESTION NO: 70 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. Answer: A C Explanation: SCCP is the only Cisco-proprietary VoIP protocol currently in use. The purpose of SCCP protocol is to provide a signaling protocol between the Cisco Unified Communications Manager and Cisco IP phones. Similar to MGCP, SCCP is a client/server protocol -> A & C are correct. Supervisory signals involves the detection of changes to the status of a circuit (on-hook, off-hook, ringing). Any event causes a message to be sent to a Cisco UCM -> We can say SCCP is more than a supervisory signaling protocol because it tells the phone exactly what to do. From the on-hook, off-hook, buttons pressed, lamp on/off, through the prompt, key settings, and even the dialtone -> B is not correct. The beauty of SCCP is that it makes the endpoints very cheap comparing to the H.323 devices. The end stations (telephones) that use SCCP are called Skinny clients, which consume less processing overhead and they do not contain call control intelligence -> D is not correct. QUESTION NO: 71 Ensure Success With Money Back Guarantee

642-437 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.) A. dial-peer voice 50 voip destination-pattern 50 session protocol sipv2 session target ipv4: 192.168.100.100 B. dial-peer voice 1000 voip destination-pattern 51 session protocol sipv2 session target ipv4: 192.168.100.100 C. dial-peer voice 91 voip session protocol sipv2 destination-pattern 91T session target 10.1.100.1 dtmf-relay rtp-nte digit-drop h245-alphanumeric D. dial-peer voice 91 voip Ensure Success With Money Back Guarantee

642-437 destination-pattern 91T session target 10.1.100.1 dtmf-relay rtp-nte digitdrop h245-alphanumeric E. dial-peer voice 1000 voip destination-pattern 51 session target 10.1.100.1 F. dial-peer voice 50 voip destination-pattern 50 session target ipv4: 172.16.14.6 Answer: BCF QUESTION NO: 72 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? Ensure Success With Money Back Guarantee

642-437 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. Answer: B QUESTION NO: 73 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 Answer: B Explanation By using trunk groups, we can choose to use either the even or odd channels first with the command: huntscheme . [even | odd ....] (notice: the full command is very long so I shorten it to the simplest form) Ensure Success With Money Back Guarantee

642-437 QUESTION NO: 74 Which item correctly describes the relationships between the feature and the category it belongs? 1 Supports analog faxes and modems on a VoIP network 2 Performs call setup and teardown between VoIP networks and the PSTN 3 Interconnects segments of the same or different VoIP networks using different media types Interconnects segments of the same or different VoIP network using different signaling types 4 A. Gateway 1 and 2 CUBE 3 and 4 B. Gateway 1 and 3 CUBE 2 and 4 C. Gateway 2 and 3 CUBE 1 and 4 D. Gateway 2 and 4 CUBE 1 and 3 Answer: A QUESTION NO: 75 In T1 CAS, where are the signaling states and control features carried for Super Frame robbed-bit signaling? A. 6th and 12th frame B. 6th, 12th, 18th, and 24th frame C. the first and seventeenth time slot D. the first and sixteenth time slot Answer: A Explanation Ensure Success With Money Back Guarantee

642-437 Each T1 has 24 channels ( or 24 DS0 digital signal level 0) that can transmit 8 bits per channel each. This give us a total of 192 bits. One more bit is used for framing, bringing the total to 193 bits. Super Frame bundles 12 of these 193-bit frames for transport. QUESTION NO: 76 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 Answer: D Explanation Calling privileges define the destination a user is allowed to dial and they are implemented on Cisco IOS gateway using Class of Restriction. Class of Restriction (COR) is the feature that determines which numbers might not be dialed on the system. COR is required only when you want to restrict the ability of some phones to make certain types of calls but allow other phones to place those calls. COR functionality provides the ability to deny certain call attempts on the basis of the incoming and outgoing CORs that are provisioned on the dial-peers. This functionality provides flexibility in network design, allows users to block calls (for example, calls to 900 numbers), and applies different restrictions to call attempts from different originators. (Reference: http://www.cisco.com/en/US/tech/tk652/tk90/technologies_configuration_ex ample09186a008019d649.shtml) QUESTION NO: 77 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 727-5551000 so that when the call is completed, network resources are returned for other uses? Ensure Success With Money Back Guarantee

642-437 A. PLAR B. trunk C. tie-line D. answer-mode Answer: C Explanation: E&M signaling supports tie-line type facilities. QUESTION NO: 78 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 612 is received? dial-peer voice 1 pots destie Corp. wants to use 5digit dialing nationpattern 612.. no digit-strip prefix 5501 port 1/0/0 Ensure Success With Money Back Guarantee

642-437 A. A nine digit number beginning with 5501 will be forwarded. B. A ten digit number beginning with 5501 will be forwarded. C. A twelve digit number beginning with 5501612 will be forwarded. D. A thirteen digit number beginning with 5501612 will be forwarded. Answer: C Explanation: This dial-peer has the no digit-strip command so no digits are stripped when this dial-peer is matched. So the whole number will be transferred with the format of5501612xxxxx (5501 is prefixed with the command prefix 5501) QUESTION NO: 79 Which command sets parameters to search a series of dial peers for a destination that is not in use? A. dial-peer rotary B. dial-peer circulate C. dial-peer hunt D. dial-peer distribute Answer: C Explanation Dial peer hunting is the process used when an originating router tries to establish a call on different dial peers if the originating router receives a user-busy invalid number or an unassignednumber disconnect cause code from a destination router. QUESTION NO: 80 Refer to the exhibit. Lighthorse Equine Management would like to investigate converging voice and data on their existing T1 Frame Relay WAN link between R1 and R2. Currently the following lists 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 Ensure Success With Money Back Guarantee

642-437 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 Answer: B QUESTION NO: 81 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. Ensure Success With Money Back Guarantee

642-437 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. Answer: BCE QUESTION NO: 82 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 deployed without redundancy checks? A. 1 byte B. 2 bytes C. 3 bytes D. 4 bytes E. 20 bytes F. 40 bytes Answer: B QUESTION NO: 83 Which device is used to allow an H.323 stream to transit a firewall? A. gatekeeper B. gateway C. proxy D. MCU Answer: C QUESTION NO: 84 In which three RAS messages is the technology prefix sent? (Choose three.) A. GRQ B. RRQ C. RCF D. IRR E. IRQ Ensure Success With Money Back Guarantee

642-437 Answer: A B E Explanation: The Cisco gatekeeper uses technology prefixes to group endpoints of the same type together. It uses the technology prefix appended in the called number to select the destination gateway or zone. This method prepends a technology prefix to the called number matched by the dialpeer. It is not used for registration, but for call setup with the Cisco gatekeeper. For example, called number 5551234 becomes 1#5551234. The technology prefix registration information is sent to the Cisco gatekeeper in the RAS Registration Request (RRQ) message. For example: GWYB1(config)#interface ethernet 0/0 GWY-B1(config-if)#h323-gateway voip techprefix 1# QUESTION NO: 85 Call Admission control (CAC) is a concept that applies to voice traffic only not data traffic. Which two types are of Call Admission Control? (Choose two.) A. resource-based B. gatekeeper-controlled RSVP C. local D. QoS-based Answer: A C Explanation: There are 3 types of CAC: + Local CAC + Measurement Based CAC + Resource-Based CAC For more information about these types, please read: http://www.cisco.com/en/US/docs/ios/solutions_docs/voip_solutions/CAC.html QUESTION NO: 86 DRAG DROP Assume a SIP voice network. Drag each characteristic to the type of SIP call setup the characteristics best describes. Ensure Success With Money Back Guarantee

642-437 Answer: Ensure Success With Money Back Guarantee

642-437 Direct call setup: + Nonscalable + UA must keep data on large number of destinations + Relies on cached information to resolve addresses Redirect Server Call Setup: + Server reports back to a UA with destination coordinates Proxy Server Call Setup: + Most dynamic address resolution capability + All setup messages to through server + UA incapable of establishing its own sessions QUESTION NO: 87 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. Strip the site access code and send four digits. B. Strip the site access code and prepend 1512555. C. Strip the site access code, send four digits, then prepend the access code when it reaches the Merrimack gateway. D. Translate 50124 to 5125550124. Ensure Success With Money Back Guarantee

642-437 E. Do nothing because the site access code matches the last five digits of the target number. Answer: A Explanation: The site access code (6) is just used to inform the originating gateway which gateway it needs to send traffic to. Therefore, after learning the traffic should be sent to Merrimack gateway, it trips off the site access code. Notice that the receiving gateway will receive 0124, which is enough information to ring the phone plugged into it. QUESTION NO: 88 In North America, which E&M signaling type is used most often for geographically separated equipment? A. Type I B. Type V C. Type III D. Type IV E. Type II Answer: E Explanation: This information is quoted from http://www.cisco.com/en/US/tech/tk1077/technologies_tech_note09186a0080093f6 0.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 backto-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. Ensure Success With Money Back Guarantee

642-437 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 offhook 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 offhook 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 NO: 89 Refer to the exhibit. All IP phones use SCCP. Fax machine F calls fax machine J. Which call setup signaling statement is correct? Ensure Success With Money Back Guarantee

642-437 A. Fax F signals Fax J directly. Call setup is handled by the fax machines. B. 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. C. Gateway A processes the call and signals gateway B. Gateway B processes the request. D. Gateway A signals the gatekeeper. The gatekeeper processes the call and signals gateway B. E. Gateway A signals the call agent. The call agent processes the call and signals gateway B. Answer: B QUESTION NO: 90 Refer to the exhibit. Which message ID can be used to track this call from the requesting endpoint? Ensure Success With Money Back Guarantee

642-437 A. 8204 B. 1640003356 C. 10.100.100.100 D. 49579 Answer: A QUESTION NO: 91 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? Ensure Success With Money Back Guarantee

642-437 A. dial-peer voice 1 pots destination pattern 5552[5-6].0 B. dial-peer voice 1 pots destination-pattern 555[2-5][56] C. dial-peer voice 1 pots destination-pattern 5552.[0-5]0 D. dial-peer voice 1 pots destination-pattern 5552[5-6][05]0 Answer: D Explanation: The numbers can be summaried as 5552(5 or 6)(5 or 0)0 so the destination-pattern should be written as 5552[5-6][05]0 or 5552[56][05]0 QUESTION NO: 92 Which best defines an ACD? A. a local company that provides phone capability and distribution from the phone company's central office B. a telephone system that is connected to the exchange to provide conventional voice services to several subscribers C. a telephone system that switches calls between users on local lines D. a telephone system that responds to a caller with a voice menu and helps to appropriately connect the call Answer: D QUESTION NO: 93 What does a gatekeeper do when it matches a technology prefix? Ensure Success With Money Back Guarantee

642-437 A. strips off the technology prefix and sends the matching zone prefix to the remote gatekeeper B. strips off both the technology prefix and zone prefix and forwards the remaining destination number C. strips off the zone prefix and forwards the technology prefix to the remote gatekeeper D. sends both the technology prefix and zone prefix to the remote gatekeeper Answer: D QUESTION NO: 94 Which statement is true about MGCP? A. Call completion is always shared, with some intelligence on the endpoint, some on the call agent. B. Call agents order and direct each step of call completion for the endpoints. C. Endpoints may act alone or cooperate with call agent to complete calls. D. Endpoints always take all actions to complete calls. Answer: B QUESTION NO: 95 Refer to the exhibit. Which two actions would be initiated by a UAS? (Choose two.) Ensure Success With Money Back Guarantee

642-437 A. returns a response on behalf of the user to the invitation originator B. contacts the user when a SIP invitation is received C. originates the ACK method to indicate that it has received a response to an invitation D. originates the INVITE method including a description of the session parameters E. originates the REFER method to initiate call termination Answer: A,B QUESTION NO: 96 Refer to the H.323 message in the exhibit. What is the gateway doing with the gatekeeper? Ensure Success With Money Back Guarantee

642-437 A. full registration B. registration retry C. lightweight registration D. initial registration Answer: C Explanation: For the first time the gateway registers with the gatekeeper, it uses full registration. Prior to H.323 Version 2, Cisco gateways reregistered with the gatekeeper every 30 seconds. Each registration renewal used the same process as the initial registration, even though the gateway was already registered with the gatekeeper. This behavior generated considerable overhead at the gatekeeper. So from H.323 version 2, gateways can re-register with the gatekeeper using lightweight registration (it still requires the full registration process for initial registration, but uses an abbreviated renewal procedure to update the gatekeeper and minimize overhead). An endpoints registration with a gatekeeper may have a limited life span. The gatekeeper specifies the registration duration for an endpoint by including a timeToLive field in the Registration Confirm (RCF) message. After the specified length of time, the registration is considered expired. The endpoint must periodically send a Registration Request (RRQ) having the keepAlive bit set prior to the expiration time. Such a message may include a minimum amount of information as described in H.225.0 and is known as a lightweight RRQ. In the exhibit above, we can see the keepAlive bit is set to TRUE -> this is a lightweight RRQ. QUESTION NO: 97 Which dial plan characteristic shows the most obvious improvement by dropping a number translation step? A. hierarchical design B. post-dial delay C. availability D. scalability Answer: B Explanation Post-dial delay is the time between when the last digit is dialed and the moment the phone rings at the receiving location. In the PSTN, people expect a short post dial delay and to hear ring back within seconds. The more translations, digit manipulations, and lookups that take place, the longer the post dial delay becomes. Overall network design, translation rules, and alternate paths affect post dial delay. Minimize the amount of dial peers and translations to reduce post-dial delay. Ensure Success With Money Back Guarantee

642-437 By dropping a number translation step, the post-dial delay time will be obvious improvement. QUESTION NO: 98 Refer to the exhibit. 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 high B. dial-peer voice 3 pots destination-pattern 5555110 preference 3 huntstop C. dial-peer voice 3 pots Ensure Success With Money Back Guarantee

642-437 destination-pattern 5555110 preference 2 D. dial-peer voice 3 pots destination-pattern 5555110 preference 0 E. dial-peer voice 3 pots destination-pattern 5555110 preference firstlast Answer: C Explanation: The router considers lower preferences to be better than higher preferences and the default preference is 0. Therefore, by setting the preference of Bobs dial-peer to 2 we guarantee Bob will be the last one to receive the call (while James priority is set to 1 and Drew uses the default configuration). QUESTION NO: 99 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 E. media flow-around F. SIP cause codes Answer: A,B,C Explanation: Media flow through and media flow around mode is supported on the Cisco Unified Border Element (CUBE). The CUBE is always involved in the call setup (signaling) portion of the call, but the media (RTP bearer stream) may flow through the CUBE or be routed around the platform. Media flow through must be used to support many of the features available like IP address translation and IP address hiding. Media flow around Ensure Success With Money Back Guarantee

642-437 allows the CUBE greater scalability in the number of calls that can be processed by one CUBE router. For Media flow through option, the media packets are passed through the CUBE, they will get terminated and reoriginates with CUBEs IP address and port number, so here we cannot find the original gateways ip address. This is one of the security feature in the CUBE. The default option is media flow-through. QUESTION NO: 100 DRAG DROP Drop Ensure Success With Money Back Guarantee

642-437 Answer: Explanation Here is how the call is established with RSVF-based Call Admission Control 1) The Cisco Unified Communications Manager (at the leftside) sends an H.225 setup to the Cisco UBE. 2) The Cisco UBE processes the call setup information and associates an outbound VoIP dial peer requiring an RSVP reservation. The Cisco UBE sends out an RSVP Reservation request to the remote Cisco UBE. 3) The remote Cisco UBE acknowledges the reservation and initiates the reservation for the return path, which is acknowledged by the local Cisco UBE. 4) The H.225 setup message is routed to the remote Cisco UBE, which then routes the call to the outbound VoIP dial peer pointing to Cisco Unified Communications Manager (at the right-side). 5) H.245 negotiation occurs with media flow-through enabled. 6) The call is established. QUESTION NO: 101 Which process changes an internal extension into a fully qualified external PSTN number before matching to a dial peer? A. forward digits B. number expansion C. prefix extension D. digit masking Answer: B QUESTION NO: 102 What is the best description of an MGCP endpoint? A. IP phones B. the gatekeepers in a VoIP network C. the interconnection between packet and traditional telephone networks D. any analog telephony device (PBX, switch, etc.) Ensure Success With Money Back Guarantee

642-437 Answer: C QUESTION NO: 103 Refer to the exhibit. Users are not able to complete a call from 678-555-1212 to 770-5551111. What is the correct diagnosis for the problem? A. incorrect session-target statement in router 2 B. missing no digit-strip on the voip dial peer in router 1 C. incorrect destination-pattern in router 1 D. incorrect port statement in router 1 pots dial peer E. incorrect POTS dial-peer statement in router 2 Answer: C Explanation The dial-peer 2 voip in Router 1 was configured destinationpattern 770555... Notice that there are only two dots (.) in the destination-pattern that means when the user presses 77055511, the voip dialpeer is matched immediately without waiting for two last 11 pressed. Therefore router R2 only receives the 777055511 number and it doesnt match with the destination-pattern in the dial-peer voice 1 pots configured in router R2. Ensure Success With Money Back Guarantee

642-437 QUESTION NO: 104 DRAG DROP Drop Answer: QUESTION NO: 105 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 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. B. The gatekeeper always forwards the call to the zone specified in the hopoff command. Ensure Success With Money Back Guarantee

642-437 C. 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. D. The gatekeeper only forwards the call to the hopoff zone if the zone prefix does not match. Answer: B QUESTION NO: 106 Refer to the exhibit. 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 A. The second call leg is from gateway A to phone G. B. The first call leg terminates at gateway B. The second call leg is from gateway B to phone G. C. The voice packets travel directly from phone to phone. D. The first call leg terminates at gateway A. 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. E. The voice packets are routed through the gatekeeper. F. The voice packets are routed through the call agent. Ensure Success With Money Back Guarantee

642-437 Answer: C QUESTION NO: 107 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? A. Phone D signals the call agent. The call agent processes the call and signals phone G. B. Phone D signals phone G directly. Call setup is handled by the phones. C. Phone D signals gateway B, which processes the call and signals phone G. D. Phone D signals gateway A, which processes the call and signals phone G. E. Phone D signals gatekeeper. The gatekeeper processes the call and signals phone G. Answer: A Explanation: This is a weird and wrong question. Maybe the phone they want to ask here is Phone A, B or C because only these phones can use SCCP to communicate with the Call Agent. Phones D and E cant use SCCP to talk with a H.323 Gateway. Ensure Success With Money Back Guarantee

642-437 Phone A, B or C are SCCP Phones so they hand over the call control intelligence to the Call Agent and the Call Agent need to process the call and signals phone G before these phones can talk with each other. QUESTION NO: 108 Using Cisco Unified Communications Manager Express, what four steps are necessary to implement COR? (Choose four.) A. Define COR labels. B. Configure dial peers and assign COR lists. C. Configure COR lists on voice ports. D. Configure SRST. E. Configure COR lists. F. Assign COR list to ephone-DN. Answer: A,B,E,F Explanation: Four steps to configure COR on Cisco IOS gateway using Cisco Unified Communications Manager Express: 1) Define COR labels. 2) Configure COR lists. 3) Configure dial peers and assign COR lists. 4) Assign COR lists to ephone-dn. For example, we will define three calling privilege classes: Local: This class should allow emergency and local calls. Long Distance: This class should allow emergency, local, and long distance calls. International: This class should allow emergency, local, long distance, and international calls. Step 1: Define the four COR labels to be used as COR list members with the command dial-peer cor custom. Router(config)#dial-peer cor custom Router(config-dp-cor)#name 911 Router(config-dp-cor)#name local Router(config-dp-cor)#name ld Router(configdp-cor)#name intl Ensure Success With Money Back Guarantee

642-437 Description 911: Allows calls to emergency 911 local: Allows local calls only ld: Allows long distance calls intl: Allows international calls Step 2: Define the COR lists that will be assigned as outgoing to the PSTN dial peers with the command dial-peer cor list. Router(config-dpcorlist)#dial-peer cor list 911call Router(config-dp-corlist)#member 911 Router(config-dp-corlist)#dial-peer cor list localcall Router(config-dpcorlist)#member local Router(config-dp-corlist)#dial-peer cor list ldcall Router(config-dp-corlist)#member ld Router(config-dp-corlist)#dial-peer cor list intlcall Router(config-dp-corlist)#member intl Define the COR lists that will be assigned as incoming from the local dial peers with the command dial-peer cor list . Router(config)#dial-peer cor list local Router(configdp-corlist)#member 911 Router(config-dp-corlist)#member local Router(config)#dial-peer cor list ld Router(config-dp-corlist)#member 911 Router(config-dp-corlist)#member local Router(config-dp-corlist)#member ld Router(config)#dial-peer cor list intl Router(config-dp-corlist)#member 911 Router(config-dp-corlist)#member local Router(config-dp-corlist)#member ld Router(config-dp-corlist)#member intl Step 4: Assign Outbound COR Lists to PSTN Dial Peers Dial peer 911 has the outgoing 911call COR list Dial peer 9911 has the outgoing 911call COR list. Dial peer 9 has the outgoing localcall COR list. Dial peer 91 has the outgoing ldcall COR list. Dial peer 9011 has the outgoing intlcall COR list. Router(config)#dial-peer voice 911 pots Router(config-dial-peer)#destinationpattern 911 Router(config-dial-peer)#forward-digits all Router(config-dialpeer)#corlist outgoing 911call Router(config-dial-peer)#port 0/0/0:23 Ensure Success With Money Back Guarantee

642-437 Router(config)#dial-peer voice 9911 pots Router(config-dialpeer)#destination-pattern 9911 Router(config-dial-peer)#forward-digits 3 Router(config-dial-peer)#corlist outgoing 911call Router(config-dialpeer)#port 0/0/0:23 Router(config)#dial-peer voice 9 pots Router(config-dialpeer)#destination-pattern 9[2-9] Router(config-dial-peer)#corlist outgoing localcall Router(config-dial-peer)#port 0/0/0:23 Router(config)#dial-peer voice 91 pots Router(config-dial-peer)#destination-pattern 91[2-9]..[2-9] Router(config-dial-peer)#prefix 1 Router(config-dial-peer)#corlist outgoing ldcall Router(config-dial-peer)#port 0/0/0:23 Router(config)#dial-peer voice 9011 pots Router(config-dial-peer)#destination-pattern 9011T Router(configdial-peer)#prefix 011 Router(config-dial-peer)#corlist outgoing intlcall Router(config-dial-peer)#port 0/0/0:23 Reference: CVoice Student Guide v6.0 (Page 4-165) QUESTION NO: 109 Refer to the exhibit. Your customers dial in to your company using a local number, and their calls cross the WAN to an IVR system. They are complaining that the IVR system does not always accept their input or may get it wrong. The IVR system has been checked and is working properly. What needs to be added to the dial peer on the incoming H.323 gateway to correct this problem? A. codec g729ar8 bytes 30 B. dtmf-relay h245-alphanumeric C. tech-prefix 1# D. no vad Ensure Success With Money Back Guarantee

642-437 Answer: B Explanation DTMF is the tone generated when you press a button on a touchtone phone. This tone is compressed at one end of a call; when the tone is decompressed at the other end, it can become distorted, depending on the codec used. The DTMF relay feature transports DTMF tones generated after call establishment out-of-band by using either a standard H.323 out-of-band method or a proprietary RTP-based mechanism. For session initiation protocol (SIP) calls, the most appropriate method to transport DTMF tones is Real-Time Transport Protocol named telephony event (RTP-NTE) or session initiation protocol notify (SIP Notify). When you press a button on the touch-tone phone, a high group frequency is combined with a low group frequency and you can hear a generated tone. Notice that you often dont see the A B C D column in most modern DTMF phones nowadays. Although DTMF is usually transported accurately when using high-bit-rate voice codecs such as G.711, low-bit-rate codecs such as G.729 and G.723.1 are highly optimized for voice patterns and tend to distort DTMF tones. As a result, interactive voice response (IVR) systems may not correctly recognize the tones. Therefore the IVR sometimes can not recognize the DTMF tones and doesnt accept their input o may get it wrong. The main advantage of the dtmf-relay command is it sends DTMF tones with greater fidelity than is possible in-band for most lowbandwidth codecs, such as G.729 and G.723. (Reference: CVoice Student Guide v6.0) QUESTION NO: 110 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. rotary B. circulate C. distribute D. request E. hunt F. query Answer: E QUESTION NO: 111 Which option is true concerning the MGCP call agent? Ensure Success With Money Back Guarantee

642-437 A. provides only call signaling and call setup B. manages all aspects of the call and voice stream C. monitors the quality of each call after setup D. acts only as a recorder of call details Answer: A Explanation: 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 NO: 112 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 1-31 type r2-digital r2compelled ani B. ds0-group 0 timeslots 1-24 type none C. pri-group timeslots 1-24 D. ds0-group 0 timeslots 1-24 type e&m-fgd E. ds0-group 0 timeslots 1-24 type fgd-eana Answer: D Explanation Ensure Success With Money Back Guarantee

642-437 To define T1 channels for compressed voice calls and the channel-associated signaling (CAS) method by which the router connects to the PBX or PSTN, enter the ds0-group controller configuration command. Below is the syntax of this command: ds0-group ds0-group-no timeslots timeslot-list type {e&m-immediate | e&m-delay | e&m-wink | e&m-fgd | fgd-eana} Description ds0-groupA value from 0 to 23 that identifies the DS0 group no timeslot-list is a single timeslot number, a single range of numbers, or multiple ranges of numbers separated by commas. For T1, allowable values are from 1 to 24. Examples are: timeslotlist 2 1-15, 17-24 1-23 2, 4, 6-12 The signaling method selection for type depends on the connection that you are making. The E&M interface allows connection for PBX trunk lines (tie lines) and telephone equipment. The options are as follows: e&m-immediate-start specifies no specific offhook and onhook signaling. e&m-delay specifies that the originating endpoint sends an offhook signal and then and waits for an offhook signal followed by an onhook signal from the destination. e&mfgd specifies E&M Type II Feature Group D. e&m-wink-start specifies that the originating endpoint sends an offhook signal and waits for a wink signal from the destination. fgd-eana specifies Group D Exchange Access North American (EANA) signaling. (There are some more options but they are omitted) (Reference: http://www.ciscosecure.net/en/US/docs/ios/12_1/12_1xd/feature/guide/hdv_f gd.html) T1 CAS always provides the ANI/DNIS delimiter on incoming T1/CAS trunk lines. Notice: type Ensure Success With Money Back Guarantee

642-437 + CAS signaling main feature is its use of user bandwidth to perform signaling functions. CAS signaling is often referred to as robbed-bitsignaling because user bandwidth is being robbed by the network for other purposes. + E&M signaling is typically used for trunks. It is normally the only way that a central office (CO) switch can provide two-way dialing with direct inward dialing. + ANI Automatic number identification. SS7 (signaling system 7) feature in which a series of digits, either analog or digital, are included in the call, identifying the telephone number of the calling device. In other words, ANI identifies the number of the calling party. + DNIS Dialed number identification service, also known as the called party number. The telephone number of the called party after translation occurs in the Public Switched Telephone Network. A given destination may have a different DNIS number based on how the call is placed (for example, 800 or direct dial). QUESTION NO: 113 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? Ensure Success With Money Back Guarantee

642-437 A. The Engineering manager and Marketing manager are on the phone. B. The Shipping manager and Marketing manager are on the phone. C. None of the managers are on the phone. D. The Engineering manager is on the phone. E. The Marketing manager is on the phone. Answer: A Explanation: With the preference 0 configured in dial-peer voice 1 pots, this dial-peer (Marketing) has the highest priority to receive call if it is idle. Dial-peer 2 (Engineering) has the next priority and dial-peer 3 (Shipping) has lowest priority so it only rings when both Marketing and Engineering phones are busy. It is a bit weird but the router considers lower preferences to be better than higher preferences. One more notice is that the default preference for a dial peer is 0. QUESTION NO: 114 Refer to the exhibit. Which dial peer configuration will block phone A from making long distance calls? Ensure Success With Money Back Guarantee

642-437 Ensure Success With Money Back Guarantee

642-437 Ensure Success With Money Back Guarantee

642-437 A. A B. B C. C D. D E. E Answer: E Explanation To block phone A from making long distance calls, phone A must belong to an incoming dial-peer which is not a member of the LD (Long Distance). In three incoming dial-peer (the three last dial-peers), there is only one dial-peer satisfies with this condition, that is the LocalLst dial-peer so the answer should be E. One tip to quickly recognizes which dial-peer is for outgoing dial-peer is that this type of dial-peer usually have only one member. In this question, the outgoing dial-peers are Em01, Local01, LD01, Intl01. Dial-peers which have more than one member are often incoming dial-peers. You can read http://www.cisco.com/en/US/tech/tk652/tk90/technologies_configuration_example 0 9186a008019d649.shtml for another example. QUESTION NO: 115 Using a standalone IOS gateway, which three steps necessary to implement COR? (Choose three.) A. Configure COR lists ports. B. Configure dial peers and assign COR lists. C. Define COR Configure COR lists. E. Assign COR list to ephone-DN. F. Configure Answer: B,C,D Ensure Success With Money Back Guarantee are on voice labels. D. SRST.

642-437 QUESTION NO: 116 Refer to the exhibit. When extension 201-555-1000 dials 404555-1200, how are the digits manipulated in R1 so that they are presented correctly at R2? A. 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. B. The digits 404-555 are stripped off by the connection trunk and R2 receives only 1200. C. The outbound VoIP dial peer is matched and all digits are sent. D. The digits 404-555 are stripped off before matching the outbound POTS dial peer. Answer: C Explanation: Ensure Success With Money Back Guarantee

642-437 When 201-555-1000 (Phone A) calls 404-555-1200 (Phone B) the dial-peer voice 1 voip at R1 is matched with the destination-pattern 404555. But notice that this is a voip dial-peer so digits are not stripped and all digits are sent to R2. QUESTION NO: 117 Which three are supervisory signals? (Choose three.) A. on hook B. busy C. off hook D. ring E. call waiting Answer: A,C,D Explanation: Supervisory signals involves the detection of changes to the status of a circuit. In other words, supervisory signaling is used to indicate the state of a circuit. Once these changes are detected, the supervisory circuit generates a predetermined response. There are three different types of supervisory signals, which are: 1. On-hook 2. Off-hook 3. Ring When a telephone handset is in the cradle, the circuit is said to be on-hook. In onhook state, the circuit is said to be open, thus preventing the current from flowing through the telephone. When the telephone handset is removed from the cradle, the circuit transitions to an offhook state and there is a current flowing through the electrical loop. When the telephone network senses the off-hook state via the current flow, it provides a signal in the form of dial-tone that it is ready to accept the call. When making a call, the caller receives a ringback tone from the telephone switch, which alerts the caller that the telephone switch is sending ringing voltage to the called party. It is important to know that only the ringing that the recipient (the called party) hears is the supervisory signal; the ringback tone that the caller hears is simply a call-progress indicator and is not a supervisory signal. Ensure Success With Money Back Guarantee

642-437 QUESTION NO: 118 What is the most common E&M type used outside North America? A. Type II B. Type I C. Type IV D. Type V E. Type III Answer: D QUESTION NO: 119 To hide its identity when initiating calls, Phone B requests that Server B place its calls for it. What kind of device is Server B? A. proxy B. user agent server C. redirect D. registrar E. user agent client Ensure Success With Money Back Guarantee

642-437 Answer: A QUESTION NO: 120 The D channel in ISDN is an example of which two signaling methods? (Choose two.) A. CCS B. out-of-band C. CAS D. in-band E. gateway Answer: A,B Explanation: There are two types of ISDN lines: Basic Rate ISDN (BRI) and Primary Rate ISDN (PRI). Both BRI and PRI types have the same 64kbps D channel that is used for call supervision. This D channel is dedicated for signaling only and contains all the necessary signaling for establishing call between two end-points so it is a kind of CCS signaling and out-of-band signaling. QUESTION NO: 121 Refer to the exhibit. 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.) Ensure Success With Money Back Guarantee

642-437 A. replace zone remote GK-B acme.com 172.16.14.99 1719 with zone local GK-B acme.com 172.16.14.99 1719 B. replace zone local GK-A acme.com 172.16.14.44 1719 with zone remote GK-A acme.com 172.16.14.44 1719 C. replace zone prefix GK-A 770 . . . . . . . Ensure Success With Money Back Guarantee

642-437 with zone prefix GK-A 770 . . . . D. replace zone prefix GK-B 404 . . . . with zone prefix GK-B 404 . . . . . . . E. replace zone local DGK acme.com with zone remote DGK acme.com Answer: B,C QUESTION NO: 122 Which statement is true about only out-of-band signaling? A. All voice packets carry their own signaling. B. A signaling bit is robbed from each frame. C. Signaling bits are sent in a special order in a dedicated signaling frame. D. All signaling is directly associated with its corresponding voice frame. Answer: C Explanation: 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. QUESTION NO: 123 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? Ensure Success With Money Back Guarantee

642-437 A. connection trunk answer-mode 408555 B. connection trunk 408555.... C. ds0group timeslots 1-23 type ext-sig D. voice-port 1/0:1 E. connection-trunk 404555.... answer-mode Answer: E QUESTION NO: 124 Which two statements describe the purpose of the technology prefix? (Choose two.) A. Technology prefixes are configured on gateways to indicate to the gatekeeper whether they support voice or video. B. Technology prefixes must always be configured on gateways. C. Technology prefixes are used to identify different types or classes of gateways. D. Technology prefixes have to be unique on each gateway. E. Technology prefixes are prepended to the destination address by the gateway. Answer: C,E QUESTION NO: 125 Refer to the exhibit. Which configuration option will allow communication between a voice-enabled router and a PBX? Ensure Success With Money Back Guarantee

642-437 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 D. voice port 1/0/0 signaling wink-start operation 4-wire type 4 Answer: A QUESTION NO: 126 When using CUBE, which two statements describe how media flow-through differs from media flow-around? (Choose two.) A. Media flowaround terminates the signaling stream and allows RTP streams to flow directly between endpoints. B. Media flow-through provides address hiding by terminating both signaling and RTP streams. C. Media flow-through terminates the RTP streams but allows signaling to flow directly between endpoints. Ensure Success With Money Back Guarantee

642-437 D. Media flow-around and media flow-through function in a similar manner, but media flow-around supports NAT traversal. E. Media flow-through terminates the signaling channel and the RTP streams flow directly between endpoints. F. Media flow-around provides address hiding by terminating both signaling and RTP streams. Answer: A,B Explanation: Media flow through and media flow around mode is supported on the Cisco Unified Border Element (CUBE). The CUBE is always involved in the call setup (signaling) portion of the call, but the media (RTP bearer stream) may flow through the CUBE or be routed around the platform. Media flow around allows the CUBE greater scalability in the number of calls that can be processed by one CUBE router. For Media flow through option, the media packets are passed through the CUBE, they will get terminated and re-originates with CUBEs IP address and port number, so here we cannot find the original gateways ip address. This is one of the security feature in the CUBE. The default option is media flow-through. QUESTION NO: 127 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.729B B. G.726 C. G.711 D. G.723.1 Answer: A QUESTION NO: 128 Ensure Success With Money Back Guarantee

642-437 Refer to the IOS configuration in the exhibit. How will the next incoming call be routed? A. The call will be routed to the least used channel. B. The call will be routed to the longest idle 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 24. E. The call will be routed to the next available channel, starting from channel 24, hunting down toward channel 1. Answer: E Explanation In the configuration, we learn that the hunt-scheme sequential is used. It specifies the sequential search method for finding an available channel in a trunk group for outgoing calls. The syntax of this command is shown below: hunt-scheme sequential [both | even | odd [up | down] ] Description: + both: Searches both even and odd numbered channels. + even: Searches for an idle even numbered channel. If no idle even numbered channel is available, an odd-numbered channel is sought. + odd: Searches for an idle odd numbered channel. If no idle odd numbered channel is available, an even-numbered channel is sought. + up: Searches channels in ascending order based within a trunk group member. + down: Searches channels in descending order within a trunk group member. Ensure Success With Money Back Guarantee

642-437 Notice that up & down parameters are used with both, even or odd. Therefore the command hunt-scheme sequential even up searches in ascending order for an even numbered idle channel starting with the trunk group member of highest precedence. I am not so sure but channel 24 will have highest precedence so the hunt begins from channel 24 down to channel 1. Therefore, E is the most suitable solution for this question. The cas-custom command is used to customize T1/CAS signaling parameters for a particular T1 channel group on a channelized T1 line. QUESTION NO: 129 Refer to the exhibit. What is the minimum WAN bandwidth required to support three simultaneous VoIP calls in this network? A. 79,200 bps B. 19,200 bps C. 247,200 bps D. 51,600 bps Answer: A QUESTION NO: 130 What is the E.164 standard? A. private numbering plan Ensure Success With Money Back Guarantee

642-437 B. national numbering plan C. international public telecommunications numbering plan D. dial plan Answer: C Explanation: E.164 is an ITU-T recommendation which defines the international public telecommunication numbering plan used in the PSTN and some other data networks. QUESTION NO: 131 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 T1 PRI D. ISDN E1 PRI Answer: D Explanation: The ISDN E1 PRI has 32 timeslots (channels). Each timeslot is 8 bits and has a data rate of 64,000 bits/second. Timeslot 0 is used for frame synchronization and alarms. Timeslot 16 is used for signaling so we can use 30 timeslots to carry calls. ISDN T1 PRI only has 24 timeslots and can not support 30 simultaneous calls. QSIG is just an ISDN based signaling protocol for signaling. QUESTION NO: 132 DRAG DROP Drop Ensure Success With Money Back Guarantee

642-437 Answer: 1) T1 or E1 with CAS or PRI: PBX to PBX 2) FXO: off-net 3) FXS: local 4) FXS or switch: on-net 5) E&M, FXO, FXS: PLAR Explanation First lets have a quick review of these types of calls: Local calls are calls that occur when both the calling and called phones are attached to the same router. On-net calls are calls that need more than one router. For example, the calling phone is from one router and the called phone attaches to another router. But notice that these routers are part of the same network. Off-net calls are calls that originate on a router but terminate on the PSTN. PLAR calls occur when a caller picks up a phone and the phone automatically dials a preconfigured number. PBX to PBX calls are on-net calls, where the source and destination are PBXs. Next we will explain the answers above: PBX to PBX connections can use T1 or E1 with CAS or PRI: Nowadays, we often make PBX connections to a network through T1 or E1 lines with channel associated signaling (CAS) or Primary Rate Interface (PRI) signaling. For off-net calls, the typical connection between the router and the PSTN is through FXO port. Ensure Success With Money Back Guarantee

642-437 A local call just need FXS ports so it is the only choice for this type of call. We can make on-net calls through FXS port (phone directly connected to the router) or FXO port (phone connected to a PBX). The switch here means that we can connect an IP phone through a switch and place on-net calls through Cisco Unified Communications Manager. A PLAR call can work with any type of signaling, including E&M, FXO, FXS interfaces. QUESTION NO: 133 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 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. gateway Ensure Success With Money Back Guarantee

642-437 B. h323-gateway voip bind srcaddr 172.16.2.2 C. h323-gateway voip h323-id GK1 D. h323-gateway voip GW1-zone2.abc.com abc.com ipaddr 172.16.2.1 Answer: A Explanation: The gateway command enables the H.323 VoIP gateway to register with the gatekeeper. This is the first command you should enter when configuring a voice gateway. QUESTION NO: 134 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. trunk group B. hunt group C. source IP group D. voice port E. dial peer Answer: A QUESTION NO: 135 Which two are types of Call Admission Control? (Choose two.) A. local B. gateway zone bandwidth C. resource-based D. gatekeeper-controlled RSVP E. QoS-based Ensure Success With Money Back Guarantee

642-437 F. topology-based Answer: A,C QUESTION exhibit. You want needs to NO: 136 Refer to the output from the debug h225 asn1 command in the You have configured a gatekeeper with two local zones, hq and br. the gateway at the branch location to register with zone BR. What be corrected in the branch gateway to resolve the issue?

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642-437 A. Change the gatekeeper-id in the h323-gateway voip id command. Ensure Success With Money Back Guarantee

642-437 B. Change the IP address in the h323-gateway voip id command. C. Change the gatekeeper-id and the IP address in the h323-gateway voip id command. D. Add a zone remote for zone BR so the gateway can register with the correct zone. Answer: A QUESTION NO: 137 Refer to the exhibit. Select the recommended QoS configuration for the LAN segments of the network in the campus and branch office. A. Configure WRR with voice as highest priority. Use ACLs to classify voice traffic. Isolate voice traffic in its own VLAN . Configure access switches to trust traffic from IP phones. B. Configure a PQ with WRR. Use ACLs to classify voice control traffic. Isolate voice traffic in its own VLAN. Configure access switches to trust traffic from IP phones. C. Configure a PQ with voice control as highest priority. Use ACLs to classify voice control traffic. Isolate voice control traffic in its own VLAN . Configure access switches to trust voice control traffic from IP phones. D. Configure a PQ with WRR. Ensure Success With Money Back Guarantee

642-437 Use ACLs to classify voice traffic. Isolate voice traffic in its own VLAN. Configure access switches to trust traffic from IP phones. Answer: D QUESTION NO: 138 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 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 translationrule 1 910 The replaced number: 21555021910 C. test translation-rule 1 512 The replaced number: 21555021512 D. test translation-rule 1 617 The replaced number: 61721555021 Ensure Success With Money Back Guarantee

642-437 Answer: C QUESTION NO: 139 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 IPIP 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. The h323-gateway voip id command has an incorrect gatekeeper ID and IP address. Ensure Success With Money Back Guarantee

642-437 C. You need to add a VoIP dial peer to the configuration. D. The h323-gateway voip id command has an incorrect IP address. Answer: C QUESTION NO: 140 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 Answer: C Explanation: First, the gateway attempts to match the called number with the incoming callednumber. If no match is found, the router or gateway attempts to match the calling number of the call set-up request with the answer-address of each dial-peers. If no match is found, it attempts to match the calling number of the call set-up request to the destination-pattern of each dial-peer. Notice that these steps are just applied for inbound dial peer. QUESTION NO: 141 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? Ensure Success With Money Back Guarantee

642-437 A. A B. B C. C D. D Answer: D Ensure Success With Money Back Guarantee

642-437 QUESTION NO: 142 Which two codes together make up the number that follows the E.164 recommendation numbering scheme? (Choose two.) A. national destination code B. country code C. subscriber code D. provider code Answer: B,C Explanation: 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: 1-602555-1212 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 Ensure Success With Money Back Guarantee

642-437 QUESTION NO: 143 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 SJ2 B. zone prefix SJ 408 gw-priority 6 SJ1, 10 SJ2 C. zone prefix SJ 408 gw-priority 0 SJ2, 10 SJ1 D. zone prefix SJ 408 gw-priority 6 SJ1 E. zone prefix SJ 408 gw-priority 10 SJ2 F. zone prefix SJ 408 gwpriority 10 SJ1 Answer: D,E Explanation The simple syntax of zone prefix command is zone prefix gatekeeper-name e164-prefix [gw-priority priority gw-alias,...] For example, the command zone prefix SJ 408 gw-priority 6 SJ1 SJ is the gatekeeper-name, 408 is the E164-prefix area code, 6 is the priority and SJ1 is the GW-alias The [gw-priority priority gw-alias,...] part defines how the gatekeeper selects gateways in its local zone for calls to numbers beginning with prefix e164-prefix. The priority ranges from 0 to 10, where 0 prevents the gatekeeper from using the gateway gw-alias for that prefix and 10 places the highest priority on gateway gw-alias. The default is 5. By assigning SJ2 a priority value higher than that of SJ1, SJ2 will be the first choice when making call to this zone. QUESTION NO: 144 Refer to the exhibit. Which protocol provides the necessary sequence numbers so that voice packets originating at R1 are played in the correct order at R5? Ensure Success With Money Back Guarantee

642-437 A. UDP B. RTCP C. LFI D. CRTP E. RTP Answer: E QUESTION NO: 145 At what point does the MGCP call agent release the setup of the call path to the residential gateways? A. does not release call path setup B. after the call agent has been notified of an event and has instructed the source residential gateway to create a connection C. after the call agent has been notified that an event occurred at the source residential gateway Ensure Success With Money Back Guarantee

642-437 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 Answer: D Ensure Success With Money Back Guarantee