Question No: 1

Cisco 640-460: Practice Exam QUESTION NO: 1 Which type of voice port will allow the gateway to terminate
23 or 30 circuits from the PSTN or a PBX? A. FXO B. FXS C. PRI T1/E1 D. E&M E. BRI Answer: C
QUESTION NO: 2 In which of the following scenarios would a Cisco switch supply PoE to an IP phone? A. In any cases as long as the switch and IP phone are Cisco products. B. If the Cisco switch supportsprestandard PoE and the IP phone supports 802.3af. C. If the Cisco switch and IP phone both support a commonPoE method. D. Only if both the switch and IP phone use 802.3afPoE. Answer: C
QUESTION NO: 3 The SBB Company is setting up Call Transfer for its Cisco Unified Communications Manager Express solution. The company uses five-digit extensions and would like to be able to transfer calls outside the network to the CEO's home. The CEO's telephone number is 866-555-2222. Which configuration command will allow this to occur?
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Cisco 640-460: Practice Exam
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A. A B. B C. C D. D E. E F. F Answer: A
QUESTION NO: 4 Refer to the exhibit.
Which Cisco Unified Communications layer is the call processing layer? A. A B. B C. C D. D Answer: C
Which two statements about SIP trunk are true? (Choose two.)
Cisco 640-460: Practice Exam A. A SIP trunk configuration is always needed for a UC500 device. B. A SIP trunk is needed only to provide internet access for your data users. C. SIP trunk configuration parameters should be provided to your service provider. D. A SIP trunk is needed only if you are using voice mail to supply the Message Waiting Indicator value to the Cisco Unity Express module. E. A SIP trunk is needed only for voice if you are planning on using VoIP through a service provider. F. A SIP trunk is not supported in akeyswitch configuration. Answer: E,F
QUESTION NO: 6 A customer wants to use a signaling protocol on the voice gateways that require registration with the Cisco Unified Communications Manager. What protocol should be recommended? A. SIP B. H.323 C. SRTP D. MGCP Answer: D
QUESTION NO: 7 DRAG DROP
Answer:
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After deploying a UC500 system, you receive a support call from a user reporting that callers are going straight to the operator instead of going to members of the hunt group. Which two tabs haveconfiguration parameters that are most likely going to resolve this issue? (Choose two.) A. Device B. System C. Network D. SIP Trunk E. Voice Features F. Dial Plan
Cisco 640-460: Practice Exam Answer: E,F
QUESTION NO: 9 What is the difference between voice VLAN and native VLAN? A. Voice VLAN uses tagged 802.1Q frames whereas native VLAN uses 802.1P frames. B. Voice VLAN uses untagged frames whereas native VLAN uses 802.1Q frames. C. Voice VLAN uses tagged 802.1Q frames whereas native VLAN uses untagged frames. D. Voice VLAN uses untagged frames only when no PCs are connected behind the phones whereas native VLAN always uses untagged frames. Answer: C
QUESTION NO: 10 Which protocol is used to inform the IP phone of its voice VLAN ID? A. Ciscokeepalives B. Cisco Discovery Protocol C. Cisco Spanning Tree Protocol D. Cisco VLAN Discovery Protocol Answer: B
QUESTION NO: 11 Which value does an administrator assign to option 150 for DHCP to operate correctly in a Cisco Unified Communications Manager Express environment? A. IP address of the DNS server B. IP address of the TFTP server C. MAC address of the TFTP server D. MAC address of the DHCP server E. MAC address of the Cisco Unified Communications Manager Express router F. IP address of the PC on which the Cisco Unity Express module is installed Answer: B
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Cisco 640-460: Practice Exam Refer to the exhibit.
Which identifies the amplitude of an analog signal stream? A. A B. B C. C D. A x C E. Voltage/Time F. Voltage/Time x B/C Answer: C
QUESTION NO: 13 Which two are considered endpoints in a Cisco Unified Communications solution? (Choose two.) A. Gateway B. Call agent C. IP telephone D. Analog phone E. H.323 gatekeeper F. Cisco Unified Communications Manager Answer: A,C
Cisco 640-460: Practice Exam QUESTION NO: 14 Refer to the exhibit.
What information can be gleaned from the output of the show ephone command? A. There are two registered IP phones.Shared number 5010 on line 2. Message waiting on line 1 of phone 1. B. There are two registered IP phones.Shared number 5010 on line 2. Message waiting on shared line. C. There are two IP phones. Phone 2 is unregistered.Shared number 5010 on line 2. Message waiting on shared line. D. There are two IP phones. Phone 2 is unregistered.Shared number 5010 on line 2. Message waiting on line 1 of phone 1. E. There are two IP phones. Phone 2 is unregistered.Shared number 5010 on line 2 of phone 2. Message waiting on line 2 of phone 1. Answer: D
Answer:
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Explanation:
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QUESTION NO: 16 Which type of delay can lead to jitter in a voice network? A. Propagation delay B. Serialization delay C. CODEC delay D. Queuing delay Answer: D
QUESTION NO: 17 IP phone A places a call to IP phone B. How many RTP streams are required for the call to be successfully completed? A. 1 "Pass Any Exam. Any Time." - www.itexamworld.com 11
Cisco 640-460: Practice Exam B. 2 C. 4 D. 6 Answer: B
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Answer:
Explanation:
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Cisco 640-460: Practice Exam QUESTION NO: 19 Which is the best description of time-division multiplexing? A. A channel is assigned an exclusive slice of the overall frequency of the circuit for the entire time of its operation. B. All sources get an interleaved slice of time, which offers the entire frequency range allocated for that timeslot. C. Individual source signals are combined into a composite signal, which allows a capacity equal to or greater than the sum of the component signals. D. Technology that increases the transmission capabilities by dividing the medium into multiple channels that are each assigned a wavelength based on statistical analysis. E. On a T1 circuit, this is the process where 24 DS-1signals are multiplexed into a single DS-0 channel, while on a T3 circuit 24 DS-0 signals are multiplexed into a single DS-3 signal. Answer: B
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Which statement about the Cisco Unified Communications Manager Express configuration is correct? A. MoH will not work because you need to add the route 10.3.130.1 command in multicast moh 239.1.1.1 port 2000. B. MoH will not work because the multicast address is not correct. C. MoH will work as is. Nothing needs to be added. D. The configuration will work only for calls that are being transferred because thetransfersystem command has been added. Answer: C
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Cisco 640-460: Practice Exam QUESTION NO: 21 Refer to the exhibit.
The Acme Corporation needs assistance in configuring their PSTN voice gateway. Which two dial peers will correctly route calls to emergency services? (Choose two.) A. dial-peer voice 1 pots destination-pattern 9911 port 1/0/0 B. dial-peer voice 911 pots destination-pattern 911 forward-digits 3 port 1/0/1 C. dial-peer voice 9911 pots destination-pattern 9911 forward-digits all port 1/0/0 D. dial-peer voice 2 pots destination-pattern 911 forward-digits 3 port 1/0/1 E. dial-peer voice 1 pots destination-pattern 9911 prefix 911 port 1/0/0 F. dial-peer voice 2 pots destination-pattern 911 forward-digits all port 1/0/0 Answer: E,F
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QUESTION NO: 22 Which three characteristics apply to ISDN PRI? (Choose three.) A. The D channel is 16 kbps B. The D channel is 64 kbps C. Can carry data, voice, or video D. Cannot support callforwarding E. Commonly used only inEurope F. Can carry vendor-specific PBX features Answer: B,C,F
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You have been asked to verify the network configuration that was performed by a colleague. Which of the following conclusions would be accurate? A. The voice VLAN should always start with VLAN 1. B. The subnet mask for the network address should be class A instead of class C. C. The CME IP address should be in a different subnet to that of the IP phones. D. The CME IP address should be in the exclusion range. Answer: D
QUESTION NO: 24 Approximately what percentage of voice packets can be dropped before voice quality becomes poor?
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Cisco 640-460: Practice Exam A. 1 ?2% B. 15% C. 5 ?10% D. Less than or equal to 1% Answer: D
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Cisco 640-460: Practice Exam Explanation:
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Which step of digitizing analog signals does this represent? A. Encoding B. Quantization C. Signal sample D. Signal compression Answer: A
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Cisco 640-460: Practice Exam Explanation: There are many definitions about voice encoding but I would like to refer it as the process of translating digital numbers into binary values . It makes us remember the concept more easily. Each time we see a binary-conversion process, we can say it is the encoding step ^^.
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What is the purpose of access code 9? A. This access code connects a caller to the operator when they press 9 on their phone keypad. B. This is a level 1 security to ensure that only users that know the access code can make calls. C. This is the access code that internal users must use to make PSTN calls. D. This access code is required when interfacing with a traditional PBX system. Answer: C
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Which step of digitizing analog signal does this represent? A. Signal sample B. Signal compression C. Sample quantization D. 8-bit digital encoding Answer: A Explanation: As you can see in the picture, an analog waveform is sampled to covert into a numeric value by a device described in question 2?s explanation. This process is called Pulse-Amplitude-Modulation (PAM). The results of this step are often decimal numbers. Later, these decimal numbers are converted to binary numbers by the encoding process
QUESTION NO: 29 What are two benefits of using an ITSP for long distance telephony services? (Choose two.) A. Connection to an ITSP is very granular and can provision from 1 to hundreds of simultaneous calls. B. Connection to an ITSP is only available in full T1/E1/PRI circuit quantities. C. The circuits are dedicated only to voice. D. Connection to an ITSP is easier to configure becausetranscoders can seamlessly convert H.323 or SIP to SCCP. E. Connection to an ITSP allows you to use the bandwidth that is guaranteed to the voice traffic for data when voice is not using the bandwidth.
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Cisco 640-460: Practice Exam Answer: A,E
QUESTION NO: 30 How does LLQ help ensure that voice quality is maintained in a converged network? A. LLQ allocates minimum bandwidth guaranteed to voice traffic. B. LLQ allocates a priority queue to voice traffic at a guaranteed rate. C. LLQ allocates a priority queue and a minimum guaranteed bandwidth queue for voice. D. LLQ ensures that all traffic is treated fairly and hence voice traffic is not severely impacted. Answer: B
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What does the parameter Overlay on button number 2 mean?
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Cisco 640-460: Practice Exam A. Overlay means that this IP phone has more than one extension configured on line 2. B. Overlay means any calls to line 1 when busy will be automatically forwarded to line 2. C. Overlay means that the user can dial using any Overlay application, for example, Click to dial application. D. Overlay means that the extension on line 2 is part of an Overlay group that handles calls that have been sitting in the queue and exceed the "Lay" limit. Answer: A
QUESTION NO: 32 What protocol does a Cisco IP phone use to indicate to the switch how much power is needed? A. PoE B. CDP C. VTP D. cRTP Answer: B
Which statement is true about the VLAN Number field? A. You can create a new voice VLAN as long as it falls in the range 1 to 1001. B. You can create a new voice and data VLAN as long as it falls within the range 1 to 1001. C. You can create a voice VLAN as long as it falls in the range of 1 to 100. D. You can only assign a voice VLAN here in the range of 1 to 1001. E. You can create and assign a voice VLAN in the range of 1 to 1001. Answer: D
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Cisco 640-460: Practice Exam QUESTION NO: 34 A customer is in the planning stages of deploying a Cisco Unified Communications solution for their company. Previously, they were leasing a traditional PBX system from the telco and they have very little experience with voice. The customer wants to know what two signaling methods between the IP phone and the Cisco Unified Communications Manager Express are available for their use. (Choose two.) A. H.323 B. SCCP C. MGCP D. SIP E. RTP Answer: B,D
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Based on this configuration, how will an incoming call to 503 be routed? A. The call will be routed to extension 503 and then sequentially to hunt groups 1 and 2. B. The call will be routed to extension 503 and then sequentially to hunt groups 2 and 1. C. The call will be routed sequentially to users that belong to hunt group 3. D. The call will be routed sequentially to users that belong to hunt group 3 and then sequentially to hunt groups 1 and 2.
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Cisco 640-460: Practice Exam Answer: C
What is the difference between a PBX and a keysystem under the Device tab? A. A PBX is where each phone has nearly identical configuration whereas akeysystem has a unique extension for each phone. B. A PBX is where each phone has a unique extension whereas akeysystem has nearly identical configurations for each phone. C. Akeysystem can be activated only by the operator by turning the Systemkey. A PBX operates 24x7 and requires no activation. D. Incoming PSTN calls tokeysystem users have to be routed through an auto-attendant or an operator. Incoming PSTN calls to PBX users can be answered by any user on any line. Answer: B
QUESTION NO: 37 Which three options can be configured using Cisco Configuration Assistant? (Choose three.) A. Voice VLAN only B. Voice and data VLANs C. Script selection for theAutoAttendant D. Sip trunk to an ITSP E. Voice-mail archive Answer: B,C,D
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QUESTION NO: 38 In which two situations would a voice gateway be required? (Choose two.) A. To connect a corporate or branch location to an IP WAN B. To connect a corporate or branch location using VoIP to the PSTN C. To connect a Cisco Unified Communications Manager to a LAN D. To connect a Cisco Unified Communications network to a PBX E. To connect a corporate or branch location to a MAN Answer: B,D
QUESTION NO: 39 Which best describes the auto-attendant in a Cisco Unified Communications environment? A. A set of call processing instructions that automatically tell the system what to do when it reaches a particular system ID B. A system that automatically allows inside or outside callers to leave voice-mail messages 24 hours, 7 days a week, even when no operator is on duty C. A function that greets and guides callers through a telephony system in a friendly and timely manner, allowing them to reach an endpoint, leave messages, or speak to an operator D. A function that allows the option of listening to, composing, replying to, forwarding, or deleting calls or voice-mail messages through a website without the need of a live telephone operator Answer: C
QUESTION NO: 40 Which statement about the Cisco Unity Express default AutoAttendant is true? A. The defaultAutoAttendant must be enabled during the Initialization Wizard process otherwise voice-mail services will not function. B. Enabling the defaultAutoAttendant is not mandatory during the Initialization Wizard process. C. The defaultAutoAttendant is enabled by default with the exception of the prompts that must be recorded via AvT. D. The defaultAutoAttendant cannot be used as is, and it must be customized for the particular environment it will be used in. Answer: B "Pass Any Exam. Any Time." - www.itexamworld.com 24
QUESTION NO: 41 What is a numbering plan? A. It describes the path from one endpoint to another. B. It describes the calling privileges of an endpoint. C. It describes how digits are manipulated in processing a call. D. It describes call coverage in a dial plan. E. It describes the endpoint addressing used in a voice system. Answer: E Explanation: A numbering plan is a world-wide standard to organize and locate telephones all around the world. In CCNA Voice, you often read about E.164 standard.
QUESTION NO: 42 What protocol is used to monitor and provide control information about the quality of an RTP session? A. UDP B. RTP C. NTP D. RTCP Answer: D
QUESTION NO: 43 Which three of these are part of the e.164 number in the ITU-T numbering plan for geographic areas? (Choose three.) A. SANC.SPID B. Station code C. Country code D. Subscriber number E. Numbering plan area F. National destination code
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Cisco 640-460: Practice Exam Answer: C,D,F
QUESTION NO: 44 Which three are components of a dial plan? (Choose three.) A. Call legs B. Endpoint addressing C. centralized control D. Call coverage E. Digit manipulation F. Decentralized control Answer: B,D,E
QUESTION NO: 45 Which type of voice port will terminate a loop start or ground start line from the PSTN or a PBX? A. L-FXO B. FXS/FXO C. PRI T1/E1 D. E&M E. BRI F. CAS T1/E1 Answer: B
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The configuration shows a dial peer that points to Cisco Unity Express. Which line of configuration is incorrect? A. B B. C C. D D. E E. F F. G Answer: E
QUESTION NO: 47 You have been tasked to configure a Cisco Unity Express system with a voicemail pilot number of 1900, an AutoAttendant with pilot number of 2900, and an Administration via Telephone pilot number of 3900. What is the minimum number of SIP dial peers required? A. 1 B. 2 C. 3 D. 4 E. 5 Answer: A
Cisco 640-460: Practice Exam What device is responsible for converting analog voice packets from traditional voice formats to IP packets? A. MGCP gateway B. H.323 gateway C. Media Termination Point D. Digital signal processor E. Transcoder Answer: D

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How many discrete call legs are needed to set up a call between the POTS phone attached to router 1 and the phone in the PSTN? A. 3 B. 4 C. 6 D. 7 E. 8 F. 10 Answer: B
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The configuration for Call Park has been performed by a colleague before going on vacation. Users are complaining that the Call Park softkey does not appear on their phones. What can rectify this issue? A. The park-slot numberephone-dn 5 needs to be added as a line to one of the ephone 1 buttons. B. The phones need to be restarted. C. The park-slot number should be outside the range of 5XXX numbers. D. The call-park command enable needs to be added under telephony-service. Answer: B
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What will happen to a call that is transferred when no second line is available? A. It will fall back to a full-blind transfer. B. It will be transferred without consultation using H.450.2 standard methods. C. It will fall back to a blind transfer for nonlocal consultation or for a nonlocal transfer target. D. It will be transferred without consultation using the proprietary Cisco transfer method. Answer: A Explanation: The full-consult parameter will perform call transfers with consultation using second phone line if available and fallback to full-blind if second line unavailable (full-blind: Perform call transfers without consultation). For example, you are talking to Mr.A but you want him to speak with another person, Mr.B , then you can press the " Trnsfer " softkey (1st time). After hearing another dial tone, you can dial Mr.B's phone number waiting him to answer the phone. At this time, if your configuration is "full-consult" then you can continue speaking with Mr.B until you press " Trnsfer " softkey a second time. This will drop you out of the conversation and let Mr.A and Mr.B speak with each other. If your configuration is "full-blind", the phone will automatically transfer the call right after you have "Pass Any Exam. Any Time." - www.itexamworld.com 31
Cisco 640-460: Practice Exam dialed Mr.B number and you don't have a chance to speak with Mr.B before transferring your conversation. This configuration does not require you press the " Trnsfer " softkey a second time. But the "full-consult" method can only work when the second line is available, if not, it will fall back to "full-blind" method. (Notice: " Trnsfer " is the text on your phone, not "Transfer")
QUESTION NO: 53 Which three commands does Cisco recommend to use first when setting up phones for Cisco Unified Communications Manager Express? (Choose three.) A. Load B. Max-dn C. Keepalive D. Max-ephones E. Createcnf files F. Telephony-service Answer: B,D,F
QUESTION NO: 54 Which statement is true concerning time-division multiplexing? A. Time-division multiplexing transmits one voice signal at a time over a four-wire path. B. Time-division multiplexing consecutively transmits multiple voice signals across four separate communication mediums. C. Time-division multiplexing simultaneously transmits multiple separate voice signals over one communication medium by quickly interleaving pieces of each signal, one after another. D. Time-division multiplexing consecutively transmits one voice signal at a time over one or more communication mediums by quickly dividing pieces of each signal into equal bandwidth sizes and sending them in the order they are received. Information from each data channel is allocated bandwidth based on the current bandwidth needed for each time slot. This is determined by whether or not there is data that needs to be transmitted. Answer: C
QUESTION NO: 55 Benson's Able Messenger Service wants to install a new UC500 solution. Which four of these provide information from the existing numbering plan that helps you design an effective VoIP dial "Pass Any Exam. Any Time." - www.itexamworld.com 32
Cisco 640-460: Practice Exam plan? (Choose four.) A. Station number B. Subscriber code C. Direct inward dial ranges D. Length of internal number extensions E. Number of digits forwarded by the local CO F. Cisco Unified Communications Manager route pattern used for internal calls Answer: A,C,D,E

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When a call is placed to 2000, phones 1 and 2 are paged. A call to 2001 pages phones 3 and 4. What command is missing so that a call to 2002 pages all four phones? A. Paging group 20 21 B. Paging group 10 20 30 40 C. Paging group all "Pass Any Exam. Any Time." - www.itexamworld.com 33
Cisco 640-460: Practice Exam D. Paging group 2000 2001 2002 E. Paging groupephones-all Answer: A
Which inbound dial peer on CMERouter1 will be matched when phone extension 1234 places a call to 2010? A. Voip dial peer 30 B. Default dial peer 30 C. None, which will cause the call to drop D. Default dial peer Answer: D
QUESTION NO: 58 The XYZ Corporation is migrating from a traditional PBX to a Cisco Unified Communications system. While the migration is taking place they need to provide common channel signaling connectivity to the PBX from the Cisco Unified Communications system. Which type of gateway would be required? A. Analog station gateway B. Digital trunk gateway C. Analog trunk gateway D. Digital station gateway Answer: B
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Cisco 640-460: Practice Exam QUESTION NO: 59 What statement about the Cisco Unity Express AIM Module and the Cisco Unity Express Network Module is true? A. The AIM module takes up a router slot whereas the network module does not. B. The network module takes up a router slot whereas the AIM module does not. C. The AIM module runs the Linux OS whereas the network module runs the Cisco IOS software. D. The network module requires the Cisco Unified Communications Manager Express to be preconfigured whereas the AIM module does not. Answer: B
QUESTION NO: 60 What is the key command to use when deploying the partially automated telephone setup process? A. Autoqos B. Auto assign C. Auto start-dn D. Auto phone-type E. Auto start-ephone-dn F. Auto telephony-service Answer: B
QUESTION NO: 61 Which two commands need to be added to this configuration to allow IP Phones in subnet 10.100.1.0 to register with Cisco Unified Communications Manager Express at address 10.10.1.1? (Choose two.)
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A. Ip helper address 10.10.1.1 under interface FastEthernet0/0.100 B. Option 150ip 10.10.1.1 under dhcp pool Phones C. Default-router 10.100.1.1 underdhcp pool Phones D. Subnet prefix-length 24 underdhcp pool Phones E. Encapsulation dot1q 10 native under interface FastEthernet0/0.10 Answer: B,C
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QUESTION NO: 63 Which ephone-dn type does the exhibit represent? A. Sharedephone-dn B. Overlaidephone-dn C. Dual-lineephone-dn D. Single-lineephone-dn E. Dual-numberephone-dn F. Twoephone-dns with one number Answer: A
QUESTION NO: 64 Which four types of high-density trunk can be used to connect to an ITSP? (Choose four.)
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Cisco 640-460: Practice Exam A. T1/E1 CAS B. FXS C. T1/E1 PRI D. E&M E. SIP F. H.323 Answer: A,B,C,D
Which two types of dial peers are needed to complete this call end-to-end? (Choose two.) A. Serial dial peer B. PSTN dial peer C. POTS dial peer D. Network dial peer E. VoIP dial peer Answer: C,E
QUESTION NO: 66 Which command is used to assign voice VLAN 110 on a Cisco IOS switch? A. switchport access vlan 110 B. switchport access voice vlan 110 C. switchport voice vlan 110 D. switchport voice access vlan 110 Answer: C
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Cisco 640-460: Practice Exam QUESTION NO: 67 What is the relationship between a call leg and a dial peer? A. A call leg is a virtual connection to set up a call whereas a dial peer is a physical connection to complete an end-to-end call. B. The call leg and the dial peers are both logical connections used to complete an end-to-end call. C. A call leg is a virtual connection that is set up and torn down before the dial peer is established. D. The call leg and the dial peer are both physical connections used to complete an end-to-end call. Answer: B
QUESTION NO: 68 What are three functions provided by a switching system in a traditional telephony network? (Choose three.) A. Call setup B. Call supervision C. Quality of service D. Codec processing E. Voice compression F. Customer IDs and telephone numbers Answer: A,B,F
QUESTION NO: 69 Which type of voice port will be most cost effective to allow the gateway to terminate two circuits from the PSTN or a PBX? A. FXO B. FXS C. PRI T1 D. E1 E. E&M F. BRI G. CAS T1 H. E1
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Cisco 640-460: Practice Exam Answer: E
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Which ephone-dn type does the exhibit represent? A. Sharedephone-dn B. Overlaidephone-dn C. Dual-lineephone-dn D. Single-lineephone-dn E. Dual-numberephone-dn F. Twoephone-dns with one number Answer: F
QUESTION NO: 73
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Cisco 640-460: Practice Exam The Woodhull Ink Company has successfully installed the CUE module in its router's chassis. Which two configuration items are necessary for the module to function properly? (Choose two.) A. A default gateway must be assigned to the service module. B. Theip numbered command must be used to save the subnet. C. A subnet must be created that appears in all of the routing tables to make the module reachable. D. The interface service engine needs to have an IP address that is on the same subnet as the service module. E. The IP address of the service engine must be static and assigned to the interface to avoid the need for a new subnet. F. Two virtual IP addresses are needed to represent the two ends of the physical Ethernet connection across the backplane. Answer: A,D
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The company is trying to configure Music on Hold for their Cisco Unified Communications Manager Express solution. What are two possible problems with this configuration? (Choose two.) A. The second multicast entry does not have a route listed. B. Theip-source address command has not been configured. C. IP phones do not support multicast at 224.x.x.x addresses. D. You must disable the use of the first moh file with the no moh command before configuring the second file. E. Port 2000 is recommended because it is already used for normal RTP media transmissions between IP phones and the router. F. Port 2000 is not recommended because it is already used for normal RTP media transmissions between IP phones and the router. Answer: C,D
Cisco 640-460: Practice Exam The Point North Company has a few slow links in its voice and data network. Which two techniques can be used to reduce delay in voice transmission? (Choose two.) A. FIFO queuing B. Buffering voice packets C. Fragmentation of large packets D. Compression of IP, RTP, and UDP headers E. Increasing priority queue sizes Answer: C,D

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You have created a new voice VLAN 110 and would like to reassign the IP phones to the new voice VLAN. The IP phones currently reside in the default voice VLAN 100. Which tab allows you to reassign the IP phones? A. Device B. System C. Network D. Voice Features Answer: C
QUESTION NO: 77 Which of the following is selected first for an incoming dial peer?
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Cisco 640-460: Practice Exam A. Answer-address B. incoming called-number C. destination-pattern D. pots port Answer: B
QUESTION NO: 78 What is the default VTP mode on Cisco switches? A. Master B. Client C. Backup D. Server Answer: D
QUESTION NO: 79 What are the three VTP modes on Cisco switches? A. Master B. Client C. Server D. Transparent Answer: B,C,D
QUESTION NO: 80 An analog telephone is connected to a A. FXO B. E1 C. T1 D. FXS Answer: D port on a router?
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Cisco 640-460: Practice Exam QUESTION NO: 81 Choose the Gateway to the proper locations.
A. 2, 3 B. 2, 5 C. 3, 5 D. 1, 3 Answer: C
QUESTION NO: 82 What is required to convert a G711ulaw call to G729? A. Voice Termination resources B. Conferencing resources C. Converter resources D. Transcoding resources Answer: D Explanation: Transcoding compresses and decompresses voice streams to match endpoint-device capabilities. Transcoding is required when an incoming voice stream is digitized and compressed (by means of a codec) to save bandwidth, and the local device does not support that type of compression -> D is the correct answer.
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QUESTION NO: 83 What is the maximum number of phones are supported on Call Manager Express? A. 48 B. 28 C. 240 D. 500 Answer: C
QUESTION NO: 84 What protocol needs to be enabled on an ATA if a fax machine is connected to the ATA? A. MGCP B. SCCP C. H323 D. SIP Answer: C
QUESTION NO: 85 Which codec is less processor intensive? A. G729 B. G729a Answer: B
QUESTION NO: 86 What protocol needs to be enabled on an ATA if an analog telephone is connected to the ATA? A. MGCP B. SCCP C. H323 D. SIP
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Cisco 640-460: Practice Exam Answer: B
QUESTION NO: 87 What device enables Call Admission Control in a CME environment? A. Gateway B. Transcoder C. Bridge D. Gatekeeper Answer: D
QUESTION NO: 88 Which protocol provides VoiP packet sequence numbering? A. IP B. TCP C. udp D. RTP Answer: D
QUESTION NO: 89 Identify the VoIP network component that provides CAC, bandwidth control and management, and address translation. A. Gateway B. Gatekeeper C. mcu D. Call agent Answer: B
QUESTION NO: 90 Which three of the following are appropriate solutions to address latency issues in a VoIP network? (Choose 3.)
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Cisco 640-460: Practice Exam A. Usedejitter buffers B. Increase bandwidth C. Fragment data packets D. Prioritize voice packets Answer: B,C,D
QUESTION NO: 91 Which three headers are compressed by cRTP? (Choose 3.) A. Data link B. IP C. UDP D. RTP Answer: B,C,D
QUESTION NO: 92 Which of the following best describes a function of RTCP? A. RTCP provides encryption, message authentication and integrity, and anti-replay service for voice streams. B. RTCP uses even-numbered UDP ports in the range 16,384-32,767 to transport voice payloads C. RTCP provides out-of-band control information for an RTP flow D. RTCP caches an RTP packet's Layer 3 and Layer 4 headers in the routers at each end of a link, resulting in lower bandwidth demand for subsequent RTP packets. Answer: C
QUESTION NO: 93 Which two of the following VoIP gateway platforms are considered to be Integrated Services Routers (ISRs)? (Choose 2.) A. Cisco 2600XM Series B. Cisco 2800 Series C. Cisco 3700 Series D. Cisco 3800 Series
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Cisco 640-460: Practice Exam Answer: B,D
QUESTION NO: 94 In a Cisco UCM single-site deployment, what is the maximum number of IP phones that can register with a UCM cluster? A. 2500 B. 7500 C. 10,000 D. 30,000 Answer: D
QUESTION NO: 95 In a Cisco UCM multisite WAN with centralized call-processing deployment model, what redundancy feature should be configured on remote site routers to provide basic IP telephony services in the event of a WAN outage? A. AAR B. SRST C. CAC D. V3PN Answer: B
QUESTION NO: 96 Which option cannot be configured using Cisco Configuration Assistant? A. voice VLAN only B. voice and data VLANs C. script selection for theAutoAttendant D. SIP trunk to an ITSP Answer: A
QUESTION NO: 97
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Cisco 640-460: Practice Exam Which four of the following are Cisco-supported IP telephony deployment models? A. Single site B. Multisite with centralized call processing C. Clustering over the IP WAN D. Multisite with distributed call processing Answer: A,B,C,D
QUESTION NO: 98 Identify two gateway supplementary services. (Choose 2.) A. Hold B. DTMF relay C. Transfer D. Transcoding Answer: A,C
QUESTION NO: 99 According to the G.114 recommendation, the maximum one-way delay for voice should ideally not exceed how much delay? A. 100 ms B. 150 ms C. 200 ms D. 250 ms Answer: B
QUESTION NO: 100 A. national destination code B. station code C. country code D. subscriber number Answer: B
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Cisco 640-460: Practice Exam QUESTION NO: 101 Identify the preferred voice quality measurement approach for VoIP networks. A. MOS B. PESQ C. QRT D. PSQM Answer: B
QUESTION NO: 102 Which method of fax relay uses a store-and-forward approach? A. T.30 B. T.37 C. T.38 D. Cisco Fax Relay Answer: B
QUESTION NO: 103 What codec is required for fax pass-through and/or modem pass-through? A. G.711 B. G.723 C. G.729 D. G.729ab Answer: A
QUESTION NO: 104 Identify three quality issues that can result because of a lack of network bandwidth. (Choose 3.) A. Jitter B. Impedance C. Delay D. Packet loss
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Cisco 640-460: Practice Exam Answer: A,C,D
QUESTION NO: 105 What protocol is used to communicate between a DSP farm configured on an IOS router and a Cisco Unified Communications Manager server? A. H.323 B. MGCP C. SIP D. SCCP Answer: D
QUESTION NO: 106 What is the Layer 2 overhead (in bytes) for Frame Relay traffic? A. 3 bytes B. 5 bytes C. 6 bytes D. 18 bytes Answer: C
QUESTION NO: 107 What are the functions of an H.323 gateway? A. Converts an alias address to an IP address B. Responds to bandwidth requests and modifications C. Transmits and receives G.711 PCM-encoded voice D. Performs translation between audio, video, and data formats Answer: D
QUESTION NO: 108 Which call control model does MGCP use?
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Cisco 640-460: Practice Exam A. Distributed B. Centralized C. Ad hoc D. Hybrid Answer: B
QUESTION NO: 109 Which command displays a count of successful and unsuccessful control commands? A. showmgcp calls B. showmgcp statistics C. showmgcp D. debug mgcp statistics Answer: B
QUESTION NO: 110 Identify four SIP servers. (Choose 4.) A. Registrar B. Gateway C. Redirect D. Location E. Proxy Answer: A,C,D,E
QUESTION NO: 111 The call leg and the dial peers are both logical connections used to complete an end-to-end call. A. TRUE B. FALSE Answer: A
QUESTION NO: 112
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Cisco 640-460: Practice Exam A call leg is a virtual connection that is set up and torn down before the dial peer is established. A. TRUE B. FALSE Answer: B
QUESTION NO: 113 The call leg and the dial peer are both physical connections used to complete an end-to-end call. A. TRUE B. FALSE Answer: B
QUESTION NO: 114 Which of the following is not the function provided by a switching system in a traditional telephony network? A. call setup B. call supervision C. customer IDs and telephone numbers D. codec processing Answer: D
QUESTION NO: 115 Which two of the following signaling protocols are peer-to-peer protocols? (Choose two) A. H.323 B. MGCP C. SIP D. SCCP Answer: A,C
QUESTION NO: 116
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Cisco 640-460: Practice Exam A PRI (Primary Rate Interface) is a telecommunication standard used in the Integrated Services Digital Network or ISDN, for carrying multiple DS0 voice and data transmissions between two physical locations. PRI was developed specifically for industrial or large quantity users. PRI is an industrial ISDN line while the Basic Rate Interface, or BRI, is used to cater to home and small enterprises. Which three characteristics apply to ISDN PRI? (Choose three) A. PRI offers 20 B channels and 1 D channel B. the D channel is 64 kbps C. can carry data, voice, or video D. can carry vendor-specific PBX features Answer: B,C,D
QUESTION NO: 117 Case Study#1 Instructions: This task requires you to access the Cisco Unified Communications 500 Series GUI to configure it per the given requirements as stated in the scenario. From the choices on the left you may select Scenario, Cisco Unified Communications Server (CUC) Device, or Topology. The Scenario link provides the details of the task to be accomplished. The Topology link displays the network. To get access to UC500, click on the PC shown in the Topology or you can click on the CUC Device button on the left. Scenario: You have been asked to implement an SBCA solution for a startup company that has two technical support staff. The lP Phones use 4 digit extensions starting with 5001. Using the information in tables 1 and 2, configure the following: 1. Configure the AA & Voicemail features 2. Add the two IP Phones with extensions/users given in table 2 including Voicemail. 3. Configure Hunt Group so that all incoming PSTN calls always go to extension 5001. If extension 5001 is not available, route the call to extension 500: If both extensions are busy or unavailable then send the call to Voicemail Table1
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Phone Primary Ext. LastName FirstName UserID 00115C0E5EDA 5001 (Phone1I Doe John Jdoe 003094C3 D18C 5002 (Phone2) Brown Jane jbrown CUC Device: Topology:
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Cisco 640-460: Practice Exam Explanation: Click on the PC to connect to the Cisco Unified Communications 500 Series
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Pressing OK to enter the main screen of CCA. Notice that you will need to use username :Cisco / password:Cisco to login. In the Configure Panel (on the left) click on Telephony and select Voice to see all the tabs we need to configure.
It is the place where we have to solve all the requirement of this lab. So let's start! Notice: Don't press OK or Apply until all the tabs are configured! Requirement 1: Configure the AA & Voicemail features UC520, by default, only uses 3-digit extensions but all the required extensions in this lab are 4digit so we have to adjust it by clicking on the Dial Plan tab. In the "Number of Digits Per Extension" type 4, just leave other configurations unchanged because we will return to this tab later.
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The first requirement of this lab- sim is to configure the AA & Voicemail features so click on this tab to see its content
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Assign extensions to this tab as described in table 1 (Auto Attendant: 5000; Voicemail: 5111)
Requirement 2: Add the two IP Phones with extensions/users given in table 2 including Voicemail. To add two IP Phones, select Users tab and assign corresponding values to these boxes
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Requirement 3: Configure Hunt Group so that all incoming PSTN calls always go to extension 5001. If extension 5001 is not available, route the call to extension 5002. If both extensions are busy or unavailable then send the call to Voicemail. + Configure Hunt Group The last parameter in table 1 is the Hunt Group Pilot number but we haven't configured it yet. To configure this parameter, click on the Voice Features tab and you will see a section for configuring Hunt Group, use these settings ( Enable Hunt Groups: 1; Pilot#1 : 5010; Hunt Type: sequential; Forward to: Voicemail)
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Next we need to assign Phone 5001 & 5002 to this group. Return to Users tab and click on the "More" text on Phone 1
In the "Primary Extension" we can assign Phone 1 to Hunt Group 1 by setting Hunt Group of Phone 1 to 5010:
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Also set the "Call Forward Busy" and "Call Forward No Answer" to 5002 (Phone 2) Do the same thing with Phone 2, but in the "Call Forward Busy" and "Call Forward No Answer" set the extensions to 5111 (Voicemail)
(Notice: In theory, we don't need to configure "Call Forward Busy" and "Call Forward No Answer" itexamworld.com but in the exam we should configure them to make sure we get full mark.) Most things are done, but when a call coming how can we forward it to Hunt Group 1? This is the last step we need to do. Select Dial Plan Tab again, in the Incoming Call handling - FXO Trunks set the text to Hunt Group . This will make another box, "Available Hunt Groups", appears; set it to Hunt Group :1 (5010)
For your information: Below is some information about Hunt Group: Hunt groups create a shared line appearance on specified phones that hunts for members of the hunt group based on configured parameters. Phones in a hunt group can ring based on a configured sequence, based on longest idle time or in a peer fashion. If no member of the hunt group is available to take the call, behaviors can be configured for forwarding the call or sending the caller to voicemail. Note: As system configuration is done color highlighting is used both on the tab label and throughout the tab pages. To indicate if steps were completed correctly (green) highlighting is used. If additional configuration is required or if the configuration is incorrect (red) highlighting is used. The configuration cannot be saved until all items that are critical to the system are configured. Once all configuration is correct it can be saved to memory.
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Cisco 640-460: Practice Exam QUESTION NO: 118 Case Study# 2 Instructions: To configure the Cisco Unified Communications Manager Express click on the console host icon that is connected to a Cisco Unified Communications Manager Express by a serial console cable (shown in the diagram as a dashed black line). You can click on the buttons below to view the different windows. Each of the windows can be minimized by clicking on the [-]. You can also reposition a window by dragging it by the title bar. The "Tab" key and most commands that use the "Control" or "Escape" keys are not supported and not necessary to complete this simulation. The help command does not display all commands of the help system.
Scenario: You are required to manually configure a Cisco Unified Unified Communications Manager Express to support two IP phones with directory numbers starting with 5001. The phones need to be configured as dual-line phones as per the following table: The Cisco Unified Communications Manager Express should be configured as a DHCP and TFTP server. The IP phones should obtain their IP addresses via DHCP in the range 10.3.130.10 10.3.130.254. IP addresses in the range 10.3.130.1 to 10.3.130.9 should be excluded. The IP phones are connected to an EtherSwitch module, interface GigabitEthernet1/0 on a Cisco 2811 router as per the topology diagram. The voice VLAN ID is 130 and default router for the IP phones is 10.3.130.1. The EtherSwitch module has been preconfigured and is not configurable. You are also required to configure a second directory number 5010 as a shared directory number on line 2 of each IP phone. Topology:
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Explanation: CME#conf t CME( config )# ip dhcp excluded-address 10.3.130.1 10.3.130.9 CME( config )# ip dhcp pool Company CME( dhcp-config )#network 103.130.0 255.255.255.0 CME( dhcp-config )#default-router 10.3.130.1 CME( dhcp-config )#option 150 ip 10.3.130.1 CME( dhcp-config )#exit CME( config )#telephony-service CME( config -telephony)#max- ephone 2 CME( config -telephony)#max- dn 3 CME( config -telephony)# ip source-address 10.3.130.1 CME( config -telephony)#create cnf -file CME( config -telephony)#exit CME( config )# ephone-dn 1 dual-line CME( config-ephone-dn )#number 5001 CME( config-ephone-dn )#exit CME( config )# ephone-dn 2 dual-line CME( config-ephone-dn )#number 5002 CME( config-ephone-dn )#exit CME( config )# ephone-dn 3 dual-line CME( config-ephone-dn )#number 5010 CME( config-ephone-dn )#exit CME( config )# ephone 1 CME( config-ephone )# mac -address 0014.1CBC.E179 CME( config-ephone )#button 1:1 2:3 "Pass Any Exam. Any Time." - www.itexamworld.com 65
Cisco 640-460: Practice Exam CME( config-ephone )#reset CME( config-ephone )#exit CME( config )# ephone 2 CME( config-ephone )# mac -address 0003.E3C4.463C CME( config-ephone )#button 1:2 2:3 CME( config-ephone )#reset CME( config-ephone )#exit CME( config )#exit CME#copy running- config startup- config
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Cisco 640-460: Practice Exam QUESTION NO: 1 Which type of voice port will allow the gateway to terminate

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Cisco 640-460: Practice Exam

QUESTION NO: 4 Refer to the exhibit.

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Cisco 640-460: Practice Exam

QUESTION NO: 5 Refer to the exhibit.

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QUESTION NO: 7 DRAG DROP

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Cisco 640-460: Practice Exam

QUESTION NO: 8 Refer to the exhibit.

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Cisco 640-460: Practice Exam Answer: E,F

QUESTION NO: 12 "Pass Any Exam. Any Time." - www.itexamworld.com 7

Cisco 640-460: Practice Exam Refer to the exhibit.

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Cisco 640-460: Practice Exam QUESTION NO: 14 Refer to the exhibit.

QUESTION NO: 15 DRAG DROP

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Cisco 640-460: Practice Exam

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Cisco 640-460: Practice Exam

Cisco 640-460: Practice Exam B. 2 C. 4 D. 6 Answer: B

QUESTION NO: 18 DRAG DROP

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QUESTION NO: 20 Refer to the exhibit.

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Cisco 640-460: Practice Exam QUESTION NO: 21 Refer to the exhibit.

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Cisco 640-460: Practice Exam

QUESTION NO: 23 Refer to the exhibit.

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QUESTION NO: 25 DRAG DROP

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Cisco 640-460: Practice Exam Explanation:

QUESTION NO: 26 Refer to the exhibit.

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QUESTION NO: 27 Refer to the exhibit.

QUESTION NO: 28 Refer to the exhibit.

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Cisco 640-460: Practice Exam

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Cisco 640-460: Practice Exam Answer: A,E

QUESTION NO: 31 Refer to the exhibit.

What does the parameter Overlay on button number 2 mean?

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QUESTION NO: 33 Refer to the exhibit.

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QUESTION NO: 35 Refer to the exhibit.

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Cisco 640-460: Practice Exam Answer: C

QUESTION NO: 36 Refer to the exhibit.

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Cisco 640-460: Practice Exam

Cisco 640-460: Practice Exam

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Cisco 640-460: Practice Exam Answer: C,D,F

QUESTION NO: 46 Refer to the exhibit.

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Cisco 640-460: Practice Exam

QUESTION NO: 48 "Pass Any Exam. Any Time." - www.itexamworld.com 27