Cisco_CertifyMe_ _v _88q_By-Amik

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1 Cisco_CertifyMe_ _v _88q_By-Amik Number: Passing Score: 800 Time Limit: 120 min File Version: Exam Cisco Code Total Questions 88 Read all Questions..! By Amik Sections 1. Voice Fundamentals 2. Analog Voice Port 3. Call Signaling 4. Internet Telephony Service Provider 5. Call Routing and Path Selection 6. VoIP Gateway 7. Digital Voice Port 8. Advanced Dial Plan 9. Dial Peer 10. VoIP Design Element 11. Gatekeeper 12.Drag and Drop

2 Exam A QUESTION 1 Which two codes together make up the number that follows the E.164 recommendation numbering scheme? (Choose two.) A. country code B. subscriber code C. national destination code D. provider code Correct Answer: AB Section: Voice Fundamentals /Reference: E.164 is an international numbering plan created by the International Telecommunication Union (ITU). Each number in the E.164 numbering plan contains the following components: Country code (CC) National destination code (NDC - optional) Subscriber number (SN) The CC consists of one, two or three digits. It is what we add in order to access different countries and often prefixed with a + The NDC is the code we often call the area code. The SN is for telephone numbering. It is given by your phone operator. E.164 numbers are limited to a maximum length of 15 digits. For example, the North American Numbering Plan E.164 is as follows: : Country code : National destination code (for North American Numbering Plan, 602 is called the area code while 555 is called Central Office Code) : Subscribe Number Answer C is also correct but just optional. E.164 Numbering Plan must have Country Code and Subscriber Code so A & B are the correct answers. QUESTION 2 Which statement is true about only out-of-band signaling? A. A signaling bit is robbed from each frame. B. Signaling bits are sent in a special order in a dedicated signaling frame. C. All signaling is directly associated with its corresponding voice frame. D. All voice packets carry their own signaling. Correct Answer: B Section: Voice Fundamentals /Reference:

3 Out-of-Band signaling is telecommunication signaling exchange of information in order to control a telephone call. Out-of-Band signaling uses common channel signaling (CCS), that means signaling information is transmitted using a separate, dedicated signaling channel. Answers A C D are characteristics of Channel associated signaling (CAS) so they are not correct. QUESTION 3 The D channel in ISDN is an example of which two signaling methods? (Choose two.) A. CCS signaling B. out-of-band signaling C. in-band signaling D. CAS signaling Correct Answer: AB Section: Voice Fundamentals /Reference: QUESTION 4 In North America, which E&M signaling type is used most often for geographically separated equipment? A. Type I B. Type II C. Type III D. Type IV E. Type V Correct Answer: B Section: Voice Fundamentals /Reference: This information is quoted from technologies_tech_note09186a f60.shtml E&M Type I - This is the most common interface in North America. Type I uses two leads for supervisor signaling: E, and M. During inactivity, the E-lead is open and the M-lead is connected to the ground. The PBX (that acts as trunk circuit side) connects the M-lead to the battery in order to indicate the off-hook condition. The Cisco router/gateway (signaling unit) connects the E-lead to the ground in order to indicate the off-hook condition. E&M Type II - Two signaling nodes can be connected back-to-back. Type II uses four leads for supervision signaling: E, M, SB, and SG. During inactivity both the E-lead and M-lead are open. The PBX (that acts as trunk circuit side) connects the M-lead to the signal battery (SB) lead connected to the battery of the signaling side in order to indicate the off-hook condition. The Cisco router / gateway (signaling unit) connects the E-lead to the signal ground (SG) lead connected to the ground of the trunk circuit side in order to indicate the off-hook condition. E&M Type III - This is not commonly used in modern systems. Type III uses four leads for supervision signaling: E, M, SB, and SG. During inactivity, the E-lead is open and the M-lead is set to the ground connected to the SG lead of the

4 signaling side. The PBX (that acts as trunk circuit side) disconnects the M-lead from the SG lead and connects it to the SB lead of the signaling side in order to indicate the off-hook condition. The Cisco router / gateway (signaling unit) connects the E-lead to the ground in order to indicate the off-hook condition. E&M Type IV - This is not supported by Cisco routers / gateways. E&M Type V - Type V is symmetrical and allows two signaling nodes to be connected back-to-back. This is the most common interface type used outside of North America. Type V uses two leads for supervisor signaling: E, and M. During inactivity the E-lead and M-lead are open. The PBX ( that acts as trunk circuit side) connects the M-lead to the ground in order to indicate the off-hook condition. The Cisco router / gateway (signaling unit) connects the E-lead to the ground in order to indicate off-hook condition. Although above information specifies E&M Type 1 is the most commonly used interface in North America but this type generates significant delay in the signaling operation when transmitting between geographically separated equipment and affects voice signal quality (because of significant inductance and capacitance of the long wires) so Type 2 is often used instead. QUESTION 5 Which three are supervisory signals? (Choose three.) A. busy B. on hook C. off hook D. call waiting E. ring Correct Answer: BCE Section: Voice Fundamentals /Reference: QUESTION 6 What is the approximate frequency range of human speech? A. 20 Hz to 20,000 Hz B. 40 Hz to 15,000 Hz C. 200 Hz to 9000 Hz D. 600 Hz to 5400 Hz Correct Answer: C Section: Voice Fundamentals

5 /Reference: QUESTION 7 What is the process of assigning audio amplitude to a unique digital code word? A. linear prediction B. encoding C. sampling D. quantization Correct Answer: D Section: Voice Fundamentals /Reference: QUESTION 8 What is the E.164 standard? A. private numbering plan B. national numbering plan C. dial plan D. international public telecommunications numbering plan Correct Answer: D Section: Voice Fundamentals /Reference: QUESTION 9 For the following items, which is the most common E&M type used outside North America? A. Type IV B. Type I C. Type II D. Type III E. Type V Correct Answer: E Section: Voice Fundamentals /Reference: QUESTION 10 A new business in Great Britain needs to have a PSTN connection that will handle a maximum of 30 inbound and outbound calls at any given time. The customer only has one slot available on the designated PSTN router. Which digital line type should be recommended?

6 A. QSIG B. ISDN BRI C. ISDN E1 PRI D. ISDN T1 PRI Correct Answer: C Section: Voice Fundamentals /Reference:

7 Exam B QUESTION 1 Refer to the exhibit for IP addresses and telephone numbers. You are working with a customer opening a small sales office in Atlanta. You want the user in Atlanta to be able to dial into the PBX in New York over the IP WAN. The New York PBX uses ground start, a two-wire operation, and DTMF dialing. Choose the correct FXO port configuration commands for New York. Exhibit: A. voice-port 1/0/0 signal ground-start operation 2-wire dial-type dtmf B. voice-port 1/1/1 destination signal ground-start operation 2-wire type 1 dial-type dtmf C. voice port 1/0/0 session target ipv4: destination signal ground-start operation 2-wire dial-type dtmf D. voice port 1/0/0 session target ipv4: source signal wink-start operation 2-wire dial-type dtmf Correct Answer: A Section: Analog Voice Port /Reference: QUESTION 2 Refer to the exhibit. Which configuration option will allow communication between a voice-enabled router and a PBX? Exhibit:

8 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 Correct Answer: A Section: Analog Voice Port /Reference: QUESTION 3 Examine the following PBX system parameters: The calling side seizes the line by going off-hook on its E-lead and sends information as DTMF digits. The voice path is 4-wires, and the voice enabled router is in another building from the PBX. Select the correct set of commands to allow communication between a voice enabled router and a PBX. A. voice port 1/0/0 signal immediate-start operation 4-wire type 2 B. voice-port 1/0/0 signal delay-dial operation 4-wire type 1 C. voice port 1/0/0 signal wink-start operation 4-wire type 3 D. voice port 1/0/0

9 signal immediate-start operation 4-wire type 4 Correct Answer: A Section: Analog Voice Port /Reference: QUESTION 4 Which three services are supported by CUBE when supporting H323-to-SIP calls? (Choose three.) A. SIP cause codes B. media flow-around C. media flow-through D. codec transparent support E. Transport Layer Security F. H.261, H.263, and H.264 video codecs Correct Answer: CDE Section: Call Signaling /Reference: QUESTION 5 Which statement is true about MGCP? A. Call completion is always shared, with some intelligence on the endpoint, some on the call agent. B. Endpoints always take all actions to complete calls. C. Endpoints may act alone or cooperate with call agent to complete calls. D. Call agents order and direct each step of call completion for the endpoints. Correct Answer: D Section: Call Signaling /Reference:

10 Exam C QUESTION 1 Which option is true concerning the MGCP call agent? A. acts only as a recorder of call details B. provides only call signaling and call setup C. manages all aspects of the call and voice stream D. monitors the quality of each call after setup Correct Answer: B Section: Call Signaling /Reference: MGCP Call Agent is a central control component to remotely control various devices. When the MGCP call agent exists in the network, calls are routed via route patterns on the Call Agent (Cisco Unified Communications Manager), not by dial peers on the gateway. The messages sent between the voice gateway and the MGCP Call Agent are just used for call signaling and call setup only. In summary, the Call Agent will instruct the gateways what to do in each stage: receive dialed digits, find the destination gateway, send connection request... Finally, the Call Agent will allow gateways to establish RTP Streams with each other. Notice that the voice streams only flow between the two voice gateways, not to the Call Agent. At the conversation finishs (one of the endpoints goes on-hook), that gateway notifies the Call Agent and the Call Agent sends Delete Connection (DLCX) Requests for both gateways. QUESTION 2 At what point does the MGCP call agent release the setup of the call path to the residential gateways? A. after the call agent has been notified that an event occurred at the source residential gateway B. after the call agent has been notified of an event and has instructed the source residential gateway to create a connection C. does not release call path setup D. after the call agent has sent a connection request to both the source and destination and has relayed a modify-connection request to the source so that the source and destination can set up the call path E. after the call agent has forwarded session description protocol information to the destination from the

11 source and has sent a modify connection to the destination and a create-connection request to the source Correct Answer: D Section: Call Signaling /Reference: Below is the call flow between two voice gateway through a MGCP Call Agent The MGCP call agent releases the setup of the call path to the residential gateways when the conversation begins. After sending the Modify Connection (MDCX), the two gateways have enough information to start the conversation so the duty of the Call Agent finishs. QUESTION 3 Which three services are supported by CUBE when supporting H323-to-SIP calls? (Choose three.) A. SIP cause codes B. media flow-around C. media flow-through

12 D. codec transparent support E. Transport Layer Security F. H.261, H.263, and H.264 video codecs Correct Answer: CDE Section: Call Signaling /Reference: QUESTION 4 Which two are attributes of SCCP? (Choose two.) A. It is Cisco proprietary. B. It is a supervisory signaling protocol. C. It is classified as client/server architecture. D. SCCP devices are considered intelligent endpoints. Correct Answer: AC Section: Call Signaling /Reference: QUESTION 5 Refer to the exhibit. All IP phones are SCCP phones. Phone D makes an internal call to phone G. Which call setup signaling statement is true? Exhibit:

13 A. Phone D signals phone G directly. Call setup is handled by the phones. B. Phone D signals gateway A, which processes the call and signals phone G. C. Phone D signals gateway B, which processes the call and signals phone G. D. Phone D signals gatekeeper. The gatekeeper processes the call and signals phone G. E. Phone D signals the call agent. The call agent processes the call and signals phone G. Correct Answer: E Section: Call Signaling /Reference: QUESTION 6 Which statement is true about MGCP? A. Call completion is always shared, with some intelligence on the endpoint, some on the call agent. B. Endpoints always take all actions to complete calls. C. Endpoints may act alone or cooperate with call agent to complete calls. D. Call agents order and direct each step of call completion for the endpoints. Correct Answer: D Section: Call Signaling /Reference:

14 QUESTION 7 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 Correct Answer: A Section: Digital Voice Port /Reference:

15 Exam D QUESTION 1 You work as a network technician, study the exhibit carefully. The Acme Corp. uses H.323 to place calls to their supplier RR Industries. Acme also has a voice connection to an ITSP for long distance over a SIP network. Which configuration should Acme use to deploy the CUBE? Hot Area: Correct Answer:

16 Section: Internet Telephony Service Provider /Reference: The Acme Corp connects to the ITSP via SIP Trunk and connects to RR industries via H.323. The Acme Corp itself uses H.323 so we have to enable protocol interworking with allow-connections commands: allow-connections h323 to h323: allow Acme Corp to communicate with RR industries (in both ways) allow-connections h323 to sip: allow Acme Corp to talk with ITSP (Acme Corp can talk and ITSP can hear but not vice versa) allow-connections sip to h323: allow ITSP to talk with Acme Corp (Acme Corp can hear and ITSP can talk but not vice versa) Notice that the configuration for H.323 and SIP interworking is unidirectional, thus if bidirectional interworking is required, you need to configure the mirror-matching statement as well. Acme Corp doesn't use SIP so we don't need to configure "allow-connections sip to sip". QUESTION 2 H.323 is an umbrella Recommendation from the ITU Telecommunication Standardization Sector (ITU-T) that defines the protocols to provide audio-visual communication sessions on any packet network. Which CUBE

17 configuration will support H.323 protocol interworking and address hiding? A. voice services voip h323 interworking media flow-around B. voice services h323 to h323 h323 interworking media flow-through C. voice services voip allow-connections h323 to h323 media flow-around D. voice service voip allow-connections h323 to h323 Correct Answer: D Section: Internet Telephony Service Provider /Reference: QUESTION 3 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) Exhibit: A. dial-peer voice 50 voip destination-pattern session protocol sipv2 session-target ipv4: B. dial-peer voice 1000 voip destination-pattern session-target ipv4: C. dial-peer 91 voip session protocol sipv2 destination-pattern 91T session-target ipv4: dtmf-relay rtp-nte digit-drop h245-alphanumeric

18 D. dial-peer 91 voip destination-pattern 91T session-target ipv4: dtmf-relay rtp-nte digit-drop h245-alphanumeric E. dial-peer voice 1000 voip destination-pattern session-target ipv4: F. dial-peer voice 50 voip destination-pattern session-target ipv4: Correct Answer: BCF Section: Internet Telephony Service Provider /Reference: QUESTION 4 Examine the example output. hostname GW1! interface Ethernet 0/0 ip address h323-gateway voip interface h323-gateway voip id GK1-zone1.abc.com abc.com ipaddr h323-gateway voip h323-id GW1 h323-gateway voip bind srcaddr ! dial-peer voice 1 voip destination-pattern session-target ras! dial-peer voice 2 pots destination-pattern no register e164! end Choose the command that will restore communication with gatekeeper functionality to this device. A. h323-gateway voip h323-id GK1 B. gateway C. h323-gateway voip bind srcaddr D. h323-gateway voip GW1-zone2.abc.com abc.com ipaddr Correct Answer: B Section: VoIP Gateway /Reference: 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.

19 Exam E QUESTION 1 When setting up a VoIP call, what is the first thing a gateway router tries to match to a dialed number? A. call leg B. IP route C. session target D. destination pattern Correct Answer: D Section: Call Routing and Path Selection /Reference: First, the gateway attempts to match the called number with the incoming called-number. 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 2 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? Exhibit:

20 A. Increase the ISDN T302 timer to allow more time for call setup. B. Add the command isdn negotiate-bchan to the serial interface. C. Add the command isdn contiguous-bchan to the serial interface. D. Change the channel selection order from descending to ascending. Correct Answer: B Section: Call Routing and Path Selection /Reference: QUESTION 3 Refer to the exhibit. The Carmichael caller dials the site access code for Merrimack (6) followed by the fourdigit 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? Exhibit:

21 A. Translate to B. Strip the site access code and send four digits. C. Strip the site access code and prepend D. Do nothing because the site access code matches the last five digits of the target number. E. Strip the site access code, send four digits, then prepend the access code when it reaches the Merrimack gateway. Correct Answer: B Section: Call Routing and Path Selection /Reference: 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 4 Which path selection mechanism lets you choose either the even or odd channels first? A. hunt groups B. trunk groups C. tailend hopoff D. Call Admission Control Correct Answer: B Section: Call Routing and Path Selection /Reference: By using trunk groups, we can choose to use either the even or odd channels first with the command: hunt-scheme... [even odd...] (notice: the full command is very long so I shorten it to the simplest form) QUESTION 5 When using CUBE, which two statements describe how media flow-through differs from media flow-around? (Choose two.) A. Media flow-around provides address hiding by terminating both signaling and RTP streams.

22 B. Media flow-through terminates the signaling channel and the RTP streams flow directly between endpoints. C. Media flow-around and media flow-through function in a similar manner, but media flow-around supports NAT traversal. D. Media flow-through terminates the RTP streams but allows signaling to flow directly between endpoints. E. Media flow-around terminates the signaling stream and allows RTP streams to flow directly between endpoints. F. Media flow-through provides address hiding by terminating both signaling and RTP streams. Correct Answer: EF Section: Call Routing and Path Selection /Reference: QUESTION 6 Refer to the IOS configuration in the exhibit. How will the next incoming call be routed? Exhibit: A. The call will be routed to the longest idle channel. B. The call will be routed to the least used channel. C. The call will be routed to a random available channel. D. The call will be routed to the next available channel, starting from channel 1, hunting up toward channel E. The call will be routed to the next available channel, starting from channel 24, hunting down toward channel

23 1. Correct Answer: E Section: Call Routing and Path Selection /Reference: 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 oddnumbered channel is sought. + odd: Searches for an idle odd numbered channel. If no idle odd numbered channel is available, an evennumbered 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. 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 <number from 0 to 23> command is used to customize T1/CAS signaling parameters for a particular T1 channel group on a channelized T1 line. QUESTION 7 Site A uses three-digit internal numbers and remote Site B uses four-digit internal numbers. All calls to the PSTN are routed through Site B. What dial plan below best represents provision simplicity, assuming the NANP numbering plan? A. Translate all called numbers within Site A to four digits. B. Translate all called numbers within Site B to three digits. C. Translate all called numbers leaving Site A to ten digits. D. Translate all called numbers at either site to ten digits. Correct Answer: C Section: Call Routing and Path Selection /Reference: QUESTION 8 Which command parameter specifies that the router should not attempt to initiate a trunk connection but should wait for an incoming call before establishing the trunk? A. codec clear-channel B. connection-trunk answer-mode

24 C. voice-port 1/0:1 D. ds0-group timeslots 1-23 type ext-sig Correct Answer: B Section: (none) /Reference: QUESTION 9 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) Exhibit: A. dial-peer voice 50 voip destination-pattern session protocol sipv2 session-target ipv4: B. dial-peer voice 1000 voip destination-pattern session-target ipv4: C. dial-peer 91 voip session protocol sipv2 destination-pattern 91T session-target ipv4: dtmf-relay rtp-nte digit-drop h245-alphanumeric D. dial-peer 91 voip destination-pattern 91T session-target ipv4: dtmf-relay rtp-nte digit-drop h245-alphanumeric E. dial-peer voice 1000 voip destination-pattern session-target ipv4: F. dial-peer voice 50 voip destination-pattern session-target ipv4:

25 Correct Answer: BCF Section: Internet Telephony Service Provider /Reference:

26 Exam F QUESTION 1 Refer to the exhibit. Choose the correct configuration command set that will allow the gateway in zone BR to register with the gatekeeper in the same zone. Exhibit: A. interface fastethernet 0/0 ip address h323-gateway voip interface h323-gateway voip id BR ipaddr h323-gateway voip h323-id BR! gateway B. interface fastethernet 0/0 ip address h323-gateway voip interface h323-gateway voip id BR ippaddr h323-gateway voip h323-id BR! gateway C. interface fastethernet 0/0 ip address h323-gateway voip interface h323-gateway voip id BR ipaddr h323-gateway voip h323-id BR! gateway D. interface fastethernet 0/0 ip address h323-gateway voip interface h323-gateway voip id BR ipaddr h323-gateway voip h323-id BR! gateway E. interface fastethernet 0/0 ip address h323-gateway voip interface h323-gateway voip id HQ ipaddr

27 h323-gateway voip h323-id BR! gateway Correct Answer: A Section: VoIP Gateway /Reference: QUESTION 2 Examine the example output. hostname GW1! interface Ethernet 0/0 ip address h323-gateway voip interface h323-gateway voip id GK1-zone1.abc.com abc.com ipaddr h323-gateway voip h323-id GW1 h323-gateway voip bind srcaddr ! dial-peer voice 1 voip destination-pattern session-target ras! dial-peer voice 2 pots destination-pattern no register e164! end Choose the command that will restore communication with gatekeeper functionality to this device. A. h323-gateway voip h323-id GK1 B. gateway C. h323-gateway voip bind srcaddr D. h323-gateway voip GW1-zone2.abc.com abc.com ipaddr Correct Answer: B Section: VoIP Gateway /Reference: 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 3 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 4 Interconnects segments of the same or different VoIP network using different signaling types A. Gateway - 1 and 2 CUBE - 3 and 4 B. Gateway - 1 and 3

28 CUBE - 2 and 4 C. Gateway - 2 and 3 CUBE - 1 and 4 D. Gateway - 2 and 4 CUBE - 1 and 3 Correct Answer: A Section: VoIP Gateway /Reference: QUESTION 4 Using Cisco Unified Communications Manager Express, what four steps are necessary to implement COR? (Choose four.) A. Configure SRST. B. Define COR labels. C. Configure COR lists. D. Assign COR list to ephone-dn. E. Configure COR lists on voice ports. F. Configure dial peers and assign COR lists. Correct Answer: BCDF Section: Advanced Dial Plan /Reference: 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(config-dp-cor)#name intl

29 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 <corlist-name>. Router(config-dp-corlist)#dial-peer cor list 911call Router(config-dp-corlist)#member 911 Router(config-dp-corlist)#dial-peer cor list localcall Router(config-dp-corlist)#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 dial-peer cor list <corlist-name>. Router(config)#dial-peer cor list local Router(config-dp-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)#destination-pattern 911 Router(config-dial-peer)#forward-digits all Router(config-dial-peer)#corlist outgoing 911call Router(config-dial-peer)#port 0/0/0:23 Router(config)#dial-peer voice 9911 pots Router(config-dial-peer)#destination-pattern 9911 Router(config-dial-peer)#forward-digits 3 Router(config-dial-peer)#corlist outgoing 911call Router(config-dial-peer)#port 0/0/0:23 Router(config)#dial-peer voice 9 pots Router(config-dial-peer)#destination-pattern 9[2-9]... Router(config-dial-peer)#corlist outgoing localcall

30 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(config-dial-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)

31 Exam G QUESTION 1 A customer needs to configure a CAS E&M circuit that will support inbound and outbound DNIS and inbound ANI. Which configuration will accomplish this task? A. pri-group timeslots 1-24 B. ds0-group 0 timeslots 1-24 type none C. ds0-group 0 timeslots 1-24 type e&m-fgd D. ds0-group 0 timeslots 1-24 type fgd-eana E. ds0-group 0 timeslots 1-31 type r2-digital r2-compelled ani Correct Answer: C Section: Digital Voice Port /Reference: 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}

32 (Reference: T1 CAS always provides the ANI/DNIS delimiter on incoming T1/CAS trunk lines. The customer wants E&M circuit so the answer should be C. Notice: + CAS signaling main feature is its use of user bandwidth to perform signaling functions. CAS signaling is often referred to as robbed-bit-signaling 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 2 In T1 CAS, where are the signaling states and control features carried for Super Frame robbed-bit signaling?

33 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 Correct Answer: A Section: Digital Voice Port /Reference: QUESTION 3 Refer to the exhibit. The Carmichael caller dials the site access code for Merrimack (6) followed by the fourdigit 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? Exhibit: A. Translate to B. Strip the site access code and send four digits. C. Strip the site access code and prepend D. Do nothing because the site access code matches the last five digits of the target number. E. Strip the site access code, send four digits, then prepend the access code when it reaches the Merrimack gateway. Correct Answer: B Section: Call Routing and Path Selection /Reference: 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 4 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?

34 A. codec clear-channel B. connection-trunk answer-mode C. voice-port 1/0:1 D. ds0-group timeslots 1-23 type ext-sig Correct Answer: B Section: (none) /Reference:

35 Exam H QUESTION 1 Which mechanism do you use to implement calling privileges on Cisco Unified Communications Manager Express? A. CoS B. QoS C. CAC D. COR E. SRST Correct Answer: D Section: Advanced Dial Plan /Reference: 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: technologies_configuration_example09186a008019d649.shtml) QUESTION 2 Using Cisco Unified Communications Manager Express, what four steps are necessary to implement COR? (Choose four.) A. Configure SRST. B. Define COR labels. C. Configure COR lists. D. Assign COR list to ephone-dn. E. Configure COR lists on voice ports. F. Configure dial peers and assign COR lists. Correct Answer: BCDF Section: Advanced Dial Plan /Reference: 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.

36 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(config-dp-cor)#name intl 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 <corlist-name>. Router(config-dp-corlist)#dial-peer cor list 911call Router(config-dp-corlist)#member 911 Router(config-dp-corlist)#dial-peer cor list localcall Router(config-dp-corlist)#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 dial-peer cor list <corlist-name>. Router(config)#dial-peer cor list local Router(config-dp-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)#destination-pattern 911 Router(config-dial-peer)#forward-digits all Router(config-dial-peer)#corlist outgoing 911call Router(config-dial-peer)#port 0/0/0:23

37 Router(config)#dial-peer voice 9911 pots Router(config-dial-peer)#destination-pattern 9911 Router(config-dial-peer)#forward-digits 3 Router(config-dial-peer)#corlist outgoing 911call Router(config-dial-peer)#port 0/0/0:23 Router(config)#dial-peer voice 9 pots Router(config-dial-peer)#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(config-dial-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 3 Refer to the exhibit. Which dial peer configuration will block phone A from making long distance calls? Exhibit:

38 A. dial-peer voice 1374 pots destination-pattern 1374 port1/0/0 dial-peer voice 100 voip corlist incoming Intl01 destination-pattern 9011T session target ipv4: B. dial-peer voice 1374 pots destination-pattern 1374 port1/0/0 dial-peer voice 100 voip corlist outgoing Intl01 destination-pattern 9011T session target ipv4: C. dial-peer voice 1374 pots corlist incoming LDLst destination-pattern 1374 port1/0/0 dial-peer voice 100 voip destination-pattern 9011T session target ipv4: D. dial-peer voice 1374 pots corlist outgoing LDLst destination-pattern 1374 port1/0/0 dial-peer voice 100 voip destination-pattern 9011T session target ipv4: E. dial-peer voice 1374 pots corlist incoming LocalLst destination-pattern 1374 port1/0/0 dial-peer voice 100 voip corlist outgoing Intl01 destination-pattern 9011T session target ipv4: F. dial-peer voice 1374 pots corlist outgoing LDLst destination-pattern 1374 port1/0/0 dial-peer voice 100 voip corlist outgoing Intl01 destination-pattern 9011T session target ipv4: Correct Answer: E Section: Advanced Dial Plan /Reference: QUESTION 4 Where would you assign COR lists in Cisco Unified Communications Manager Express? A. ephone

39 B. ephone-dn C. voice register dn D. voice register pool Correct Answer: B Section: Advanced Dial Plan /Reference: For Cisco Unified Communications Manager Express, the COR list is directly assigned to the appropriate Ethernet phone-dn (ephone-directory number) QUESTION 5 The D channel in ISDN is an example of which two signaling methods? (Choose two.) A. CCS signaling B. out-of-band signaling C. in-band signaling D. CAS signaling Correct Answer: AB Section: Voice Fundamentals /Reference: QUESTION 6 What is the process of assigning audio amplitude to a unique digital code word? A. linear prediction B. encoding C. sampling D. quantization Correct Answer: D Section: Voice Fundamentals /Reference:

40 Exam I QUESTION 1 Refer to the exhibit. You have been asked to configure a dial peer on R2 that will match only the extensions of the four telephones attached. Which dial-peer statement will you use? Exhibit: A. dial-peer voice 1 pots destination-pattern 5552.[0-5]0 B. dial-peer voice 1 pots destination pattern 5552[5-6].0 C. dial-peer voice 1 pots destination-pattern 555[2-5][56] D. dial-peer voice 1 pots destination-pattern 5552[5-6][05]0 Correct Answer: D Section: Dial Peer /Reference: 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 2 Refer to the exhibit. When extension dials , how are the digits manipulated in R1 so that they are presented correctly at R2?

41 Exhibit:

42 A. The outbound VoIP dial peer is matched and all digits are sent. B. The digits are stripped off before matching the outbound POTS dial peer. C. The digits are stripped off by the connection trunk and R2 receives only D. R1 collects the 1200 and prepends the tie-line digits That number is matched to a VoIP dial peer and sent to the appropriate address. Correct Answer: A Section: Dial Peer

43 /Reference: When (Phone A) calls (Phone B) the dial-peer voice 1 voip at R1 is matched with the destination-pattern But notice that this is a voip dial-peer so digits are not stripped and all digits are sent to R2. QUESTION 3 Refer to the exhibit. Your customer wants to converge the existing PBX network with the IP network. The three remote offices have various types of PBXs. The customer is using a combination of tie-lines and trunks to connect the PBXs today. Which kind of connection should be implemented to allow calls to be placed from to so that when the call is completed, network resources are returned for other uses? Exhibit: A. PLAR B. trunk C. tie-line D. answer-mode Correct Answer: C Section: Dial Peer /Reference: E&M signaling supports tie-line type facilities. QUESTION 4 Which dial plan characteristic shows the most obvious improvement by dropping a number translation step? A. availability

44 B. post-dial delay C. scalability D. hierarchical design Correct Answer: B Section: Dial Peer /Reference: 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. By dropping a number translation step, the post-dial delay time will be obvious improvement. QUESTION 5 Refer to the exhibit. Users are not able to complete a call from to What is the correct diagnosis for the problem? Exhibit:

45 A. incorrect destination-pattern in router 1 B. incorrect POTS dial-peer statement in router 2 C. incorrect session-target statement in router 2 D. incorrect port statement in router 1 pots dial peer E. missing no digit-strip on the voip dial peer in router 1 Correct Answer: A Section: Dial Peer /Reference: QUESTION 6 Refer to the exhibit. Three department managers share the directory number 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

46 manager's phone? Exhibit: A. The Marketing manager is on the phone. B. None of the managers are on the phone. C. The Engineering manager is on the phone. D. The Shipping manager and Marketing manager are on the phone. E. The Engineering manager and Marketing manager are on the phone. Correct Answer: E Section: Dial Peer /Reference: 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 7 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? Exhibit: A. no vad

47 B. tech-prefix 1# C. codec g729ar8 bytes 30 D. dtmf-relay h245-alphanumeric Correct Answer: D Section: Dial Peer /Reference: QUESTION 8 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 destination-pattern no digit-strip prefix 5501 port 1/0/0 A. A nine digit number beginning with 5501 will be forwarded. B. A ten digit number beginning with 5501 will be forwarded. C. A twelve digit number beginning with will be forwarded. D. A thirteen digit number beginning with will be forwarded. Correct Answer: C Section: Dial Peer /Reference: 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 of xxxxx (5501 is prefixed with the command prefix 5501) QUESTION 9 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 Correct Answer: C Section: Dial Peer /Reference: 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 unassigned-number disconnect cause code from a destination router. QUESTION 10 On the basis of the provided 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

48 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. Exhibit: A. dial-peer voice 3 pots destination-pattern preference 2 B. dial-peer voice 3 pots destination-pattern preference firstlast C. dial-peer voice 3 pots destination-pattern preference 0 D. dial-peer voice 3 pots destination-pattern preference high Correct Answer: A Section: Dial Peer /Reference: The router considers lower preferences to be better than higher preferences and the default preference is 0. Therefore, by setting the preference of Bob's 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).

49 QUESTION 11 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.) Exhibit: A. Implement NBAR. B. Implement admission control. C. Mark voice traffic as EF in DSCP. D. Mark voice traffic highest priority in 802.1p. E. Use crtp to maximize bandwidth utilization. F. Configure access switches to trust traffic from IP phones. Correct Answer: BCE Section: VoIP Design Element /Reference: QUESTION 12 Refer to the exhibit. 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? Exhibit: A. proxy

50 B. redirect C. registrar D. user agent client E. user agent server Correct Answer: A Section: VoIP Design Element /Reference:

51 Exam J QUESTION 1 Refer to the exhibit. Lighthorse Equine Management would like to investigate converging voice and data on their existing T1 Frame Relay WAN link between New York and Atlanta. Currently the following list of applications are consuming no more bandwidth than what is listed on this segment of the network. T1 link 1536 kbps 75 kbps internet 200 kbps Oracle 500 kbps FTP 250 kbps Total 1025 kbps The customer has allocated 25% of the WAN link for routing updates and other overhead. They would like to increase the number of samples encapsulated in each PDU to 40 ms. You have calculated 6 bytes of overhead for Frame Relay, no crtp, and the use of the G.711 codec. How many simultaneous calls could be placed on this link? Exhibit: A. 0 calls B. 1 call C. 2 calls D. no more than 5 calls E. no more than 10 calls F. no more than 20 calls Correct Answer: B Section: VoIP Design Element /Reference: QUESTION 2 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.) Exhibit:

52 A. Implement NBAR. B. Implement admission control. C. Mark voice traffic as EF in DSCP. D. Mark voice traffic highest priority in 802.1p. E. Use crtp to maximize bandwidth utilization. F. Configure access switches to trust traffic from IP phones. Correct Answer: BCE Section: VoIP Design Element /Reference: QUESTION 3 Refer to the exhibit. Users are not able to complete a call from to What is the correct diagnosis for the problem? Exhibit:

53 A. incorrect destination-pattern in router 1 B. incorrect POTS dial-peer statement in router 2 C. incorrect session-target statement in router 2 D. incorrect port statement in router 1 pots dial peer E. missing no digit-strip on the voip dial peer in router 1 Correct Answer: A Section: VoIP Design Element /Reference: QUESTION 4 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?

54 A. dial peer B. voice port C. hunt group D. trunk group E. source IP group Correct Answer: D Section: VoIP Design Element /Reference: QUESTION 5 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 Correct Answer: B Section: VoIP Design Element /Reference: QUESTION 6 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 ^ rule 1 ^ rule 2 ^ rule 3 ^ rule 4 ^ rule 5 ^ rule 6 ^ rule 7 ^ rule 8 ^ rule 9 ^ A. test translation-rule The replaced number: B. test translation-rule The replaced number: C. test translation-rule The replaced number: D. test translation-rule The replaced number:

55 Correct Answer: A Section: VoIP Design Element /Reference: QUESTION 7 Which device is used to allow an H.323 stream to transit a firewall? A. gatekeeper B. gateway C. proxy D. MCU Correct Answer: C Section: VoIP Design Element /Reference: QUESTION 8 Refer to the exhibit. 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? Exhibit: A. proxy B. redirect C. registrar D. user agent client E. user agent server Correct Answer: A Section: VoIP Design Element /Reference:

56 QUESTION 9 Refer to the H.323 message in the exhibit. What is the gateway doing with the gatekeeper? Exhibit: A. initial registration B. full registration C. lightweight registration D. registration retry Correct Answer: C Section: Gatekeeper /Reference: QUESTION 10 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.)

57 A. zone prefix SJ 408 gw-priority 6 SJ1 B. zone prefix SJ 408 gw-priority 6 SJ2 C. zone prefix SJ 408 gw-priority 10 SJ1 D. zone prefix SJ 408 gw-priority 10 SJ2 E. zone prefix SJ 408 gw-priority 0 SJ2, 10 SJ1 F. zone prefix SJ 408 gw-priority 6 SJ1, 10 SJ2 Correct Answer: AD Section: Gatekeeper /Reference:

58 Exam K QUESTION 1 The SJ local zone contains a gatekeeper that controls two gateways, SJ1 and SJ2. Both gateways provide access to area code 408. Which two command strings should be entered into the gatekeeper to give the SJ2 gateway priority over the SJ1 gateway? (Choose two.) A. zone prefix SJ 408 gw-priority 6 SJ1 B. zone prefix SJ 408 gw-priority 6 SJ2 C. zone prefix SJ 408 gw-priority 10 SJ1 D. zone prefix SJ 408 gw-priority 10 SJ2 E. zone prefix SJ 408 gw-priority 0 SJ2, 10 SJ1 F. zone prefix SJ 408 gw-priority 6 SJ1, 10 SJ2 Correct Answer: AD Section: Gatekeeper /Reference: QUESTION 2 Refer to the H.323 message in the exhibit. What is the gateway doing with the gatekeeper? Exhibit:

59 A. initial registration B. full registration C. lightweight registration D. registration retry Correct Answer: C Section: Gatekeeper /Reference: QUESTION 3 In which three RAS messages is the technology prefix sent? (Choose three.) A. GRQ B. RRQ C. RCF D. IRR E. IRQ

60 Correct Answer: ABE Section: Gatekeeper /Reference: QUESTION 4 Refer to the output from the debug h225 asn1 command in the exhibit. You have configured a gatekeeper with two local zones, hq and br. You want the gateway at the branch location to register with zone BR. What needs to be corrected in the branch gateway to resolve the issue? A. Change the IP address in the h323-gateway voip id command. B. Change the gatekeeper-id in the h323-gateway voip id command. C. Add a zone remote for zone BR so the gateway can register with the correct zone. D. Change the gatekeeper-id and the IP address in the h323-gateway voip id command. Correct Answer: B Section: Gatekeeper /Reference: QUESTION 5 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? A. gatekeeper zone local GKVIA acme.com zone remote GK407 ITSP.com zone prefix GK no shutdown B. gatekeeper zone local GKVIA acme.com zone remote GK407 ITSP.com invia outvia GKVIA zone prefix GK * no shutdown C. gatekeeper zone local GKVIA acme.com zone remote GK407 ITSP.com invia GK407 outvia GK407 zone prefix GK no shutdown D. gatekeeper zone local GKVIA acme.com zone remote GK407 ITSP.com invia GKVIA outvia GKVIA zone prefix GK *

61 no shutdown Correct Answer: D Section: Gatekeeper /Reference: QUESTION 6 Refer to the exhibit 1and 2. You have configured a gatekeeper and an IP-IP gateway on the same router. When you look at the output from the show gatekeeper endpoint command, the IP-IP gateway is not registered with the gatekeeper. What needs to be configured to resolve this issue? 1 (exhibit):

62 2 (exhibit):

63 A. You need to stop and restart the gateway. B. You need to add a VoIP dial peer to the configuration. C. The h323-gateway voip id command has an incorrect IP address. D. The h323-gateway voip id command has an incorrect gatekeeper ID and IP address. Correct Answer: B Section: Gatekeeper /Reference: QUESTION 7 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 Correct Answer: AC Section: Gatekeeper

64 /Reference: QUESTION 8 The SJ local zone contains a gatekeeper that controls two gateways, SJ1 and SJ2. Both gateways provide access to area code 408. Which two command strings should be entered into the gatekeeper to give the SJ2 gateway priority over the SJ1 gateway? (Choose two.) A. zone prefix SJ 408 gw-priority 6 SJ1 B. zone prefix SJ 408 gw-priority 6 SJ2 C. zone prefix SJ 408 gw-priority 10 SJ1 D. zone prefix SJ 408 gw-priority 10 SJ2 E. zone prefix SJ 408 gw-priority 0 SJ2, 10 SJ1 F. zone prefix SJ 408 gw-priority 6 SJ1, 10 SJ2 Correct Answer: AD Section: Gatekeeper /Reference: QUESTION 9 You are a Acme network administrator, your new task is 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? A. gatekeeper zone local GKVIA acme.com zone remote GK407 ITSP.com zone prefix GK no shutdown B. gatekeeper zone local GKVIA acme.com zone remote GK407 ITSP.com invia outvia GKVIA zone prefix GK * no shutdown C. gatekeeper zone local GKVIA acme.com zone remote GK407 ITSP.com invia GK407 outvia GK407 zone prefix GK no shutdown D. gatekeeper zone local GKVIA acme.com zone remote GK407 ITSP.com invia GKVIA outvia GKVIA zone prefix GK * no shutdown Correct Answer: D Section: Gatekeeper /Reference: QUESTION 10 You are the director of the Acme VoIP network, based on the exhibit. You have a client that is testing a

65 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.) Exhibit: A. replace zone local GK-A acme.com with zone remote GK-A acme.com B. replace zone local DGK acme.com with zone remote DGK acme.com C. replace zone prefix GK-B with zone prefix GK-B D. replace zone prefix GK-A

66 with zone prefix GK-A Correct Answer: AD Section: Gatekeeper /Reference: QUESTION 11 Click and drag the type of call on the above to the type of voice port it applies to on the below. Select and Place: Correct Answer:

67 Section: Drag and Drop /Reference: 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

68 Exam L QUESTION 1 The proper call-signaling term to the correct box in the diagram to establish RSVF-based Call Admission Control between the two Cisco Unifield Border Elements: Cisco UBEs. Some option is may be user more than once. Select and Place: Correct Answer:

69 Section: Drag and Drop /Reference: Here is how the call is established with RSVF-based Call Admission Control 1) The Cisco Unified Communications Manager (at the left-side) 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 2 Click and drag the type of call on the above to the type of voice port it applies to on the below. Select and Place:

70 Correct Answer: Section: Drag and Drop /Reference: 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

71 QUESTION 3 Assume a SIP voice network. Drag each characteristic to the type of SIP call setup the characteristics best describes. Select and Place: Correct Answer:

72 Section: Drag and Drop /Reference: 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 4 Refer to the exhibit. Which configuration option will allow communication between a voice-enabled router and a PBX? Exhibit:

73 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 Correct Answer: A Section: Analog Voice Port /Reference: QUESTION 5 Which dial plan characteristic shows the most obvious improvement by dropping a number translation step? A. availability B. post-dial delay C. scalability D. hierarchical design Correct Answer: B Section: Dial Peer /Reference: 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. By dropping a number translation step, the post-dial delay time will be obvious improvement.