Collection. Passing Score: 790 Time Limit: 120 min File Version: Cisco

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1 Collection Passing Score: 790 Time Limit: 120 min File Version: Cisco Implementing Cisco Unified Communications Voice over IP and QoS v8.0 (CVOICE v8.0) Practice Test Version: 5.02 Cisco : Practice Exam with explanation. Sections 1. Describe the need to implement QOS for voice and video 2. Describe and configure the Diffserv QOS Model 3. Implement cisco unified Border Element 4. Implement a gateway 5. Describe components of a gateway 6. Implement CUCME to support endpoints using CLI 7. Describe the basic operation and components involved in a voip call 8. Describe a dial plan

2 Exam A QUESTION 1 What is the function of class-based marking? A. Marking packets based on CoS value, IP precedence value, or DSCP value allows Layer 3 frames to be identified and distinguished from other packets. B. Marking frames based only on CoS value or IP precedence value allows Layer 2 frames to be identified and distinguished from other frames. C. Marking packets or frames sets information in the Layer 2 and Layer 3 headers of a packet so that the packet or frame can be identified and distinguished from other packets or frames. D. Marking frames only sets information in the Layer 2 headers of a frame so that the frame can be identified and distinguished from other packets or frames. Correct Answer: C Section: Describe the need to implement QOS for voice and video /Reference: : The Class-Based Packet Marking feature provides users with a user-friendly command-line interface (CLI) for efficient packet marking by which users can differentiate packets based on the designated markings. CoS is classification of specific traffic by manipulating the class of service bits in the Ethernet frame header whereas IP Precedence and DSCP is configured by changing the TOS Field in IP Header frame. Marking a packet with an IP precedence or IP DSCP marking allows users to classify traffic based on an IP precedence or IP DSCP value, depending on which value is marked. These marking can be used to identify traffic within the network, and other interfaces can match traffic based on the IP Precedence or DSCP markings. Marking a packet with a local CoS value allows users to associate a Layer 2 Class of Service value with a packet. The value can then be used to classify packets based on user-defined requirements. Layer 2 to Layer 3 mapping can also be configured by matching on the CoS value, since switches already have the capability to match and set CoS values. QUESTION 2 Voice packets are arriving at a destination with a variance of between 20 and 50 ms. If the jitter buffer has a capacity of 30 ms, what is the impact on the audio at the receiving IP phone? A. The jitter buffer will replay the previous voice packets to replace those packets that exceed 30 msto avoid speech gaps. B. The audio stream at the receiving IP phone will be delayed and garbled. C. The DSP will automatically increase the jitter buffer size after sampling the range of incoming voice packets to accommodate the wider range in variation of voice packet arrival times to avoid voice gaps. D. The IP phone will negotiate, in mid-call, a lower bandwidth codec to reduce the delay in the arrival of voice packets to avoid voice gaps. Correct Answer: B Section: Describe the basic operation and components involved in a voip call /Reference: :

3 By default, De-Jitter buffer runs in an adaptive mode where it dynamically adjusts to the amount of jitter present up to a point. The DSP algorithms in the codec take samples throughout the voice call and adjust the value of the average delay as network jitter conditions change. The size of the jitter buffer is adjusted upward or downward as needed to ensure smooth transmission of voice frames to the codec. If the jitter is so large that it causes packets to be received out of the range of this buffer, the out-of-range packets are discarded and dropouts are heard in the audio. For losses as small as one packet, the DSP interpolates what it thinks the audio should be and no problem is audible. When jitter exceeds what the DSP can do to make up for the missing packets, audio problems are heard. QUESTION 3 Refer to the exhibit. Which type of ephone-dn is configured for the two ephones that are shown? Exhibit: A. single-line-octo B. hunt line C. shared-line, nonexclusive D. two directory numbers with one telephone number E. shared-line, overlay F. Octo-line Correct Answer: E Section: Describe the basic operation and components involved in a voip call /Reference: : The above exhibit shows the configuration for a simple shared-line overlay set. The primary ephone-dn that is configured for each phone is unique while the remaining ephone-dns 10, 11, and 12 are shared in the overlay set on both phones. The primary ephone-dn in a shared- line overlay set is configured unique to the phone to guarantee that the phone has a line available for outgoing calls, and to ensure that the phone user can obtain dial-tone even when there are no idle lines available in the rest of the shared-line overlay set. Using a unique ephone-dn also provides a unique calling party identity on outbound calls made by the phone so that the called user can see which specific phone is calling. l#wp QUESTION 4 Refer to the exhibit.

4 Cisco Unified Communications Manager Express has been partially configured to support 6 IP phones and 12 directory numbers. The Cisco Unified Communications Manager Express will use the IP address /24. Which two elements of the configuration are missing from the command output and need to be added so that phones do not auto-register, but can manually register with Cisco Unified Communications Manager Express? (Choose two.) Exhibit: A. ip address B. no reg-ephone C. create profile D. ip source-address E. create cnf-files F. no auto-reg-ephone Correct Answer: DF Section: Describe the basic operation and components involved in a voip call /Reference: : To identify the IP address and port through which IP phones communicate with a CiscoUnifiedCME router, use the ip source-address command in telephony-service or group configuration mode. This command enables a router to receive messages from CiscoUnifiedIPphones through the specified IP address and port. The CiscoUnifiedCME router cannot communicate with CiscoUnifiedCME phones if the IP address of the port to which they are attached is not configured. Normally when you configure basic telephony-service parameters, then phone can register with CME although no DN will be assigned to them. You can disable this by using the no auto-reg- ephone command. After this command the phone which will try to register will receive message "Registration Rejected: No configuration entry...".. When automatic registration is blocked, CiscoUnifiedCME records the MAC addresses of phones that attempt to register but cannot because they are blocked In config, ip source-address was missing, CUCME will not work without "ip source-address". For manually register with CUCME we need to use "no auto-reg-ephone". QUESTION 5 How does LLQ ensure that voice traffic is always expedited? A. LLQ adds a strict priority class to CBWFQ. This class allows delay-sensitive data such as voice to be dequeued and sent first. B. LLQ uses CBWFQ to prioritize voice traffic and dequeue the voice packets so that they can be handled first. C. The strict priority queue has a higher weight than the queues in CBWFQ. This weight allows the delaysensitive data such as voice to be dequeued and sent first.

5 D. The LLQ strict priority queue allows delay-sensitive data such as voice to be dequeued and sent first (before packets in other queues are dequeued), giving delay-sensitive data preferential treatment over other traffic. Correct Answer: D Section: Describe the need to implement QOS for voice and video /Reference: : Without Low Latency Queueing, CBWFQ provides weighted fair queueing based on defined classes with no strict priority queue available for real-time traffic. This scheme poses problems for voice traffic that is largely intolerant of delay, especially variation in delay. For voice traffic, variations in delay introduce irregularities of transmission manifesting as jitter in the heard conversation. The Low Latency Queueing feature provides strict priority queueing for CBWFQ, reducing jitter in voice conversations. Configured by the priority command, Low Latency Queueing enables use of a single, strict priority queue within CBWFQ at the class level, allowing you to direct traffic belonging to a class to the CBWFQ strict priority queue. QUESTION 6 Your PSTN carrier sends digits to your T1 PRI circuit in a digit-by-digit format. How must the T1 PRI circuit be configured to support this capability? A. The T1 PRI controller supports either en-bloc or digit-by-digit formats natively. B. The serial interface that is associated with the T1 controller needs to include the isdn incoming- voice command. C. The T1 controller needs to include the isdn overlap-receiving command. D. The serial interface that is associated with the T1/E1 controller needs to include the isdn overlap-receiving command. Correct Answer: D Section: Implement a gateway /Reference: : Configuring Overlap-receiving on the D-channel changes the way routers behave when receiving ISDN calls. Overlap receiving allows the matching of dial peers as the digits are being received. The router responds to the setup message with a SETUP ACK. This informs the network that it is ready to receive further information messages containing additional call routing elements. Reference: QUESTION 7 You have a Cisco Unified Border Element configured to provide H.323 to SIP interworking. Which command

6 will verify that you have a single H.323 and a single SIP call leg when the call is placed? A. show call active voice B. debug voip ipipgw C. show dialpeer voice D. debug voice dialpeer Correct Answer: A Section: Implement cisco unified Border Element /Reference: : The show call active voice command allows you to display the contents of the active call table. The show call active voice command displays data from the plain old telephone service (POTS) and VoIP call legs on the voice gateway. The information presented includes call times, dial peers, connections, quality of service parameters, and gateway handling of jitter. This information can be useful when you troubleshoot a range of voice quality problems.

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8 To display call information for voice calls in progress, use the show call active voice command in user EXEC or privileged EXEC mode. QUESTION 8 Which QoS technology provides a strict priority queuing scheme that allows delay-sensitive data such as voice to be dequeued and sent before packets in other queues are dequeued, and also works with WFQ and CBWFQ. A. header compression B. IP RTP Priority and Frame Relay IP RTP Priority C. RSVP D. low latency queuing E. FRF.12 Correct Answer: D Section: Describe the need to implement QOS for voice and video /Reference: : The Low Latency Queuing feature brings strict priority queuing to Class-Based Weighted Fair Queuing (CBWFQ). Strict priority queuing allows delay-sensitive data such as voice to be dequeued and sent first (before packets in other queues are dequeued), giving delay- sensitive data preferential treatment over other traffic. QUESTION 9 What is the decimal equivalent of the DSCP value AF21? A. 16 B. 17 C. 18 D. 21 Correct Answer: C Section: Describe the basic operation and components involved in a voip call

9 /Reference: : Assured Forwarding (AF) is a means to offer different levels of forwarding assurances for IP packets. Four AF classes are defined, where each AF class is in each DS node allocated a certain amount of forwarding resources(buffer space and bandwidth). Within each AF class IP packets are marked with one of three possible drop precedence values. A congested node tries to protect packets with a lower drop precedence value from being lost by preferably discarding packets with a higher drop precedence value. Classes 1 to 4 are referred to as AF classes. The following table illustrates the DSCP coding for specifying the AF class with the probability. Bits DS5, DS4 and DS3 define the class; bits DS2 and DS1 specify the drop probability; bit DS0 is always zero. Reference: qos_6dscp_val.pdf QUESTION 10 Refer to the exhibit. Drag the appropirate IOS command from the left and drop them in the spaces on the right in order to configure Cisco Unified Border Element. The ITSP does not support early offer. Not all boxes are used. Exhibit:

10 Select and Place: Correct Answer:

11 Section: Implement cisco unified Border Element /Reference: QUESTION 11 Refer to the exhibit. Drag the signaling methods from the left and drop them in the correct position in the graphic on the right. Some method are used more than once, and some method may not be used at all. Select and Place: Correct Answer:

12 Section: Implement a gateway /Reference: QUESTION 12 A new Cisco 7965 IP phone is installed on a Cisco Unified Communications Manager Express system. When the phone requests the.loads file from the TFTP server, it sees that the versions are different. What does the IP phone do to resolve this issue? A. The IP phone requests the SEP<mac>.cfg file and reboots. B. The IP phone attempts to obtain the new firmware file image from the TFTP server. C. The IP phone boot requests the XMLDefault.cnf.xml file and boots up. D. The IP phone does not boot up and will require manual intervention to factory reset the phone before a new firmware image can be downloaded. Correct Answer: B Section: Describe the basic operation and components involved in a voip call /Reference: : Cisco IP Phone Initialization Process: 1. At initialization, the Cisco IP phone sends a request to the DHCP server to get an IP address, DNS server

13 address, and TFTP server name or address, if appropriate. Options are set in DHCP server (Option 066, Option 150, and so on). It also gets a default gateway address if set in DHCP server (Option 003). 2. If a DNS name of the TFTP sever is sent by DHCP, then a DNS sever IP address is required to map the name to an IP address. This step is bypassed if the DHCP server sends the IP address of the TFTP server. In this case study, the DHCP server sent the IP address of TFTP because DNS was not configured. 3. If a TFTP server name is not included in the DHCP reply, then the Cisco IP phone uses the default server name. 4. The configuration file (.cnf) file is retrieved from the TFTP server. All.cnf files have the name SEP<mac_address>.cnf, where "SEP" is an acronym for Selsius Ethernet Phone. If this is the first time the phone is registering with the Cisco CallManager, then a default file, SEPdefault.cnf, is downloaded to the Cisco IP phone. 5. All.cnf files include the IP address(es) of the primary and secondary Cisco CallManager(s). The Cisco IP phone uses the IP address to contact the primary Cisco CallManager and register. 6.Once the Cisco IP phone has connected and registered with Cisco CallManager, the Cisco CallManager tells the Cisco IP phone which executable version (called a load ID) to run. If the specified version does not match the executing version on the Cisco IP phone, the Cisco IP phone will request the new executable from the TFTP server and reset automatically. QUESTION 13 You are configuring a network to support voice to the PSTN. One important aspect to the configuration is to be able to determine the individual slot, subunit, and port number from the gateway endpoint identifier. Which signaling protocol is appropriate for this situation? A. H.323 B. SIP C. SCCP D. MGCP Correct Answer: D Section: Describe a dial plan /Reference: : Endpoints are any of the voice ports on the designated gateway. These voice ports provide connectivity to both analog ports and digital trunks to the PSTN. Ports on gateways are identified by endpoints in very specific ways. It is important to note that gateways can have multiple endpoints dependent on the number of ports it contains, and that the endpoints are case insensitive. A sample MGCP endpoint addressing scheme is provided below. QUESTION 14 The voice gateway selects an inbound VoIP dial peer by matching the information elements in the message with the dial-peer attributes. From the list on the left, drag the elements to the right and drop them in the order in witch a voice gateway

14 matches inbound calls. Not all options are used. Select and Place: Correct Answer: Section: Describe components of a gateway /Reference:

15 QUESTION 15 How many IP phone calls can be sent across a 64-kb/s Frame Relay link that uses the G.729 codec? The sampling rate is 50 times a second, with 20 bytes per sample. There are 8 bytes of Frame Relay header overhead with no checksum, and header compression is used. A. 3 B. 4 C. 5 D. 7 Correct Answer: C Section: Describe the basic operation and components involved in a voip call

16 /Reference: : Bandwidth Calculation FormulasThese calculations are used: QUESTION 16 When Cisco Unified Border Element is configured to support RSVP-based CAC, at which point during call setup are the RSVP path and reservation messages sent and received? A. The path message is sent immediately after the call setup message is received and the reservation message is received after H.245 capabilities negotiation is completed. B. The reservation message is sent immediately after the call setup message is received and the path message is received after H.225 call setup messages have been sent. C. The path and reservation messages are sent and received after the H.245 capabilities negotiation is completed. D. The path and reservation messages are sent and received immediately after the call setup message is received. Correct Answer: D Section: Implement cisco unified Border Element /Reference: : The H.323 setup is suspended before the destination phone, triggered by the H.225 alerting message, starts ringing. The RSVP reservation is made in both directions because a voice call requires a two-way speech path and therefore bandwidth in both directions. The terminating gateway ultimately makes the CAC decision based on whether or not both reservations succeed. At that point the H.323 state machine continues either with an H.225 Alerting/Connect (the call is allowed and proceeds), or with an H.225 Reject/Release (call is denied). The RSVP reservation is in place by the time the destination phone starts ringing and the caller hears ringback.

17 QUESTION 17 Which command should be included in order to trust the DSCP-marked traffic from the distribution layer? A. mis qos trust cos B. mis trust dscp-cos C. mis qos trust dscp D. mis qos trust dscp-cos Correct Answer: C Section: Implement CUCME to support endpoints using CLI /Reference: : To configure the multilayer switching quality of service port trust state and to classify traffic by examining differentiated services code point (DSCP) value, use the mls qos trust dscp command in interface configuration mode. This will enable the device to trust incoming packets that have DSCP values (the most significant 6 bits of the 8-bit service-type field). QUESTION 18 What are the PHBs that DiffServ use? A. resource reservation and admission control B. default, AF, and EF PHBs C. AF, EF, and CS PHBs D. AF and EF PHBs E. default, AF, EF, and CS PHBs

18 Correct Answer: E Section: Describe and configure the Diffserv QOS Model /Reference: : A Per Hop Behavior refers to the packet scheduling, queuing, policing, or shaping behavior of a node on any given packet belonging to a Behavior Aggregate, and as configured by a Service Level Agreement (SLA) or policy. To date, four standard PHBs are available to construct a DiffServ-enabled network and achieve coarse-grained, end-to-end CoS and QoS: The Default PHB, Class-Selector PHBs, Expedited Forwarding PHB and Assured Forwarding PHB. 2f_ps6610_Products_White_Paper.html QUESTION 19 Drag the delay type on the left and drop it on the correct description on the right. Select and Place:

19 Correct Answer: Section: Describe the basic operation and components involved in a voip call /Reference: QUESTION 20 Refer to the exhibit. Your companyatms QoS policy states that all traffic that is arriving at access layer switches from IP phones should be marked with a DSCP value of 46 and that all untagged traffic that is arriving from a PC that is attached to an IP phone should be marked with a CoS value of 1. Which two options will satisfy the requirements for the CoS-to-DSCP map and are the correct QoS commands? (Choose two.)

20 Exhibit: A. mis qos 1 B. mis qos map cos-dscp C. mis qos cos 1 D. mis qos map dscp E. mis qos map cos F. mis qos dscp 1 Correct Answer: BC Section: Describe the need to implement QOS for voice and video /Reference: : To define the ingress Class of Service (CoS)-to-differentiated services code point (DSCP) map for trusted interfaces, use the mls qos map cos-dscp command in global configuration mode. mls qos map cos-dscp dscp1...dscp8 dscp1...dscp8 - Defines the CoS-to-DSCP map. For dscp1...dscp8, enter eight DSCP values that correspond to CoS values 0to 7. Separate consecutive DSCP values from each other with a space. The supported DSCP values are 0, 8, 10, 16, 18, 24, 26, 32, 34, 40, 46, 48, and 56. To define the default multilayer switching (MLS) class of service (CoS) value of a port or to assign the default CoS value to all incoming packets on the port, use the mls qos cos command in interface configuration mode. mls qos cos cos-value cos-value - Assigns a default CoS value to a port. If the port is CoS trusted and packets are untagged, the default CoS value is used to select one output queue as an index into the CoS-to-DSCP map. The CoS range is 0 to 7. The default is 0. QUESTION 21

21 What is the reason that an outgoing call succeeds when there is no COR list that is applied to the incoming dial peer and a COR list is applied to the outgoing dial peer? A. The COR list for incoming calls on the incoming dial peer is a superset of COR lists for outgoing calls on the outgoing dial peer. B. COR lists for incoming calls on the incoming dial peer are not a superset of COR lists for outgoing calls on the outgoing dial peer. C. The outgoing dial peer, by default, has the lowest priority. D. The incoming dial peer, by default, has the highest COR priority when no COR is applied. Correct Answer: D Section: Describe a dial plan /Reference: : By default, an incoming call leg has the highest COR priority and the outgoing COR list has the lowest COR priority. This means that if there is no COR configuration for incoming calls on a dial-peer, then you can make a call from this dial-peer (a phone attached to this dial-peer) going out of any other dial-peer, irrespective of the COR configuration on that dial-peer shtml QUESTION 22 Refer to the exhibit. When an inbound PSTN call from arrives at the ISDN port that is shown in the exhibit, which dial peer will be matched for the inbound leg? Exhibit: A. Dial-peer 123, because destination-pattern takes precedence over answer-address. B. Dial-peer 2123, because answer-address takes precedence over destination-pattern. C. The matching inbound dial peer will be selected at random. D. Although dial-peer 2123 takes precedence, it will not be matched because the command direct- inward-dial

22 is missing. E. Dial-peer 123 will be matched because dial-peer 2123 will strip all the digits. Correct Answer: B Section: Describe a dial plan /Reference: : The inbound call will first try to match the with the incoming called-number command. We can also use `answer-address command' which is searched if `incoming called- number' is not present. And if there is no `incoming called-number command' and `answer-address command', then the gateway will hunt for dialpeer with destination-pattern of calling party number. QUESTION 23 Calls are failing to egress the local PSTN gateway that uses an E1 PRI circuit. Which debug command would be most useful in determining which dialed digits are being sent to the PSTN? A. debug voice dial-peer B. debug isdn q921 C. debug isdn q931 D. ccapi inout Correct Answer: C Section: Describe the basic operation and components involved in a voip call /Reference: : Debug isdn q931 command to display information about call setup and teardown of ISDN network connections (Layer 3).In order to verify the layer 3 signaling we need to enable layer 3 signaling command. ISDN q921 is for layer2. Debug isdn q931 shows the calling number and called number. If the calls are failing, we can also see the ISDN cause codes from the debug isdn q931 command. QUESTION 24 Refer to the exhibit. An administrator is migrating a PBX telephony system to an IP Phone solution using a fixed numbering plan. The extension numbers and PSTN DIDs cannot be changed. Which of the following methods can be used in order to reach the individual extensions at Site B when called via the PSTN? Exhibit:

23 A. The administrator can add a 1 to the DID for Site B to become xxx. B. The administrator needs to map the last four digits in the DID to the extension numbers and prefix a site code. C. The administrator needs to map the last four digits in the DID to the extension numbers and prefix an intersite code. D. The administrator needs to map the last four digits in the DID to the extension numbers using translation rules. E. No changes are necessary because PSTN calls are preceded with access code 9 Correct Answer: D Section: Describe a dial plan /Reference: : Since the extension and PSTN DID is one and the same for the customer, no manipulation is required the Route Plan to reach individual extensions from PSTN DID QUESTION 25 Drag the components that make up Cisco Fax Relay and T.38 from the left and drop them under the appropriate category on the right. Select and Place:

24 Correct Answer: Section: Describe the basic operation and components involved in a voip call /Reference: QUESTION 26 The router with the IP address of needs to be configured to use the device as the clock source. Which configuration command will accomplish this task? A. clock source B. ntp server C. clock set D. ntp source ip addr E. ntp client server

25 Correct Answer: B Section: Implement CUCME to support endpoints using CLI /Reference: : To configure your routers to use a NTP server for time synchronization, the command ntp server, followed by the IP address or hostname of the NTP server, is used. To specify additional timeservers for redundancy, simply repeat the ntp server command with the IP address of each additional server. QUESTION 27 Which two functions are associated with a voice gateway? (Choose two.) A. switches voice channels between connected analog and digital voice circuits B. provides voice-messaging services to connected analog and digital voice circuits C. interconnects two logically separate VoIP networks D. negotiates endpoint capabilities E. controls opening and closing of logical channels that are used to carry media streams Correct Answer: AE Section: Describe components of a gateway /Reference: : The basic function of a gateway is to translate between different types of networks. In a VoIP environment, voice gateways are the interface between a VoIP network and the public switched telephone network (PSTN), a private branch exchange (PBX), or analog devices such as fax machines. In its simplest form, a voice gateway has an IP interface and a legacy telephone interface, and it handles the many tasks involved in translating between transmission formats and protocols. The gateway allows communication between the two networks by performing tasks such as Interfacing with the IP network and the PSTN or PBX. Supporting IP call control protocols, Performing call setup and teardown for calls between the VoIP and PSTN networks by terminating and reoriginating the call media and signaling, Providing supplementary services, such as call hold and transfer, Relaying dual tone multifrequency (DTMF) tones, Supporting analog fax and modems over the IP network. 2.pdf QUESTION 28 Refer to the exhibit. When an inbound PSTN call to is received by the router that is shown in the exhibit, what is the resulting called number? Exhibit:

26 A B C D E. 4Q Correct Answer: D Section: Describe a dial plan /Reference: : /^.*\(...$\) Truncates Numbers down to the last 4 digits. QUESTION 29 Which three functions are associated with MGCP? (Choose three.) A. Control is implemented by a series of plain-text commands that are sent over UDP port 2427 between Cisco Unified Communications Manager and the gateway. B. A PRI backhaul channel forwards PRI Layer 2 (0.921) signaling information via a TCP connection from the gateway to the call agent. C. MGCP uses a separate channel for backhauling signaling information between the call agent and the gateway. D. The gateway maintains a separate dial plan for redundancy in case the call agent fails. E. Users query the call agent to determine the location of the call recipient. F. A call agent uses control messages to direct its gateways and their operational behavior. Correct Answer: ACF Section: Describe components of a gateway /Reference: :

27 MGCP is a plain-text protocol used by call-control devices to manage IP Telephony gateways. MGCP is a master/slave protocol that allows a call control device to take control of a specific port on a gateway. With this protocol, the Cisco CallManager knows and controls the state of each individual port on the gateway. It allows complete control of the dial plan from Cisco CallManager, and gives CallManager per-port control of connections to the PSTN, legacy PBX, voice mail systems, POTS phones and so forth. This is implemented with the use of a series of plain-text commands sent over User Datagram Protocol (UDP) port 2427 between the Cisco CallManager and the gateway. Another concept relevant to the MGCP implementation with Cisco CallManager is PRI Backhaul. This occurs when Cisco CallManager takes control of the Q.931 signaling data used on an ISDN PRI. The one thing that distinguishes a PRI from other interfaces is the fact that the data that is received from the PSTN on the D-channel and needs to be carried in its raw form back to the Cisco CallManager to be processed. The gateway does not process or change this signaling data, it simply passes it onto the Cisco CallManager through TCP port The gateway is still responsible for the termination of the Layer 2 data. That means that all the Q.921 data-link layer connection protocols are terminated on the gateway, but everything above that (Q.931 network layer data and beyond) is passed onto the Cisco CallManager. This also means that the gateway does not bring up the D-channel unless it can communicate with Cisco CallManager to backhaul the Q.931 messages contained in the D-channel. QUESTION 30 Drag the function that are associated with H.245 from the list on the left ot the boxes on the right. Select and Place: Correct Answer:

28 Section: Implement a gateway /Reference: QUESTION 31 Refer to the exhibit. Callers dial 0 to reach an outside line. When they try to place calls to directory services (322) or services (422), they hear the reorder tone. What needs to be edited in the dial peer to allow these calls to complete successfully? Exhibit: A. The destination pattern is incorrect. It needs to start with a 9. B. A "prefix 1" statement needs to be added to the dial-peer configuration. C. The forward-dig its all command needs to be applied to the dial peer. D. The destination pattern needs to be edited so that the first digit that is matched is a 0. E. The destination pattern needs to be edited so that the first digit that is matched is a 0 and the forward-digits all command needs to be added to the dial peer. F. The destination pattern needs to be edited so that the first digit that is matched is a 1 and the forward-digits all command needs to be added to the dial peer. G. The destination pattern needs to be edited so that the first digit that is matched is a 0 and the forward-digits 3 command needs to be added to the dial peer. Correct Answer: G Section: Describe the basic operation and components involved in a voip call /Reference: :

29 Since the callers dial 0 before any actual number to go outside line, they should have a destination pattern starting with 0 to place a successful call to directory services or other services. The forward-digits command controls the number of digits that are stripped before the dialed string is passed to the telephony interface. On outbound POTS dial peers, the terminating router normally strips off all digits that explicitly match the destination pattern in the terminating POTS dial peer. Only digits matched by the wildcard pattern are forwarded. The forward-digits command can be used to forward a fixed number of dialed digits, or all dialed digits, regardless of the number of digits that explicitly match the destination pattern. QUESTION 32 In which way does SIP Early Media differ from SIP Delayed Offer? A. In SIP Early Media mode, the SDP media capabilities are exchanged in the INVITE and the 200 OK messages. B. SIP Early Media uses session indicator 183. C. The INVITE message includes an SDP-formatted list of media capabilities (including codecs) that are supported by the originating UA. D. The 200 OK message carries an SDP message with the final media description that has been selected by the terminating UA based on the received list and the locally supported options. Correct Answer: B Section: Describe components of a gateway /Reference: : Early Media is the ability of two user agents to communicate before a call is actually established. Support for early media is important both for interoperability with the Public Switched Telephone Network (PSTN) and billing purposes. Early Media allows the sending of media from the called party or an application server to the caller, prior to the call being accepted. This media is generally sent from the PSTN such as ringing tone or announcements. Current implementations support early media through the 183 response code. When the called party wishes to send early media to the caller, it sends a 183 response to the caller. This response contains the Session Description Protocol (SDP). When the caller receives the response, it suppresses any local alerting of the user (for example, audible ring tones or a pop-up window) and begins playing out the media that it receives. The SDP in the 183 response provides an address, to which the real- time control protocol (RTCP) packets can be sent. QUESTION 33 Refer to the exhibit. Which class is always present even though it is not in the configuration snip? Exhibit:

30 A. class best-effort B. class class-default C. default class D. best-effort class E. class class-scavenger Correct Answer: B Section: Implement a gateway /Reference: : The class-default is in every policy-map by default and it cannot be removed. The class-default class is used to classify traffic that does not fall into one of the defined classes. Once a packet is classified, all of the standard mechanisms that can be used to differentiate service among the classes apply. The class-default class was predefined when you created the policy map, but you must configure it. If no default class is configured, then by default the traffic that does not match any of the configured classes is flow classified and given best-effort treatment.

31 QUESTION 34 A small office needs to provide outbound dialing and inbound DID without the cost of a T1 circuit. All signaling is loop-start. Which analog port configuration will support these requirements? A. voice-port 0/0/0/ description fxs-did signal did loop-start i voice-port 0/1/0 description fxo signal loop-start i dial-peer voice 1 pots incoming called-number. direct-inward-dial port 0/0/0 i dial-peer voice 90 pots destination-pattern 9T port 0/1/0 B. voice-port 0/0/0 signal loop-start i voice-port 0/1/0 signal loop-start i dial-peer voice 1 pots incoming called-numbert direct-inward-dial! dial-peer voice 90 pots destination-pattern 9T port 0/1/0 C. voice-port 0/1/0 signal did loop-start! " dial-peer voice 1 pots incoming called-number! dial-peer voice 90 pot destination-pattern 9T port 0/1/0 D. voice-port 0/0/0 signal did loop-start! " dial-peer voice 1 pots incoming called-number direct-inward-dial i dial-peer voice 90 pots destination-pattern 9T port 0/0/0 Correct Answer: B Section: Describe the basic operation and components involved in a voip call /Reference:

32 : To configure loop-start in an analog port, first enter into voice-port configuration mode and enter the command: signal loop-start. vp_cfg_analog_vps_ps6350_tsd_products_configuration_guide_chapter.html#wp QUESTION 35 In which situation would an administrator configure telephony services, but not configure any individual ephones? A. Phones that are controlled by Cisco Unified Communications Manager Express B. Cisco Unified Communications Manager SRST fallback C. Cisco Unified Communications Manager Express with HSRP D. Remotely located phones that are controlled by a third-party PBX E. This is not a valid scenario. Ephones are always required. Correct Answer: B Section: Implement CUCME to support endpoints using CLI /Reference: : When a phone registers for SRST service with a Cisco Router and the router discovers that the phone was configured with a specific extension number, the router searches for an existing prebuilt ephone-dn with that extension number and then assigns that ephone-dn number to the phone. If there is no prebuilt ephone-dn with that extension number, the system automatically creates one. In this way, extensions without prebuilt configurations are automatically populated with extension numbers and features as the numbers and features are "learned" by the Cisco router in SRST mode when the phone registers to the router after a WAN link fails. QUESTION 36 Drag the statement from the left to the protocol name that is associated with it on the right. Select and Place:

33 Correct Answer: Section: Describe the basic operation and components involved in a voip call /Reference: QUESTION 37 When configuring Cisco AutoQoS VoIP on a Cisco catalyst switch, how is the configuration performed? A. The auto qos voip command is applied to each interface.

34 B. The auto qos voip command is applied globally in the switch. C. Each interface will need either the auto qos voip cisco-phone or auto qos voip trust on each interface, depending on the upstream device D. Each interface will need either the auto qos voip trust cisco-phone or auto qos voip trust trust on each interface, depending on the upstream device. Correct Answer: C Section: Describe the need to implement QOS for voice and video /Reference: : The QoS mechanisms on a Catalyst switch differ from those QoS mechanisms found on a router. For example, while a router uses LLQ as a priority queuing strategy, a Catalyst switch might use weighted round-robin (WRR) as a priority queuing strategy. Fortunately, the AutoQoS feature available on some Catalyst switch models applies voice-specific QoS features globally to a Catalyst switch and also at the port level. To configure AutoQoS on supported Catalyst switch platforms, issue the following command from interface configuration mode: Switch(config-if)#auto qos voip [trust cisco-phone] If the trust option is used in the previous command, the Catalyst switch makes queuing decisions based on Layer 2 Class of Service (CoS) markings. However, if the cisco-phone option is used, the Catalyst switch makes queuing decisions based on CoS markings originating from a Cisco IP phone. The switch detects the presence of a Cisco IP phone via the CDP. QUESTION 38 Which types of voice ports allow a small office to provide outbound DNIS and inbound DID? A. FXS and FXO B. FXO and E&M C. FXS and FXS-DID D. FXS and E&M E. FXS-DID and FXO Correct Answer: E Section: Describe components of a gateway /Reference: : An FXO trunk is one of the simplest analog trunks available. Because Dialed Number Information Service (DNIS) information can only be sent out to the PSTN, no direct inward dialing (DID) is possible. ANI is supported for inbound calls. Two signaling types exist, loopstart and groundstart, with groundstart being the preferred method. An FXS DID trunk can receive only inbound calls, thus a combination of FXS DID, and FXO ports is required for inbound and outbound calls

35 QUESTION 39 Refer to the exhibit. Drag the appropriate IOS commands from the left and drop them in the space on the right in order to configure thre dial peer for the Cisco Unified Border Element. The IP WAN connection between the Cisco Unified Border Elements uses SIP. Not all commands and spaces are used. Exhibit: Select and Place:

36 Correct Answer: Section: Implement cisco unified Border Element /Reference: Please beaware of signaling protocol.if signaling is not the same on incoming and outgoing, we must use media flow-around instat of media flow-through.

37 QUESTION 40 When a Cisco 7965 IP phone downloads an SCCP firmware file package, which file is downloaded first by the IP phone from the TFTP server to describe the files it should request from the TFTP server? A. term45.default.loads B. cnu sbn C. term65.default.loads D. dsp sbn E. apps es2.sbn F. SCCP SR1S.loads G. dsp sbn H. jar45sccp sbn I. cvm45sccp sbn Correct Answer: C Section: Describe the basic operation and components involved in a voip call /Reference: :

38 TermXX.default.loads is the "firmware master list" file that the phone searches when it has been restored to factory configuration. This loads file is essentially just a packing list showing all the OS and application files the phone needs to function. The files include a cnu, cvm, dsp, app and jar files. The files listed in the termxx.default.loads file should be loaded into the TFTP Server for the phone to download. ("xx" will be either "45" for the CP-7945G model, or "65" for the CP-7965G model.) QUESTION 41 Which two statements are true regarding SCCP? (Choose two.) A. SCCP requires each endpoint or gateway event to be communicated to Cisco Unified Communications Manager B. Endpoints can operate autonomously if communication with Cisco Unified Communications Manager is lost. C. SCCP may interoperate with H.323 endpoints if it is implemented with Cisco Unified Communications Manager. D. Endpoints and gateways maintain the dial plan. E. SCCP uses hex messages for communication. Correct Answer: AC Section: Describe components of a gateway /Reference: : The Skinny client (i.e. an Ethernet Phone) uses TCP/IP to transmit and receive calls. Skinny messages are carried above TCP and use port Cisco IP Phones that use SCCP can coexist in an H.323 environment. When used with CUCM, the SCCP client can interoperate with H.323-compliant terminals. The client communicates with the CUCM using TCP/IP-based communication to establish a call with another H.323- compliant end station. Once the CUCM has established the call, the two H.323 end stations use connectionless UDP/IP-based communication for audio transmissions. The CUCM acts as a proxy by processing all H.323 and SIP transactions. This allows the IP Phone to process the VoIP RTP data stream. uide/sccp/ sccpaaph.pdf QUESTION 42 All call over the IP WAN use G.279. IP phones A and B use Cisco Unified Communications Manager Express. IP phone A is on a call with IP phone B. IP phone A conferences in analog phone C with IP phone B. Software conference resources are not being used. Drag the appropriate DSP resource for each gateway from the list to the correct locations in the graphic so the call can be complete. Select and Place:

39 Correct Answer: Section: Implement a gateway

40 /Reference: QUESTION 43 Refer to the exhibit. What will the class map do if a packet arrives that is marked with a CoS of 6 and a DSCP value of EF? Exhibit: A. The class map will match the packet and forward it to the policy map to be marked. B. The class map will not map the packet and no OoS will be applied. C. The class map will wait for the next packet in the stream to see if it has a CoS marking of 5 and then forward both packets to the policy map D. For the packet to be forwarded to the policy map, it must have a CoS of 5 and a DSCP value of EF. Correct Answer: B Section: Describe the need to implement QOS for voice and video /Reference: : If there is no match for a packet, no QoS processing occurs on the packet and the switch offers best-effort service to the packet. uration/guide/ swqos.html QUESTION 44 How many VoIP G.729 calls can be made simultaneously over a 128-kb/s Frame Relay circuit (Layer 3) if 50 percent of the circuit is dedicated to voice and 50 percent is dedicated to data? A. 1 B. 2 C. 3 D. 4 E. 5 Correct Answer: B

41 Section: Describe the basic operation and components involved in a voip call /Reference: : Bandwidth Calculation FormulasThese calculations are used: tml QUESTION 45 In a voice gateway, the configured codec complexity of the DSPs on a voice card can be changed. What is the impact on the DSPs if high codec complexity is configured? A. The codec complexity affects call density, which is the number of calls that are reconciled on the DSPs. This results in lower call density when high complexity is configured. B. With higher codec complexity, more calls can be processed. C. Lower codec complexity supports the fewest number of voice channels, provided that the lower complexity is compatible with the particular codecs that are in use. D. The DSP will process codecs that support high complexity transparently and shift to flex mode for those codecs that are not high complexity Correct Answer: A Section: Describe components of a gateway /Reference: : The difference between medium and high complexity codecs is the amount of CPU utilization necessary to process the codec algorithm, and therefore, the number of voice channels that can be supported by a single DSP. For this reason, all the medium complexity codecs can also be run in high complexity mode, but fewer (usually half) of the channels are available per DSP. com QUESTION 46 Refer to the exhibit. A new Cisco Unified Communications Manager Express system has been deployed and the technician is trying to add the first new IP phone to the system. The phone powers up, but it does not register with the system. The technician has verified that the phone is getting the proper VLAN information from Cisco Discovery Protocol. The phone is also getting the correct IP address and TFTP server address from DHCP. The phone has been assigned to an ephone and the correct MAC address is configured. With the information provided, which two of the following does the administrator need to verify to resolve this situation? (Choose two.) Exhibit: A. Verify that the ip helper-address is correctly configured. B. Verify that telephony-service has been configured.

42 C. Verify that the ephone has a button assigned. D. Verify that the tftp-server path has been configured. E. Verify that the Cisco Unified Communications Manager Express service is running. F. Verify that the correct phone type files are in the tftp-server path Correct Answer: DF Section: Implement CUCME to support endpoints using CLI /Reference: : Since the phone is getting the correct TFTP address, the next thing that needs to be verified is the TFTP Server path and IP Reachablity for the IP Phone to the TFTP Server. Once the TFTP settings has been verified, check if the files mentioned in the termxx.defaults.loads file is available in the TFTP Server for the phone to download. tion/ guide/7960trbs.html QUESTION 47 How does Packet Loss Concealment improve voice quality? A. Cisco Packet Loss Concealment technology decreases the voice sampling rate to 10 ms of the voice payload to smooth gaps in the voice stream. B. Packet Loss Concealment intelligently analyzes missing packets and generates a reasonable replacement packet to improve the voice quality C. Packet Loss Concealment will buffer 20 to 50 ms of a voice stream to minimize lost or out-of- order voice packets. D. Packet Loss Concealment will compensate for packet loss rates between 1 and 5 percent by generating a reasonable replacement packet to improve the voice quality. Correct Answer: B Section: Describe the need to implement QOS for voice and video /Reference: :

43 Packet loss concealment is a technology designed to minimize the practical effect of lost packets in VOIP. PLC mitigates against the effects of packet loss, which is the failure of one or more transmitted packets to arrive at their destination, by artificially regenerating the packet received prior to the lost one, followed by insertion of the duplicated packet into the gap. The digital value of the dropped packet is estimated by interpolation and an artificially generated packet inserted on that basis. html QUESTION 48 Which three methodologies are specific types of QoS used on a 512-kb/s point-to-point IP WAN link? (Choose three.) A. header compression B. FRF.12 C. LLQ D. MultilinkPPP E. RSVP Correct Answer: ABD Section: Describe the need to implement QOS for voice and video /Reference:

44 : Cisco IOS software offers a number of link-layer efficiency mechanisms or features designed to reduce latency and jitter for network traffic. These mechanisms work with queuing and fragmentation to improve the efficiency and predictability of the application service levels. Multilink PPP - provides packet interleaving, packet fragmentation, and packet resequencing across multiple logical data links, Frame Relay Fragmentation - simplifies the configuration of low-latency, low-jitter quality of service by enabling the queueing policy and fragmentation configured on the main interface to apply to all permanent virtual circuits (PVCs) and subinterfaces under that interface, Header Compression - A mechanism that compresses the IP header in a packet before the packet is transmitted. Configuration_Guide_Chapter.html QUESTION 49 Drag the appropriate IOS commands from the left and drop them in the spaces om the right to create a dial peer that will match all inbound call and prevent two-stage dialing on a T1 PRI cricuit. Not all boxes are used and not all options are used. Select and Place: Correct Answer:

45 Section: Implement a gateway /Reference: QUESTION 50 Which three of the following methods are used by a Cisco Unified Border Element to provide network hiding? (Choose three.) A. back-to-back user agent, replacing all H.323-embedded IP addressing B. IP network security boundary C. media flow-through D. RSVP E. IP network privacy F. intelligent IP address translation for call media and RTP flows Correct Answer: BEF Section: Implement cisco unified Border Element /Reference: : Cisco Unified Border Element can protect the network by hiding the network addresses and names for both the access (customer) side and the backbone (network core) side. A CUBE is designed to provide IP network privacy and topology hiding, IP network security boundary, Intelligent IP address translation for call media and signaling, Back-to-back user agent, replacing all SIP-embedded IP addressing, History information based topology hiding and call routing. A. was lead to wrong. back-to-back is work on SIP, not H323.

46 QUESTION 51 Drag the components from the left to drop them under the appropriate categories on the right. Select and Place:

47 Correct Answer: Section: Describe the basic operation and components involved in a voip call /Reference: QUESTION 52 Refer to the exhibit. Consider an outgoing call that is being placed in all three scenarios that are shown in the exhibit. What is the result of the call, going down the table from top to bottom?

48 Exhibit: A. success,success,success B. success, success, fail C. success, fail, success D. success, fail, fail E. fail, success, success F. fail, success, fail Correct Answer: A Section: Describe a dial plan /Reference: : Various combinations of COR lists and the results are shown in this table: COR List on Incoming dial-peer COR List on Outgoing dial-peer Result Reason No COR No COR Call succeeds COR is not in the picture No COR COR list applied for outgoing calls Call succeeds The incoming dial-peer, by default, has the highest COR priority when no COR is applied. Therefore, if you apply no COR for an incoming call leg to a dial-peer, then this dial-peer can make calls out of any other dialpeer, irrespective of the COR configuration on the outgoing dial-peer The COR list applied for incoming calls No COR Call succeeds The outgoing dial-peer, by default, has the lowest priority. Since there are some COR configurations for incoming calls on the incoming/originating dial-peer, it is a super set of the outgoing call COR configurations on

49 the outgoing/terminating dial-peer The COR list applied for incoming calls (super set of COR lists applied for outgoing calls on the outgoing dial-peer) The COR list applied for outgoing calls (subset of COR lists applied for incoming calls on the incoming dialpeer) Call succeeds The COR list for incoming calls on the incoming dial-peer is a super set of COR lists for outgoing calls on the outgoing dial-peer The COR list applied for incoming calls (subset of COR lists applied for outgoing calls on the outgoing dialpeer) The COR list applied for outgoing calls (super set of COR lists applied for incoming calls on the incoming dialpeer) Call cannot be completed using this outgoing dial-peer COR lists for incoming calls on the incoming dial-peer are not a super set of COR lists for outgoing calls on the outgoing dial-peer shtml QUESTION 53 Which three Cisco IOS commands are required to configure a voice gateway as a DHCP server to support a data subnet with the IP address of /24 and a default gateway of /24? (Choose three.) " A. ip dhcp pool B. subnet C. ip dhcp pool data D. network /24 E. network F. default-gw J24 G. default-router Correct Answer: CEG Section: Implement CUCME to support endpoints using CLI /Reference: : 1) To configure the DHCP address pool name and enter DHCP pool configuration mode, use the following command in global configuration mode: Router(config)# ip dhcp pool name - Creates a name for the DHCP Server address pool and places you in DHCP pool configuration mode 2) To configure a subnet and mask for the newly created DHCP address pool, which contains the range of available IP addresses that the DHCP Server may assign to clients, use the following command in DHCP pool configuration mode: Router(dhcp-config)# network network-number [mask /prefix-length] - Specifies the subnet network number and mask of the DHCP address pool. The prefix length specifies the number of bits that comprise the address prefix. The prefix is an alternative way of specifying the network mask of the client. The prefix length must be preceded by a forward slash (/). 3) After a DHCP client has booted, the client begins sending packets to its default router. The IP address of the default router should be on the same subnet as the client. To specify a default router for a DHCP client, use the following command in DHCP pool configuration mode: Router(dhcp-config)# default-router address [address2... address8] - Specifies the IP address of the default router for a DHCP client. One IP address is required; however, you can specify up to eight addresses in one command line.

50 QUESTION 54 Refer to the exhibit. When an international call to is placed from extension 2001, which of the following statements is true? Exhibit: A. The call will fail because no incoming COR list is applied. B. The call will succeed because the incoming COR list is a superset of the outgoing COR list. C. The call will fail because the incoming COR list is not a superset of the outgoing COR list. D. The call will succeed because the incoming COR list has the highest priority, by default, when no incoming COR list is applied. Correct Answer: D Section: Describe a dial plan /Reference: : By default, an incoming call leg has the highest COR priority and the outgoing COR list has the lowest COR priority. This means that if there is no COR configuration for incoming calls on a dial-peer, then you can make a call from this dial-peer (a phone attached to this dial-peer) going out of any other dial-peer, irrespective of the COR configuration on that dial-peer shtml QUESTION 55 In which situation would the trust boundary be located at the access layer? A. if the endpoints, both IP phones and PCs, are incapable of marking traffic properly B. if PCs are switched through an IP phone and the IP phone traffic can be trusted to mark both traffic streams

51 properly C. if the access layer switch cannot trust or re-mark incoming traffic from endpoints properly D. if there are endpoints that cannot be trusted and connect directly to the distribution layer Correct Answer: B Section: Implement cisco unified Border Element /Reference: : Network QoS policies need to be designed and implemented considering the entire borderless network. This includes defining trust points and determining which policies to enforce at each device within the network. Developing the trust model guides policy implementations for each device. Access layer switches communicate with devices that are beyond the network boundary and within the internal network domain. The QoS trust boundary at the access layer communicates with various devices that could be deployed in different trust models (trusted, conditional-trusted, or untrusted). The enterprise network administrator must identify and classify each of these device types into one of three different trust models, each with its own unique security and QoS policies to access the network: Untrusted - An unmanaged device that does not pass through the network security policies, for example, an employee-owned PC or network printer. Packets with 802.1p or DSCP marking set by untrusted endpoints are reset to default by the access layer switch at the edge. Trusted - Devices that passes through network access security policies and are managed by the network administrator. Even when these devices are maintained and secured by the network administrator, QoS policies must still be enforced to classify traffic and assign it to the appropriate queue to provide bandwidth assurance and proper treatment during network congestion. Conditionally-trusted - A single physical connection with one trusted endpoint and an indirect untrusted endpoint must be deployed as conditionally-trusted model. The trusted endpoints are still managed by the network administrator, but it is possible that the untrusted user behind the endpoint may or may not be secure (for example, a Cisco Unified IP phone and a PC). These deployment scenarios require a hybrid QoS policy that intelligently distinguishes and applies different QoS policies to the trusted and untrusted endpoints that are connected to the same port. / BN_Campus_QoS.html QUESTION 56 An access layer switch is configured to extend priority to an IP phone. Cisco Discovery Protocol is enabled on all ports. What are the three possible ways that an IP phone can be instructed to treat the Layer 2 CoS priority value of the attached PC? (Choose three.) A. trusted IEEE 802.1Q B. configured DSCP level C. configured CoS level D. trusted E. configured IEEE 802.1Q F. untrusted Correct Answer: CDF Section: Describe the need to implement QOS for voice and video /Reference: :

52 To process tagged data traffic (in IEEE 802.1Q or IEEE 802.1p frames), you can configure the switch to send CDP packets to instruct the phone how to send data packets from the device attached to the access port on the Cisco IP Phone. The PC can generate packets with an assigned CoS value. The phone can be configured to not change (trust) or to override (not trust) the priority of frames arriving on the phone port from connected devices. switchport priority extend {cos value trust} - Set the priority of data traffic received from the Cisco IP Phone access port: cos value - Configure the phone to override the priority received from the PC or the attached device with the specified CoS value. The value is a number from 0 to 7, with 7 as the highest priority. The default priority is cos 0. trust - Configure the phone access port to trust the priority received from the PC or the attached device uration/guide/ swvoip.html QUESTION 57 Refer to the exhibit. When is being matched with the outgoing dial peer that is shown in the exhibit, which of the following called numbers will be sent to the PSTN? Exhibit: A B C. 555 D. Null

53 E. 5 F Correct Answer: F Section: Describe a dial plan /Reference: : On outbound POTS dial peers, the terminating router normally strips off all digits that explicitly match the destination pattern in the terminating POTS dial peer. Only digits matched by the wildcard pattern are forwarded. The forward-digits command can be used to forward a fixed number of dialed digits, or all dialed digits, regardless of the number of digits that explicitly match the destination pattern. QUESTION 58 Drag the signaling streams to support SIP Early Offer from thre left and drop them in the correct box in the graphic on the right. Select and Place: Correct Answer:

54 Section: Implement a gateway /Reference: QUESTION 59 Drag the attributes of a scaleable numbering plan from the left and place them in the boxes on the right. Select and Place: Correct Answer:

55 Section: Describe a dial plan /Reference: QUESTION 60 Refer to the exhibit. The Acme Corp. uses H.323 to place calls to their supplier RR Industries. Acme also has a voice connection to an ITSP for long distance over a SIP network. Which configuration should Acme use to deploy the CUBE? Exhibit: A. service voice voip allow-connections h323 to h323 allow-connections h323 to sip B. voice service voip allow-connections h323 to h323 allow-connections h323 to sip C. service voice voip

56 allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to sip allow-connections sip to h323 D. voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 Correct Answer: D Section: Implement CUCME to support endpoints using CLI /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)allowconnections 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" ead0f.shtml QUESTION 61 Refer to the exhibit. Your customer has connected an existing PBX to the IP network. The PBX users can make calls to other extensions on the PBX but are unable to call the test extension All other applications on the IP network are working correctly. Compare the PBX system requirements to the configuration for R1 in the exhibit. Which configuration change will resolve the problem? Exhibit: A. configure operation 2-wire and type 5 on voice-port 1/1/0

57 B. configure operation 4-wire and type 5 on voice-port 1/1/0 C. configure forward digits all in dial-peer 1 POTS D. configure wink-start signaling on voice-port 1/1/0 Correct Answer: B Section: Implement CUCME to support endpoints using CLI /Reference: : Here, the PBX is configured to use type 5 signaling. Hence the Cisco device must be configured to use type V signaling. As is the case with Type I E&M signaling, Type V is a six- wire E&M signaling type that uses six leads: E, M, T, R, T1, and R1. So, 4 wire operations shouldbe configured along with Type V signaling. QUESTION 62 Where would you assign COR lists in Cisco Unified Communications Manager Express? A. ephone-dn B. voice register dn C. voice register pool D. ephone Correct Answer: A Section: Implement CUCME to support endpoints using CLI /Reference: : The corlist command sets the dial-peer COR parameter for dial-peers and the directory numbers that are created for Cisco IP phones associated with the Cisco CallManager Express router. To apply a class of restriction to a directory number, COR lists must be created in dial peers & the Directory number to which COR is to be applied must be configured in Cisco Unified CME. ephone-dn dn-tag : Enters ephone-dn configuration mode. corlist {incoming outgoing} cor-list-name : Configures a COR on the dial peers associated with an ephone-dn shtml QUESTION 63 Refer to the exhibit. Choose the correct configuration command set that will allow the gateway in zone BR to register with the gatekeeper at HQ so that the gateway is placed in zone BR. Exhibit:

58 A. B. C.

59 D. Correct Answer: C Section: Implement CUCME to support endpoints using CLI /Reference: : Notice that the router at zone Br is functioned as both gateway and gatekeeper and it uses the IP address of as the "zone local BR". Therefore if we want "the gateway in zone BR to register with the gatekeeper in the same zone" we must use in the command: h323-gateway voip id BR ipaddr In which, BR is the zone name defined in the "zone local BR" command (of the gatekeeper) and the is the IP address of an interface of the gatekeeper and it should be > B, D and E are not correct. A can be correct but it is not as clear as answer C. Notice that last command h323-gateway voip h323-id BRgw specifies the BRgw is the name of the gateway to communicate with the gatekeeper. =========================== : Gatekeeper is using the IP address and the Zone "local BR" is configured in it. Therefore if we want the gateway to register with zone BR in gatekeeper, we must use in the command:h323-gateway voip id BR ipaddr Notice that last command h323-gateway voip h323-id BRgw specifies that BRgw is the name of the gateway to communicate with the gatekeeper. QUESTION 64 Site A uses three-digit internal numbers and remote Site B uses four-digit internal numbers. All calls to the PSTN are routed through Site B. What dial plan below best represents provision simplicity, assuming the NANP numbering plan? A. Translate all called numbers at either site to ten digits. B. Translate all called numbers within Site B to three digits. C. Translate all called numbers within Site A to four digits. D. Translate all called numbers leaving Site A to ten digits. Correct Answer: D Section: Describe components of a gateway /Reference: : North American Numbering Plan (NANP) is designed around a 10-digit numbering plan:

60 (Sometimes you will see it as NXX NXXX XXXX, which means that the first and fourth digits can't be zero or one) It consists of 3-digit area codes and 7-digit telephone. For telephone numbers that are located within an area code, the PSTN uses a seven-digit dial plan numbers. Notice that "Site B uses four-digit internal numbers" means we need ten digits to access site B from an outside PSTN. Therefore, if people from Site A want to call people at site B and sometimes they just press 4 digits then the administrators should translate the called numbers to ten digits before leaving Site A. QUESTION 65 Which CUBE configuration will support H.323 protocol interworking and address hiding? A. B. C. D. Correct Answer: D Section: Implement CUCME to support endpoints using CLI /Reference: : Address hiding is a security feature of the CUBE which will hide the IP address of the originating gateway. This feature is turn on by default so we don't need to set it. A and B are not correct because the command "h323 interworking" doesn't exist (moreover A uses "media flow-around" feature which will turn off the address hiding feature). C is not correct because it uses "media flow-around" feature too ead0f.shtml QUESTION 66

61 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: Describe the basic operation and components involved in a voip call /Reference: : Human speech ranges from 200 Hz to 9000 Hz but Nyquist cut the sampling frequency range to 4000 Hz to save bandwidth although this cut down the quality of voice too. per0900aecd806fa57a.html QUESTION 67 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: Describe the basic operation and components involved in a voip call /Reference: : Quantization is the process of converting each analog sample value into a discrete value that can be assigned a unique digital code word. As the input signal samples enter the quantization phase, they are assigned to a quantization interval. All quantization intervals are equally spaced (uniform quantization) throughout the dynamic range of the input analog signal. Each quantization interval is assigned a discrete value in the form of a binary code word. The standard word size used is eight bits. If an input analog signal is sampled 8000 times per second and each sample is given a code word that is eight bits long, then the maximum transmission bit rate for Telephony systems using PCM is 64,000 bits per second. QUESTION 68 Refer to the graphic for IP addresses and telephone numbers. You are working with a customer that is opening a small sales office in R2. You would like to be able to have the user in R2 be able to dial into the PBX in R1 over the IP WAN. The R1 PBX uses loop start, a two-wire operation, and DTMF dialing. Please choose the correct FXO port configuration for R1. Exhibit:

62 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: Implement CUCME to support endpoints using CLI /Reference: : The commands specifying destination or session target should be configured inside a dial-peer and not on a Voice Port. Hence, B,C & D are not the correct choices. If in a dial-peer, the destination-pattern command is used, the voice port or session target must also be configured. QUESTION 69 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

63 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 signal immediate-start operation 4-wire type 4 Correct Answer: A Section: Implement CUCME to support endpoints using CLI /Reference: : Immediate-start is the simplest method of E&M access signaling. The calling side seizes the line by going offhook on its E-lead and sends address information as dual-tone multifrequency (DTMF) digits (or as dialed pulses on Cisco 2600 and Cisco 3600 series routers) following a short, fixed-length pause. E&M Type IV signaling is not supported on Cisco Platforms, Hence A is the correct choice. QUESTION 70 Which two are attributes of SCCP? (Choose two) A. It is Cisco proprietary. B. It is a supervisory signaling protocol. C. It is classified as client/server architecture. D. SCCP devices are considered intelligent endpoints. Correct Answer: AC Section: Describe components of a gateway /Reference: : SCCP is the only Cisco-proprietary VoIP protocol currently in use. The purpose of SCCP protocol is to provide a signaling protocol between the Cisco Unified Communications Manager and Cisco IP phones. Similar to MGCP, SCCP is a client/server protocol - A & C are correct. Supervisory signaling involves the detection of changes to the status of a circuit. We can say SCCP is more than a supervisory signaling protocol because it tells the phone exactly what to do. From the on-hook, off-hook, buttons pressed, lamp on/off, through the prompt, key settings, and even the dial tone - B is not correct. The beauty of SCCP is that it makes the endpoints very cheap comparing to the H.323 devices. The end stations (telephones) that use SCCP are called Skinny clients, which consume less processing overhead and they do not contain call control intelligence -D is not correct.

64 QUESTION 71 Which four types of ephone-dns are supported by SCCP in Cisco Unified Communication Manager Express? (Choose four) A. single-line B. dual-line C. shared-line, nonexclusive D. two directory numbers with one telephone number E. dual number F. octo-line Correct Answer: ABDF Section: Implement CUCME to support endpoints using CLI /Reference: : The following sections describe the types of directory numbers in a Cisco Unified CME system: Single-Line Dual-Line Octo-Line SIP Shared-Line (Nonexclusive) Two Directory Numbers with One Telephone Number Dual-Number Shared Line (Exclusive) Monitor Mode for Shared Lines Overlaid

65 l#wp QUESTION 72 When a Cisco Unified Border Element connects two VoIP streams using flow-around media, which of the following options call that the flow around and the components that flow through the device? A. All security information flows through the Cisco Unified Border Element, and all call signaling and RTP flows around the device B. Call signaling flows through and call media flows around the device. C. Call media flows through and call signaling flows around the device. D. the Initial call-signaling traffic flows through the device to initiate the call and all subsequent calls flow around the device. Correct Answer: C Section: Implement cisco unified Border Element /Reference: : Media flow through and media flow around mode is supported on the Cisco Unified Border Element (CUBE). The CUBE is always involved in the call setup (signaling) portion of the call, but the media (RTP bearer stream) may flow through the CUBE or be routed around the platform. Media flow through must be used to support many of the features available like IP address translation and IP address hiding. Media flow around allows the CUBE greater scalability in the number of calls that can be processed by one CUBE router. For "Media flow through" option, the media packets are passed through the CUBE, they will get terminated and re-originates with CUBE's IP address and port number, so here we cannot find the original gateway's ip address. This is one of the security feature in the CUBE. The default option is "media flow-through". QUESTION 73 Which codec complexity type will offer the greatest number of voice channels, provided that complexity type that are in use? A. low complexity B. medium complexity C. high complexity D. flex complexity Correct Answer: A Section: Describe the need to implement QOS for voice and video /Reference: : Usage Guidelines Codec complexity refers to the amount of processing required to perform voice compression. Codec complexity affects the call density--the number of calls reconciled on the DSPs. With higher codec complexity, fewer calls can be handled. Select a higher codec complexity if that is required to support a particular codec or combination of codecs. Select a lower codec complexity to support the greatest number of voice channels, provided that the lower complexity is compatible with the particular codecs in use. Reference: QUESTION 74 Which voice translation rule will expand extensions that are in the range to a 10 digit number?

66 A. /^...\(... $/^1/ B. /3.../ / / C. /^3...\(...$\)/ /408555\1/ D. /^3...\(...$\) / / / E. /^...$/ / &/ Correct Answer: C Section: Describe a dial plan /Reference: Reference: QUESTION 75 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 protocol sipv2 session target ipv4: C. dial-peer voice 91 voip session protocol sipv2 destination-pattern 91T... session target dtmf-relay rtp-nte digit-drop h245-alphanumeric

67 D. dial-peer voice 91 voip destination-pattern 91T... session target dtmf-relay rtp-nte digit-drop h245-alphanumeric E. dial-peer voice 1000 voip destination-pattern session target F. dial-peer voice 50 voip destination-pattern session target ipv4: Correct Answer: BCF Section: Implement CUCME to support endpoints using CLI /Reference: : The extension for ACME Corp is in the range of through Hence, Destination Pattern will be the correct match for routing calls to (AMCE's Call Agent) - B is the correct. Since ITSP is using a SIP Trunk, the dial-peer must be configured to use SIP signaling to complete the call leg. A basic SIP gateway configuration consists of simply adding one line under a VoIP dial peer configuration: session protocol sipv2 - C is correct The extension for RR Industries is in the range of through With 5 digit dialing enabled, Destination Pattern will be the correct match for routing calls to F is the correct. QUESTION 76 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:

68 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: Implement CUCME to support endpoints using CLI /Reference: : Notice that the call terminates with a cause Channel Unacceptable. This cause indicates a called user cannot negotiate for a B-channel other than that specified in the SETUP message. To enable the router to accept a B channel that is different from the B channel requested in the outgoing call setup message, use the isdn negotiate-bchan command in interface configuration mode. B-channel negotiation is not enabled. Most PRI switch types set the default channel ID to Exclusive in the setup message. QUESTION 77 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: Describe a dial plan /Reference:

69 : A hunt scheme is a selection procedure for choosing an interface or voice port. A trunk group with several trunk group members uses a hunt scheme to select an idle channel for routing an outgoing call. Suppose huntscheme least-used even down is enabled. The search goes through the trunk group members in descending order (C, B, A) to determine which member has the highest number of even-numbered idle channels. After selecting that trunk group member, the search looks for an even-numbered idle channel. If successful, the search selects an even- numbered idle channel to use for routing the call. If unsuccessful, the search goes through the trunk group members in the same descending order to select an odd-numbered idle channel. If successful, the search selects an odd-numbered idle channel for routing the call. QUESTION 78 Which item correctly describes the relationships between the feature and the category it belongs? Supports analog faxes and modems on a VoIP network Performs call setup and teardown between VoIP networks and the PSTN Interconnects segments of the same or different VoIP networks using different media types Interconnects segments of the same or different VoIP network using different signaling types A. Gateway 1 and 2 CUBE 3 and 4 B. Gateway 1 and 3 CUBE 2 and 4 C. Gateway 2 and 3 CUBE 1 and 4 D. Gateway 2 and 4 CUBE 1 and 3 Correct Answer: A Section: Implement a gateway /Reference: : Gateway Functionality : Gateways are responsible Media stream handling and speech path integrity, DTMF relay, Fax relay and pass-through, Digit translation and call processing, Dial peers and codec filtering, Carrier ID handling, Termination and re-origination of signaling and media The Cisco Unified Border Element is a session border controller designed to provide easy, secure, and costeffective connectivity between independent unified communications networks or network domains for different enterprises. It provides interconnection between incompatible applications within the enterprise network, between different enterprises for business-to-business applications, and between enterprise networks and service provider Session Initiation Protocol (SIP) trunks. The Cisco Unified Border Element provides key session management capabilities, H.323 and SIP interworking functions, and network-to-network interface security and demarcation capabilities. It performs most of the same functions of a public switched telephone network (PSTN)-to-IP gateway but joins two VoIP call legs. Media packets can either flow through (thus hiding the networks from each other) or around the Cisco Unified Border Element platform QUESTION 79 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

70 D. the first and sixteenth time slot Correct Answer: A Section: Describe the need to implement QOS for voice and video /Reference: : Each T1 channel carries a sequence of frames. These frames consist of 192 bits and an additional bit designated as the framing bit, for a total of 193 bits per frame. Super Frame (SF) groups twelve of these 193 bit frames together and designates the framing bits of the even numbered frames as signaling bits. CAS looks specifically at every sixth frame for the timeslot's or channel's associated signaling information. These bits are commonly referred to as A- and B-bits. Extended super frame (ESF), due to grouping the frames in sets of twenty-four, has four signaling bits per channel or timeslot. These occur in frames 6, 12, 18, and 24 and are called the A-, B-, C-, and D-bits respectively. QUESTION 80 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: Implement CUCME to support endpoints using CLI /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: 649.shtml) QUESTION 81 What is the decimal equivalent of the DSCP value AF11? A. 10 B. 14 C. 20 D. 12

71 Correct Answer: A Section: (none) /Reference: QUESTION 82 there was a question abt one way acceptable delay, jitter and packet loss. A. delay ms B. jitter - 30ms C. packet loss- 1% D. Correct Answer: ABC Section: (none) /Reference: QUESTION 83 What is default power used by a cisco 802.3af compatible ip phone. A. 6.3 watt B. 7 watt C watt

72 D. 0 watt Correct Answer: C Section: (none) /Reference: QUESTION 84 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: (none) /Reference: QUESTION 85 In North America, which E&M signaling type is used most often for geographically separated equipment?

73 A. Type I B. Type II C. Type III D. Type IV E. Type V Correct Answer: B Section: (none) /Reference: E&M Type I This is the most common interface in North America. * Type I uses two leads for supervisor signaling: E, and M. * During inactivity, the E-lead is open and the M-lead is connected to the ground. * The PBX (that acts as trunk circuit side) connects the M-lead to the battery in order to indicate the off-hook condition. * The Cisco router/gateway (signaling unit) connects the E-lead to the ground in order to indicate the off-hook condition. E&M Type II Two signaling nodes can be connected back-to-back. * Type II uses four leads for supervision signaling: E, M, SB, and SG. * During inactivity both the E-lead and M-lead are open. * The PBX (that acts as trunk circuit side) connects the M-lead to the signal battery (SB) lead connected to the battery of the signaling side in order to indicate the off-hook condition. * The Cisco router / gateway (signaling unit) connects the E-lead to the signal ground (SG) lead connected to the ground of the trunk circuit side in order to indicate the off-hook condition. E&M Type III This is not commonly used in modern systems. * Type III uses four leads for supervision signaling: E, M, SB, and SG. * During inactivity, the E-lead is open and the M-lead is set to the ground connected to the SG lead of the signaling side. * The PBX (that acts as trunk circuit side) disconnects the M-lead from the SG lead and connects it to the SB lead of the signaling side in order to indicate the off-hook condition. * The Cisco router / gateway (signaling unit) connects the E-lead to the ground in order to indicate the offhook 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 86 Which two codes together make up the number that follows the E.164 recommendation numbering scheme?

74 (Choose two.) A. country code B. subscriber code C. national destination code D. provider code Correct Answer: AB Section: (none) /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 87 When using 802.3af inline power, what is the default power wattage that the Catalyst switch provides to an IP phone? A. 0 watts B. 4 watts C. 7 watts D watts Correct Answer: D Section: (none) /Reference:

75 QUESTION 88 The voice gateway selects an inbound VoIP dial peer by matching the information elements in the message with the dial-peer attributes. From the list on the left, drag the elements to the right and drop them in the order in witch a voice gateway matches inbound calls. Select and Place: Correct Answer:

76 Section: Describe components of a gateway /Reference:

77 QUESTION 89 What is the best description of an MGCP endpoint? A. IP phones B. the gatekeepers in a VoIP network C. the interconnection between packet and traditional telephone networks D. any analog telephony device (PBX, switch, etc.) Correct Answer: C Section: (none)

78 /Reference: : QUESTION 90 Click and drag the type of call on the left to the type of voice port it applies to on the right. Select and Place: Correct Answer: Section: (none) /Reference: