ccna voice question (1).docx

8/11/2019 CCNA Voice Question (1).docx

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CCNA Voice Question

(Q) You are CCNA VOICE associate in Certkey.com. You need configure a voice port

that will allow the gateway to terminate 23 or 30 circuits from the PSTN. Which type

of voice port you will configure?

A. BRI

B. FXS

C. PRI T1/E1

D. E&M

(Q)Which of the following is selected first for an incoming dial peer?

A. Answer-address

B. incoming called-number

C. destination-pattern

D. pots port

(Q)Power over Ethernet (PoE) is the ability for the LAN switching infrastructure to provide

power over a copper Ethernet cable to an endpoint (Powered Device). This capability, once

referred to as "inline power", was originally developed and first delivered by Cisco in 2000 to

support the emerging IP Telephony deployments. IP Telephones, like desktop PBX phones,

need power for their operation and Power over Ethernet enables scalable and manageable

power delivery and simplifies deployments of IP Telephony. As Wireless networking emerged,

Power over Ethernet was also used to power these devices to allow for deployments in

locations where local power access did not exist. While IP telephones and wireless access

points are the most intuitive uses for PoE, the advent of 802.3af standardization of PoE opens

the door to a new generation of networked-attached devices such as video cameras, point-of-

sale devices, security access control (card scanners), building automation and industrial

automation just to name a few. When a Cisco switch supply PoE to an IP phone?

A. whenever switch and IP phone are Cisco products.


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B. when Cisco switch supports prestandard PoE and the IP phone supports 802.3af.

C. when Cisco switch and IP phone both support a common PoE method.

D. Only when both the switch and IP phone use 802.3af PoE.

(Q)What is the default VTP mode on Cisco switches?

A. Master

B. Client

C. Backup

D. Server

(Q)Which Cisco Unified Communications layer is the call processing layer?

A. Layer A including video,communications client, software-based phones, IP Phone

B. Layer B including Voice mail,Presence,Contact Center

C. Layer C including Cisco Unified Communications manger,Directory Server.

D. Layer D including Gateway, Router, Switch.

(Q)What are the three VTP modes on Cisco switches?

A. Master

B. Client

C. Server

D. Transparent

(Q)A SIP Trunk is a logical connection between an IP PBX and a Service Providers application

servers that allows voice over IP traffic to be exchanged between the two. When a call is

placed from an internal phone to an external number, the PBX sends the necessary

information to the SIP Trunk provider who establishes the call to the dialed number and acts

as an intermediary for the call. All signaling and voice traffic between the PBX and the provider


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is exchanged using SIP and RTP protocol packets over the IP network. Which two statements

about SIP trunk are true?

A. A SIP trunk configuration is mandatory for a UC500 device.

B. A SIP trunk is needed for internet access

C. A SIP trunk is needed only for voice if you are planning on using VoIP through a service provider.

D. A SIP trunk is not supported in a keyswitch configuration.

(Q)You are CCNA VOICE associate in Certkey.com. One user from your company wants to use

a signaling protocol on the voice gateways that require registration with the Cisco Unified

Communications Manager. Which protocol should you suggest to him?

A. SIP

B. Frame relay

C. SRTP

D. MGCP

(Q)An analog telephone is connected to a _______ port on a router?

A. FXO

B. E1

C. T1

D. FXS


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(Q)What is required to convert a G711ulaw call to G729?

A. Voice Termination resources

B. Conferencing resources

C. Converter resources

D. Transcoding resources

(Q)Whhich statement is true abou the difference between voice VLAN and native

VLAN?

A. Voice VLAN uses untagged frames whereas native VLAN uses 802.1Q frames

B. Voice VLAN uses tagged 802.1Q frames whereas native VLAN uses 802.1P frames

C. Voice VLAN uses tagged 802.1Q frames whereas native VLAN uses untagged frames

D. Voice VLAN uses untagged frames only when no PCs are connected behind the phones

whereas native VLAN always uses untagged frames

(Q) What is the maximum number of phones are supported on Call Manager

Express?

A. 48

B. 28

C. 240

D. 500

(Q)You are CCNA VOICE associate in Certkey.com. Which value should you assign to

option 150 for DHCP to operate correctly in a Cisco Unified Communications Manager

Express environment?

A. FTP server of the DNS server

B. MAC address of the DHCP server


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C. MAC address of the TFTP server

D. IP address of the TFTP server

(Q)What protocol needs to be enabled on an ATA if a fax machine is connected to

the ATA?

A. MGCP

B. SCCP

C. H323

D. SIP

(Q) You are CCNA VOICE associate in Certkey.com. Which two are considered

endpoints in a Cisco Unified Communications solution?

A. router

B. call agent

C. IP telephone

D. gateway

(Q) What protocol needs to be enabled on an ATA if an analog telephone is

connected to the ATA?

A. MGCP

B. SCCP

C. H323

D. SIP


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(Q) Jitter is an unwanted variation of one or more characteristics of a periodic signal in

electronics and telecommunications. Jitter may be seen in characteristics such as the interval

between successive pulses, or the amplitude, frequency, or phase of successive cycles. Jitter is

a significant factor in the design of almost all communications links (e.g. USB, PCI-e, SATA,

OC-48). In clock recovery applications it is called timing jitter. Which type of delay can lead to

jitter in a voice network?

A. transmission delay

B. serialization delay

C. CODEC delay

D. queuing delay

(Q)What device enables Call Admission Control in a CME environment?

A. Gateway

B. Transcoder

C. Bridge

D. Gatekeeper

(Q)You are CCNA VOICE associate in Certkey.com. You use IP phone A to place a

call to IP phone B. How many RTP streams are required for the call to be

successfully completed?

A. 2

B. 3

C. 4

D. 5


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(Q)Please match the protocol to the characteristic.

(1)SIP

(2)SCCP

(3)H.323

(4)MGCP

(I)Peer-to-Peer

(II)Client/Server

A. (I)-(1 2);(II)-(3 4)

B. (I)-(2 4);(II)-(1 3)

C. (I)-(1 4);(II)-(2 3)

D. (I)-(1 3);(II)-(2 4)

(Q)Which protocol provides VoiP packet sequence numbering?

A. IP

B. TCP

C. UDP

D. RTP

(Q) Time-Division Multiplexing (TDM) is a type of digital or (rarely) analog multiplexing in

which two or more signals or bit streams are transferred apparently simultaneously as sub-

channels in one communication channel, but are physically taking turns on the channel. Which

is the best description of time-division multiplexing?

A. One TDM frame consists of multiple timeslot per sub-channel.

B. the time domain is only one recurrent timeslots of fixed length


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C. Individual source signals are combined into a composite signal, which allows a capacity

equal to or greater than the sum of the component signals.

D. All sources get an interleaved slice of time, which offers the entire frequency range allocated for that timeslot

(Q)Which codec is less processor intensive?

A. G729

B. G729a

(Q)A PRI (Primary Rate Interface) is a telecommunication standard used in the Integrated

Services Digital Network or ISDN, for carrying multiple DS0 voice and data transmissions

between two physical locations. PRI was developed specifically for industrial or large quantity

users. PRI is an industrial ISDN line while the Basic Rate Interface, or BRI, is used to cater to

home and small enterprises. Which three characteristics apply to ISDN PRI? (Choose three.)

A. PRI offers 20 B channels and 1 D channel

B. the D channel is 64 kbps

C. can carry data, voice, or video

D. can carry vendor-specific PBX features

(Q)Identify the VoIP network component that provides CAC, bandwidth control and

management, and address translation.

A. Gateway

B. Gatekeeper

C. MCU

D. Call agent


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(Q) Which two of the following signaling protocols are peer-to-peer protocols?

(Choose 2.)

A. H.323

B. MGCP

C. SIP

D. SCCP

(Q)Which three headers are compressed by cRTP? (Choose 3.)

A. Data link

B. IP

C. UDP

D. RTP

(Q)Which of the following best describes a function of RTCP?

A. RTCP provides encryption, message authentication and integrity, and anti-replay service for

voice streams.

B. RTCP uses even-numbered UDP ports in the range 16,384-32,767 to transport voice

payloads

C. RTCP provides out-of-band control information for an RTP flow

D. RTCP caches an RTP packets Layer 3 and Layer 4 headers in the routers at each end of a

link, resulting in lower bandwidth demand for subsequent RTP packets.


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(Q)An ITSP (Internet Telephony Service Provider) offers an Internet data service for making

telephone calls using VoIP (Voice over IP) technology. Most ITSPs use SIP, H.323, or IAX

(although H.323 use is declining)[citation needed] for transmitting telephone calls as IP data

packets. Customers may use traditional telephones with an analog telephony adapter (ATA)

providing RJ11 to Ethernet connection.What are two benefits of using an ITSP for long

distance telephony services?

A. The circuits are dedicated only to voice

B. Connection to an ITSP is only available in full T1/E1/PRI circuit quantities.

C. Connection to an ITSP is very granular and can provision from 1 to 100 simultaneous calls.

D. Connection to an ITSP allows you to use the bandwidth that is guaranteed to the voice traffic for data when voice is

not using the bandwidth.

(Q)Please describe how does LLQ help ensure that voice quality is maintained in a

converged network?

A. The Low Latency Queueing feature allocates minimum bandwidth guaranteed to voice

traffic.

B. The Low Latency Queueing feature allocates a priority queue to voice traffic at a guaranteed rate

C. The Low Latency Queueing feature allocates a priority queue and a minimum guaranteed

bandwidth queue for voice.

D. The Low Latency Queueing feature ensures that all traffic is treated fairly and hence voice

traffic is not severely impacted.

(Q)Which three of the following are appropriate solutions to address latency issues

in a VoIP network? (Choose 3.)

A. Use dejitter buffers

B. Increase bandwidth

C. Fragment data packets

D. Prioritize voice packets


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(Q) Which two of the following VoIP gateway platforms are considered to be

Integrated Services Routers (ISRs)? (Choose 2.)

A. Cisco 2600XM Series

B. Cisco 2800 Series

C. Cisco 3700 Series

D. Cisco 3800 Series

(Q)You are CCNA VOICE associate in Certkey.com.You are planning stages of deploying a

Cisco Unified Communications solution. Previously, Certkey was leasing a traditional PBX

system from the telcom and very little experience with voice. What two signaling methods

between the IP phone and the Cisco Unified Communications Manager Express are available

for usage?

A. H.323

B. SCCP

C. MGCP

D. SIP

(Q) In a Cisco UCM single-site deployment, what is the maximum number of IP

phones that can register with a UCM cluster?

A. 2500

B. 7500

C. 10,000

D. 30,000


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(Q)Which option cannot be configured using Cisco Configuration Assistant?

A. voice VLAN only

B. voice and data VLANs

C. script selection for the AutoAttendant

D. SIP trunk to an ITSP

(Q)You are CCNA VOICE associate in Certkey.com. When would you require a voice

gateway? (Choose two.)

A. When you connect a branch location to IP WAN

B. When you connect a branch location using VoIP to the PSTN

C. When you connect a Cisco Unified Communications Manager to a LAN

D. When you connect a Cisco Unified Communications network to a PBX

(Q) Which four of the following are Cisco-supported IP telephony deployment

models?

A. Single site

B. Multisite with centralized call processing

C. Clustering over the IP WAN

D. Multisite with distributed call processing

(Q) In telephony, an automated attendant (also auto attendant or auto-attendant, or

sometimes autoattendant or AA) system allows callers to be automatically transferred to a

users extension without the intervention of a receptionist. A receptionist, who acts as the

telephone operator, can be reached by pressing 0 on most systems. Although an automated

attendant is usually a feature on modern PBX and key phone systems, it is possible to provide

one on standard lines and phones .Which best describes the auto-attendant in a Cisco Unified

Communications environment?

A. automatically intruct the system what to do when it reaches a particular system ID


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B. automatically allows callers to leave voice-mail messages 24 * 7

C. greets and guides callers through a telephony system in a friendly and timely manner, allowing them to reach an

endpoint, leave messages, or speak to an operator

D. allows the option of listening to, composing, replying to, forwarding, or deleting calls or

voice-mail messages through a website without the need of a live telephone operator

(Q)Please describe the Cisco Unity Express default AutoAttendant

A. The default AA must be enabled during the Initialization Wizard process

B. Enabling the default AA is not mandatory during the Initialization Wizard process.

C. The default AA is enabled by default with the exception of the prompts that must be

recorded via AvT.

D. The default AA cannot be used as is, and it must be customized for the particular

environment it will be used in.

(Q)Identify two gateway supplementary services. (Choose 2.)

A. Hold

B. DTMF relay

C. Transfer

D. Transcoding

(Q)A telephone numbering plan is a plan for allocating telephone number ranges to countries,

regions, areas and exchanges and to non-fixed telephone networks such as mobile phone

networks. Dial plans should not be confused with numbering plans. A closed numbering plan,

such as found in North America, features fixed length area codes and local numbers. An open

numbering plan, as found in assorted countries that have not yet standardized, features

variance in length of area code or local number, or both. The rules for dialing the numbers

assigned by the numbering plan vary by defining which digits need not always be dialed

(codes) and digits that must always be dialed (local number). What is a numbering plan?

A. numbering plan defines the path from one endpoint to another


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B. numbering plan defines the calling privileges of an endpoint

C. numbering plan defines how digits are manipulated in processing a call.

D. numbering plan defines the endpoint addressing used in a voice system

(Q)According to the G.114 recommendation, the maximum one-way delay for voice

should ideally not exceed how much delay?

A. 100 ms

B. 150 ms

C. 200 ms

D. 250 ms

(Q) E.164 is an ITU-T recommendation which defines the international public

telecommunication numbering plan used in the PSTN and some other data networks. It also

defines the format of telephone numbers. E.164 numbers can have a maximum of fifteen

digits and are usually written with a + prefix. To actually dial such numbers from a normal

fixed line phone the appropriate international call prefix must be used.

Which of the following is not the part of the e.164 number in the ITU-T numbering plan forgeographic areas?

A. SANC.SPID

B. national destination code

C. station code

D. country code

E. subscriber number


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(Q) You are CCNA VOICE associate in Certkey.com.Pleaes point out three

components of a dial plan. (Choose three.)

A. endpoint protocol

B. endpoint addressing

C. digit manipulation

D. call coverage

(Q)Identify the preferred voice quality measurement approach for VoIP networks.

A. MOS

B. PESQ

C. QRT

D. PSQM

(Q)You are CCNA VOICE associate in Certkey.com. You need configure a voice port

that will terminate a loop or ground start line from the PSTN or a PBX. Which type of

voice port you will configure?

A. FXO

B. FXS/FXO

C. PRI T1/E1

D. BRI

(Q)Which method of fax relay uses a store-and-forward approach?

A. T.30

B. T.37

C. T.38

D. Cisco Fax Relay


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(Q) You are CCNA VOICE associate in Certkey.com. You need to configure a Cisco Unity

Express system with a voicemail pilot number of 1900, an AutoAttendant with pilot number of

2900, and an Administration via Telephone pilot number of 3900. What is the minimum

number of SIP dial peers required?

A. 0

B. 1

C. 2

D. 3

(Q)What codec is required for fax pass-through and/or modem pass-through?

A. G.711

B. G.723

C. G.729

D. G.729ab

(Q) Identify three quality issues that can result because of a lack of network bandwidth.

(Choose 3.)

A. Jitter

B. Impedance

C. Delay

D. Packet loss


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(Q) You are CCNA VOICE associate in Certkey.com. What will happen to a call that is

transferred when no second line is available?

APSC(config-telephony-service)#transfer-system full-consult

A. The call will be terminated

B. The call will be transferred without consultation using H.450.2 standard methods.

C. The call will fall back to a full-blind transfer.

D. The call will be transferred without consultation using the proprietary Cisco transfer

method.

(Q)What protocol is used to communicate between a DSP farm configured on an IOS

router and a Cisco Unified Communications Manager server?

A. H.323

B. MGCP

C. SIP

D. SCCP

(Q)You are CCNA VOICE associate in Certkey.com. You are setting up phones for

Cisco Unified Communications Manager Express. Which three commands should you

use first?

A. reload

B. max-dn

C. telephony-service

D. max-ephones


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(Q) Time-Division Multiplexing (TDM) is a type of digital or (rarely) analog multiplexing in

which two or more signals or bit streams are transferred apparently simultaneously as sub-

channels in one communication channel, but are physically taking turns on the channel. The

time domain is divided into several recurrent timeslots of fixed length, one for each sub-

channel. A sample byte or data block of sub-channel 1 is transmitted during timeslot 1, sub-

channel 2 during timeslot 2, etc. One TDM frame consists of one timeslot per sub-channel.

After the last sub-channel the cycle starts all over again with a new frame, starting with the

second sample, byte or data block from sub-channel 1, etc. Which statement is true

concerning time-division multiplexing?

A. TDM transmits one voice signal at a time over a four-wire path.

B. TDM transmits multiple voice signals across four separate communication mediums

C. TDM transmits multiple separate voice signals over one communication medium by quickly interleaving pieces of

each signal, one after another.

D. TDM transmits one voice signal at a time over one or more communication mediums by

quickly dividing pieces of each signal into equal bandwidth sizes and sending them in the order

they are received. Information from each data channel is allocated bandwidth based on the

current bandwidth needed for each time slot. This is determined by whether or not there is

data that needs to be transmitted.

(Q)What is the Layer 2 overhead (in bytes) for Frame Relay traffic?

A. 3 bytes

B. 5 bytes

C. 6 bytes

D. 18 bytes


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(Q) You are CCNA VOICE associate in Certkey.com and you want to install a new UC500

solution. Which information from the existing numbering plan that helps you design an

effective VoIP dial plan? (Choose all the apply.)

A. station number

B. number of digits forwarded by the local CO

C. direct inward dial ranges

D. length of internal number extensions

(Q)What are the functions of an H.323 gateway?

A. Converts an alias address to an IP address

B. Responds to bandwidth requests and modifications

C. Transmits and receives G.711 PCM-encoded voice

D. Performs translation between audio, video, and data formats

(Q) You are CCNA VOICE associate in Certkey.com.Your company is migrating from a

traditional PBX to a Cisco Unified Communications system. During migration process you need

to provide common channel signaling connectivity to the PBX from the Cisco Unified

Communications system. Which type of gateway would be required?

A. voice gateway

B. digital trunk gateway

C. analog trunk gateway

D. digital station gateway

(Q)Which call control model does MGCP use?

A. Distributed

B. Centralized

C. Ad hoc D. Hybrid


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(Q)Which statement is true about the Cisco Unity Express Network Module?

A. The network module takes up a router slot

B. The network module does not take up a router slot

C. The network module does not run the Linux OS

D. The network module require the Cisco Unified Communications Manager Express to be

preconfigured

(Q)Which statement is true about Cisco Unity Express AIM Module?

A. The AIM module takes up a router slot

B. The AIM module does not take up a router slot

C. The AIM module runs the Linux OS

D. The AIM module does not require the Cisco Unified Communications Manager Express to be

preconfigured

(Q) Which command displays a count of successful and unsuccessful control

commands?

A. show mgcp calls

B. show mgcp statistics

C. show mgcp

D. debug mgcp statistics

(Q)You are CCNA VOICE associate in Certkey.com. You are setting up the partially

automated telephone. What is the key command to use?

A. auto qos

B. auto assign

C. auto start-dn

D. auto phone-type


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(Q)Identify four SIP servers. (Choose 4.)

A. Registrar

B. Gateway

C. Redirect

D. Location

E. Proxy

(Q)An ITSP (Internet Telephony Service Provider) offers an Internet data service for making

telephone calls using VoIP (Voice over IP) technology. Most ITSPs use SIP, H.323, or IAX

(although H.323 use is declining)[citation needed] for transmitting telephone calls as IP data

packets. Customers may use traditional telephones with an analog telephony adapter (ATA)

providing RJ11 to Ethernet connection .Which types of high-density trunk can be used to

connect to an ITSP? (Choose all the apply.)

A. T1/E1 CAS

B. FXS

C. T1/E1 PRI

D.E&M

(Q)You are CCNA VOICE associate in Certkey.com.You need to assign voice VLAN

110 on a Cisco IOS switch. Which command should you use?

A. switchport vlan 110

B. switchport access voice vlan 110

C. switchport voice vlan 110

D. switchport access vlan 110


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(Q)A call leg is a virtual connection to set up a call whereas a dial peer is a physical

connection to complete an end-to-end call.

A. TRUE

B. FALSE

(Q)The call leg and the dial peers are both logical connections used to complete an end-to-

end call.

A. TRUE

B. FALSE

(Q)What is the relationship between a call leg and a dial peer?

A. A call leg is a virtual connection to set up a call whereas a dial peer is a physical connection

to complete an end-to-end call.

B. The call leg and the dial peers are both logical connections used to complete an end-to-end call.

C. A call leg is a virtual connection that is set up and torn down before the dial peer is

established.

D. The call leg and the dial peer are both physical connections used to complete an end-to-end

call.

(Q)Which of the following is not the function provided by a switching system in a

traditional telephony network?

A. call setup

B. call supervision

C. customer IDs and telephone numbers

D. codec processing


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(Q)You are CCNA VOICE associate in Certkey.com. You need configure a voice port

that will allow the gateway to terminate two circuits from the PSTN or a PBX. Which

type of voice port should you configure?

A. PRI T1/E1

B. BRI

C. PRI T1

D. E&M

(Q)You are CCNA VOICE associate in Certkey.com. Your company has successfully installed

the CUE module in its routers chassis. Which two configuration items are necessary for the

module to function properly?

A.You need assign a default gateway to the service module

B. You need use the ip numbered command to save the subnet

C. You need create a subnet that appears in all of the routing tables to make the module

reachable.

D. The interface service engine needs to have an IP address that is on the same subnet as the service module.

(Q)You are CCNA VOICE associate in Certkey.com. Certkey is setting up Call Transfer for its

Cisco Unified Communications Manager Express solution. Certkey uses 5-digit extensions and

wants to transfer calls outside the network to the CEOs home. The CEOs telephone number is

877-666-3333.Which configuration command will allow this to occur?

A. SBB(config)#transfer-pattern .T

B. SBB(config-telephony-service)#transfer-pattern .T

C. SBB(config-telephony-service)#transfer-pattern 8776663333

D. SBB(config-ephone-dn)#transfer-pattern 8776663333


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(Q)You are CCNA VOICE associate in Certkey.com. Your company has a few slow links in its

voice and data network. Which two techniques can be used to reduce delay in voice

transmission?

A. Framerelay

B. buffering voice packets

C. fragmentation of large packets

D. compression of IP, RTP, and UDP headers

(Q)You are CCNA VOICE associate in Certkey.com. Certkey is setting up Call Transfer for its

Cisco Unified Communications Manager Express solution. Certkey uses 5-digit extensions and

wants to transfer calls outside the network to the CEOs home. The CEOs telephone number is

877-666-3333.

Which configuration command will allow this to occur?

A. SBB(config)#transfer-pattern .T

B. SBB(config-telephony-service)#transfer-pattern .T

C. SBB(config-telephony-service)#transfer-pattern 8776663333D. SBB(config-ephone-dn)#transfer-pattern 8776663333

(Q)What protocol does a Cisco IP phone use to indicate to the switch how much

power is needed?

A. PoE

B. CDP

C. VTP

D. cRTP


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(Q) What protocol is used to monitor and provide control information about the

quality of an RTP session?

A. UDP

B. RTP

C. NTP

D. RTCP

(Q)What device is responsible for converting analog voice packets from traditional

voice formats to IP packets?

A. MGCP gateway

B. H.323 gateway

C. Media Termination Point

D. Digital signal processor

E. Transcoder

(Q)Which protocol is used to inform the IP phone of its voice VLAN ID?

A. Cisco keepalives

B. Cisco Discovery Protocol

C. Cisco Spanning Tree Protocol

D. Cisco VLAN Discovery Protocol

(Q)What protocol does a Cisco IP phone use to indicate to the switch how much

power is needed?

A. PoE

B. CDP C VTP D. cRTP


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(Q) Power over Ethernet (PoE) is the ability for the LAN switching infrastructure to provide

power over a copper Ethernet cable to an endpoint (Powered Device). This capability, once

referred to as inline power, was originally developed and first delivered by Cisco in 2000 to

support the emerging IP Telephony deployments. IP Telephones, like desktop PBX phones,

need power for their operation and Power over Ethernet enables scalable and manageable

power delivery and simplifies deployments of IP Telephony. As Wireless networking emerged,

Power over Ethernet was also used to power these devices to allow for deployments in

locations where local power access did not exist. While IP telephones and wireless access

points are the most intuitive uses for PoE, the advent of 802.3af standardization of PoE opens

the door to a new generation of networked-attached devices such as video cameras, point-of-

sale devices, security access control (card scanners), building automation and industrial

automation just to name a few. When a Cisco switch supply PoE to an IP phone?

A. whenever switch and IP phone are Cisco products.

B. when Cisco switch supports prestandard PoE and the IP phone supports 802.3af.

C. when Cisco switch and IP phone both support a common PoE method.

D. Only when both the switch and IP phone use 802.3af PoE.

(Q) In a Cisco UCM multisite WAN with centralized call-processing deployment model, what

redundancy feature should be configured on remote site routers to provide basic IP telephony

services in the event of a WAN outage?

A. AAR

B. SRST

C. CAC

D. V3PN

(Q)Place the steps for inbound dial-peer matching in the correct order. (1)If no matches are

found, the system uses the default dial peer. (2)Look for the destination-pattern command in

a dial peer that matches the calling number or ANI string of the incoming call leg.

(3)Look for the POTS dial peer port configuration that matches the voice port associated with


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the incoming call(POTS dial peers only)

(4)Look for the incoming called-number command in the dial peer that matches the called

number or DNIS string in the inbound call leg.

(5)Look for the answer-address command in a dial peer that matches the calling number or

ANI string of the inbound call leg.

(I)Step1

(II)Step2

(III)Step3

(IV)Step4

(V)Step5

A. (I)-(4);(II)-(2);(III)-(5);(IV)-(3);(V)-(1)

B. (I)-(4);(II)-(5);(III)-(2);(IV)-(3);(V)-(1)

C. (I)-(5);(II)-(4);(III)-(2);(IV)-(3);(V)-(1)

D. (I)-(4);(II)-(5);(III)-(2);(IV)-(1);(V)-(3)

(Q)Please match the phrase to the correctly completes the statement for each task.

(1)to be on the same subnet as the router

(2)with a static IP address or IP address unnumbered (3)to be the same as the service engine

(I)Configure the service engine interface

(II)Configure the service-module IP address

(III)Configure the Cisco Unity Express IP default gateway

A. (I)-(3);(II)-(2);(III)-(1)

B. (I)-(1);(II)-(2);(III)-(3)

C. (I)-(2);(II)-(1);(III)-(3)

D. (I)-(2);(II)-(3);(III)-(1)


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(Q)Which identifies the amplitude of an analog signal stream?

A. A x C

B. A x B

C. A

D. C

(Q)You are CCNA VOICE associate in Certkey.com. You execute the show ephone command

in the Cisco Unified Communications manger as below. What information will you get from the

output?

A. There is only one IP phone.
http://cdn.aiotestking.com/wp-content/uploads/2011/08/CCNA-Voice-18.pnghttp://cdn.aiotestking.com/wp-content/uploads/2011/08/CCNA-Voice-15.pnghttp://cdn.aiotestking.com/wp-content/uploads/2011/08/CCNA-Voice-18.pnghttp://cdn.aiotestking.com/wp-content/uploads/2011/08/CCNA-Voice-15.png


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B. There are two registered IP phones. Shared number 5010 on line 2. Message waiting on

shared line.

C. There are two IP phones. Phone 2 is unregistered. Shared number 5010 on line 2. Message

waiting on shared line

D. There are two IP phones. Phone 2 is unregistered. Shared number 5010 on line 2. Message waiting on line 1 of

phone

(Q)Which statement about the Cisco Unified Communications Manager Express configuration

is correct based on the following information?

A.Nothing needs to be added. MoH will work fine

B. MoH will not work because the multicast address is incorrect

C. MoH will not work because you need to add the route 10.3.130.1 command in multicast

moh 239.1.1.1 port 2000.

D. The configuration will work only for calls that are being transferred because the transfer-

system command has been added.
http://cdn.aiotestking.com/wp-content/uploads/2011/08/CCNA-Voice-27.png


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(Q)You are CCNA VOICE associate in Certkey.com. You need configure PSTN voice gateway.

Which two dial peers will correctly route calls to emergency services?

A. dial-peer voice 1 pots destination-pattern 9911 prefix 911 port 1/0/0

B. dial-peer voice 2 pots destination-pattern 911 forward-digits all port 1/0/0

C. dial-peer voice 9911 pots destination-pattern 9911 forward-digits all port 1/0/0

D. dial-peer voice 2 pots destination-pattern 911 forward-digits 3 port 1/0/1

(Q) You are CCNA VOICE associate in Certkey.com. You need to verify the network

configuration that was performed by a colleague. Which of the following conclusions would be

accurate?

A. The network subnet mask should be class A.

B. The network subnet mask should be class C.

C. The CME IP address should be in a different subnet to that of the IP phones.


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D. The CME IP address should be in the exclusion range.

(Q) Please describe approximately percentage of voice packets can be dropped

before voice quality becomes poor.

A. 1 -2%

B. 15%

C. 5 -10%

D. Less than or equal to 1%

(Q)Please choose the correct length arrow(s) that represent call legs to proper position to

show how call leg(s) are used in a call. Arrows may be used more than once, and not all may

apply.


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Answer: Pending

(Q)You are CCNA VOICE associate in Certkey.com. Which step of digitizing analog signals

does this represent based on the following information?


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A. encoding

B. quantization

C. signal sample

D. signal compression

(Q)You are CCNA VOICE associate in Certkey.com. What is the purpose of access

code 9 based on the following information?

A. The access code 9 connects a caller to the operator when they press 9 on their phone

keypad.

B. This is a level 1 security

C. The access code 9 is the access code that internal users must use to make PSTN calls.

D. he access code 9 is required when interfacing with a traditional PBX system


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(Q)Refer to the exhibit. Which step of digitizing analog signal does this represent?

A. signal sample

B. encoding

C. sample quantization

D. 8-bit digital encoding


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(Q)Please explain the parameter Overlay on button number 2.

A. It means that this IP phone has more than one extension configured on line 2.

B. It means any calls to line 1 when busy will be automatically forwarded to line 2.

C. It means that the user can dial using any Overlay application, for example, Click to dial

application.

D. It means that the extension on line 2 is part of an Overlay group that handles calls that

have been sitting in the queue and exceed the Lay limit.


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(Q)You are CCNA VOICE associate in Certkey.com. Which statement is true about the VLAN

Number field based on the following information?

A. Creating a new voice VLAN as long as it falls in the range 1 to 1001.

B. Creating a new voice and data VLAN as long as it falls within the range 1 to 1001

C. Creating a voice VLAN as long as it falls in the range of 1 to 100

D. Assigning a voice VLAN here in the range of 1 to 1001

(Q)You are CCNA VOICE associate in Certkey.com. How will an incoming call to 503 be routed

based on this configuration?

A. Incoming call will be routed sequentially to users that belong to hunt group 1.

B. The call will be routed sequentially to users that belong to hunt group 2.

C. The call will be routed sequentially to users that belong to hunt group 3.


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D. The call will be routed sequentially to users that belong to hunt group 3 and then

sequentially to hunt groups 1 and 2.

(Q)You are CCNA VOICE associate in Certkey.com. What is the difference between a PBX and

a keysystem under the Device tab?

A. each phone has nearly identical configuration in PBX but a keysystem has a unique

extension for each phone.

B. each phone has a unique extension in PBX but a keysystem has nearly identical configurations for each phone.

C. A PBX operates 247 and requires no activation.

D. A keysystem can be activated only by the operator by turning the Systemkey.

(Q)You are CCNA VOICE associate in Certkey.com. You configure a dial peer that

points to Cisco Unity Express. Which line of configuration is incorrect?

A. TEST-B

B. TEST-C


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C. TEST-D

D. TEST-F

(Q)You are CCNA VOICE associate in Certkey.com. How many discrete call legs are needed to

set up a call between the POTS phone attached to TEST-Router 1 and the phone in the PSTN?

A. 3

B. 4

C.6

D. 7


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(Q)You are CCNA VOICE associate in Certkey.com. When you place a call to 2000, phones 1

and 2 are paged. When you place a call to 2001 pages phones 3 and 4. What command is

missing so that a call to 2002 pages all phones1,2,3,4?

A.paging group 20 21

B. paging group 10 20

C. paging group all

D. paging group 2000 2001 2002


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(Q)You are CCNA VOICE associate in Certkey.com. You place a phone call from ext1234 to

ext 2010. Which inbound dial peer on TEST-CMERouter1 will be matched?

A. None, the call will drop

B. default dial peer 30

C. voip dial peer 30

D. default dial peer


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(Q)You are CCNA VOICE associate in Certkey.com. The configuration for Call Park has been

set. Users complain that the Call Park softkey does not appear on their phones. How to solve

this problem?

A. You should add park-slot number ephone-dn 5

B. You should restart the phones.

C. You should configure park-slot number outside the range of 5XXX numbers.

D. The call-park command enable needs to be added under telephony-service.

(Q)You are CCNA VOICE associate in Certkey.com. You want to allow IP Phones in subnet

10.100.1.0 to register with Cisco Unified Communications Manager Express at address

10.10.1.1. Which two commands should you use?


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A. ip helper address 10.10.1.1 under interface FastEthernet0/0.100

B. option 150 ip 10.10.1.1 under dhcp pool Phones

C. default-router 10.100.1.1 under dhcp pool Phones

D. subnet prefix-length 24 under dhcp pool Phones

(Q)Which ephone-dn type does the exhibit represent?

A. shared ephone-dn

B. overlaid ephone-dn

C. dual-line ephone-dn

D. single-line ephone-dn


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(Q)10.1.1. Which two commands should you use?

A. ip helper address 10.10.1.1 under interface FastEthernet0/0.100

B. option 150 ip 10.10.1.1 under dhcp pool Phones

C. default-router 10.100.1.1 under dhcp pool Phones

D. subnet prefix-length 24 under dhcp pool Phones


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(Q)Which two types of dial peers are needed to complete this call end-to-end?

A. serial dial peer

B. parallel dial peer

C. POTS dial peer

D. VoIP dial peer

(Q)Please match the description to the signaling type it corresponds to. (1)events that occur

on the trunk, including seizure, wink, and answer (2)tones such as ringing or busy and

announcements such as no longer in service (3)digits dialed or called party number that can

be sstem-specific or variant-specific (I)Address Signaling

(II)Supervisory Signaling

(III)Informational Signaling

A. (I)-(3);(II)-(2);(III)-(1)

B. (I)-(1);(II)-(2);(III)-(3)

C. (I)-(3);(II)-(1);(III)-(2)

D. (I)-(1);(II)-(3);(III)-(2)


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(Q)The company is trying to configure Music on Hold for their Cisco Unified Communications

Manager Express solution. What are two possible problems with this configuration? (Choose

two)

AIOTestKing(config)# telephony-service

AIOTestKing(config-telephony)# moh minuet.au

AIOTestKing(config-telephony)# multicast moh 224.10.16.4 port 2000 route 10.10.29.17

10.10.29.33

AIOTestKing(config-telephony)# exit

AIOTestKing(config)#telephony-service

AIOTestKing(config-telephony)# moh rockin.au

AIOTestKing(config-telephony)# multicast moh 224.10.16.4 port 2001

AIOTestKing(config-telephony)# exit

A. The second multicast entry does not have a route listed.

B. The ip-source address command has not been configured

C. IP phones do not support multicast at 224.x.x.x addresses.

D. You must disable the use of the first moh file with the no moh command before configuring the second file

E. Port 2000 is recommended because it is already used for normal RTP media transmissions

between IP phones and the router.

F. Port 2000 is not recommended because it is already used for normal RTP media

transmissions between IP phones and the router.


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(Q)Which ephone-dn type does the picture represent?

A. shared ephone-dn

B. overlaid ephone-dn

C. two ephone-dns with one number

D. single-line ephone-dn


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(Q)Configure Hunt Group so that all incoming PSTN calls always go to extension 5001. If

extension 5001 is not available, route the call to extension 5002. If both extensions are busy

or unavailable then send the call to Voicemail. About this question, we do not have proper

answer, if you have then please contact us.

No

Answer: Pending

(Q) Please match the CAS component to the office that it will apply.

(1)CAS E1 (2)CAS T1 (3)1.544 mb/s (4)2.048 mb/s

(5)RBS in-band (6)24 voice channels (7)30 voice channels (8)out-of-band signaling in time

slot (I)London, United Kingdom (II)Chicago, United States

A. (I)-(1 4 5 8);(II)-(2 3 7 6)

B. (I)-(2 4 7 8);(II)-(1 3 5 6)

C. (I)-(1 4 7 8);(II)-(2 3 5 6)

D. (I)-(1 3 7 8);(II)-(2 4 5 6)

(Q)3.130.254. IP addresses in the range 10.3.130.1 to 10.3.130.9 should be excluded. The

IP phones are connected to an EtherSwitch module, interface GigabitEthernet1/0 on a Cisco

2811 router as per the topology diagram. The voice VLAN ID is 130 and default router for the

IP phones is 10.3.130.1.

The EtherSwitch module has been preconfigured and is not configurable. You are also required

to configure a second directory number 5010 as a shared directory number on line 2 of each

IP phone.
http://cdn.aiotestking.com/wp-content/uploads/2011/08/CCNA-Voice-121-1.png


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Answer:
http://cdn.aiotestking.com/wp-content/uploads/2011/08/CCNA-Voice-121-2.png


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http://cdn.aiotestking.com/wp-content/uploads/2011/08/CCNA-Voice-121-b.pnghttp://cdn.aiotestking.com/wp-content/uploads/2011/08/CCNA-Voice-121-a.pnghttp://cdn.aiotestking.com/wp-content/uploads/2011/08/CCNA-Voice-121-b.pnghttp://cdn.aiotestking.com/wp-content/uploads/2011/08/CCNA-Voice-121-a.png


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(Q)You just deployed a UC500 system, you receive a support call from a user reporting that

callers are going straight to the operator instead of going to members of the hunt group.

Which two tabs that are most likely going to resolve this problem?

Answer:


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http://cdn.aiotestking.com/wp-content/uploads/2011/08/CCNA-Voice-114-a.png

ccna voice question (1).docx

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