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New Updated Questions 3

January 26th, 2013 in Premium Zone Go to comments

Note: You can test your knowledge with these questions first via this link.

Question 1

A user exists within an office of IP phones, all of which have the same pickup group. A call comes into a phone of another office worker who is currently at lunch. What should the user do to have the call redirected to his or her phone?

A. Press the “Pickup” softkey and then the line that is ringing.
B. Press the “Call Pickup” softkey only.
C. Press the “Call Pickup” softkey and then the line that is ringing.
D. Press the “Pickup” softkey only.

 

Answer: B

Explanation

Call pickup allows you to answer another ringing phone in the organization from your local phone. This is accomplished by pushing the PickUp softkey onthe IP phone while another phone is ringing. The call automatically transfers to the local phone, where you can answer it. Of course, the organization is large, and there could be many ringing phones at the same time, so call pickup gives you the opportunity to divide the phones into groups. You can assign each of these groups a number in the CME configuration.

(Reference: CCNA Voice 640-461 Official Certification Guide)

Question 2

Which command is useful to see if network layer information is being received at a PSTN gateway?

A. show gateway status
B. show isdn q931
C. show ccm-manager status
D. show isdn status
E. show isdn q921

 

Answer: D

Explanation

The picture below shows the output of the “show isdn status” command.

show_isdn_status.jpg

The “show isdn status” command can be used to display the status of Layer 1, 2 and 3. Make sure Layer 1 is in ACTIVE state and Layer 2 is in MULTIPLE_FRAME_ESTABLISHED state or something is wrong with these layers.For layer 3, it is the number of active calls and some other information.

Note: The “TEI_ASSIGNED” state of layer 2 indicates that the PRI is not exchanging Layer 2 frames with the switch.

(Detailed description about “show isdn status” command: http://www.cisco.com/en/US/tech/tk801/tk379/technologies_tech_note09186a0080094b78.shtml)

Question 3

What is the maximum amount of jitter that the engineer should set to maintain a high-quality call?

A. 5 ms
B. 50 ms
C. 10 ms
D. 30 ms

 

Answer: D

Explanation

Jitter is defined as a variation in the delay of received packets. At the sending side, packets are sent in a continuous stream with the packets spaced evenly apart. Due to network congestion, improper queuing, or configuration errors, this steady stream can become lumpy, or the delay between each packet can vary instead of remaining constant.

There are some recommended requirements for high-quality voice calls:

+ End-to-end (one-way) delay: 150 ms or less
+ Jitter: 30 ms or less
+ Packet loss: 1% or less

(Reference: http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a00800945df.shtml and CCNA Voice 640-461 Official Certification Guide)

Question 4

Which report in Cisco Unified Reporting should you use to track the number of users with one or more phones?

A. Unified CM User Device Count
B. Unified CM Device Distribution Summary
C. Unified CM Table Count Summary
D. Unified CM Data Summary

 

Answer: A

Explanation

Unified CM User Device Count provides information about associated devices; for example, this report lists the number of phones with no users, the number of users with one phone, and the number of users with more than one phone.

(Reference: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/service/7_1_2/report/curptg.html)

Question 5

A phone is configured with an ephone-dn number of A100. Which CLI command is used in ephone-dn configuration mode to enable the Intercom feature to dial this phone?

A. intercom extension A100
B. intercom number A100
C. intercom A100 enable
D. intercom A100

 

Answer: D

Explanation

Cisco CME supports single-button push-to-talk and push-to-respond intercom lines. You can create an intercom arrangement between any two (multiline) IP phones that support speakerphone operation. You can even operate an intercom across a VoIP connection using either SIP or H.323. Cisco CME’s intercom function is built using two functions:

+ Autodial at the initiating end of the intercom
+ Autoanswer-with-mute at the receiving end

To create an intercom you assign a line button on each of the two phones to operate as an intercom line. Pressing the intercom line button selects the line and triggers the autodial function toward the second phone. The receiving phone receives the incoming intercom call on its intercom line. This line autoanswers the call and activates the phone in speakerphone mode and sounds a beep. It also forces the speakerphone to mute to protect the privacy of the intercom recipient. The audio path is open from the initiator to the receiver. To respond to the intercom, the recipient simply presses the mute button to unmute the audio path back to the originator.

Note: The “intercom A100” should be configured on another phone than the phone configured with an ephone-dn number of A100 so that this phone can fast dial the phone configured with ephone-dn number of A100.

(Reference: http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/command/reference/cme_i1ht.html and Cisco IP Communications Express: CallManager Express with Cisco Unity Express)

Question 6

An administrator is attempting to add a new IP phone to the network. The phone does not register and continues to cycle through the registration process. The administration checks and notices that the IP address assigned to the phone is not correct network. What is the cause of this issue?

A. The TFTP server is reconfigured
B. The DHCP server is giving out false IP addresses
C. The Cisco Unified Communication Manager is down
D. The switch port that the phone is connected to is configured with the wrong voice LAN
E. The PSTN gateway is down

 

Answer: D

Explanation

The IP Phone Registration Process is shown below:

1. SCCP phone obtains the Power (PoE or AC adapter).
2. The phone loads its locally stored firmware image.
3. The phone learns the Voice VLAN ID via CDP from the switch.
4. The phone uses DHCP to learn its IP address, subnet mask, default gateway and TFTP server address.
5. The phone contacts the TFTP server and requests its configuration file. Each phone has a customized configuration file named SEP.cnf.xml created by CUCM and uploaded to TFTP when the administrator creates or modifies the phone.
6. The phone registers with the primary CUCM server listed in its configuration file. CUCM then sends the softkey template to the phone using SCCP messages.

As the question stated “The administration checks and notices that the IP address assigned to the phone is not correct network” but the IP Phone’s IP address is learned automatically via DHCP so maybe the problem here is the phone was learned an incorrect Voice VLAN ID, causing wrong IP address assignment -> D is correct.

Answer B “The DHCP server is giving out false IP addresses” may be correct but it is difficult to happen because this is an operating system and everything is working fine.

(Reference: https://supportforums.cisco.com/docs/DOC-21496)

The picture below shows a quick summary how IP Phone operates with a switch, just for your reference:

Cisco_IP_Phone_data_voice_VLANs.jpg

Question 7

Which device would allow you to place calls from a Cisco Unified Communications Manager that is configured with SCCP phones to a Cisco Unified Communications Manager Express that is configured with SIP phones?

A. gatekeeper
B. gateway
C. H.323 trunk
D. SIP trunk
E. Cisco Unified Border Element

 

Answer: E

Explanation

SCCP uses Cisco-proprietary messages to communicate between IP devices and Cisco Unified Communications Manager. It needs a Session Border Controller (SBC) device like Cisco Unified Border Element to communicate with other protocols like SIP or H.323.

Note: There are many functions that a SBC can do but I want to refer it as a signaling translation device (to communicate between SIP, H.323) and provide us some more security to our local network.

Question 8

Which three characteristics are associated with voice? (Choose three)

A. greedy
B. TCP retransmits
C. UDP priority
D. delay sensitive
E. drop insensitive
F. benign
G. benign or greedy

 

Answer: C D F

Explanation

There are five requirements for voice traffic, they are:
+ Smooth (not bursty)
+ Benign
+ Drop Sensitive (less than 1%)
+ Delay Sensitive (no more than 150 ms for one-way)
+ UDP Priority

Question 9

Which Cisco IOS command should you use to view the configuration of voice dial peer 911?

A. show dialplan dialpeer 911
B. show dialplan number 911
C. show dial-peer voice 911
D. show event-manager consumers 911

 

Answer: C

Comments (3) Comments
  1. Watcher
    December 13th, 2013

    WARNING!!!!!!!!
    Cisco has changed the exam!
    All new questions and the minimum score to pass is now 880/1000.
    Took the test today and failed.
    🙁

  2. caltex67
    December 17th, 2013

    Hi Watcher,

    are they these new questions or newer questions than the ones here?

  3. Anonymous
    May 8th, 2014

    on question 1 the answer based on the explanation should be “pickup” soft key and not “Call pickup ” soft key. Could you please clarify this.

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