Premium Zone Category

Composite Quizzes

January 26th, 2013 voicetut 6 comments

We have 15 composite quizzes currently, each comprises of 50 questions of all topics of CCNA Voice – ICOMM exam. If you can pass all of them then congratulation, you have full knowledge for the CCNA Voice exam.

Note: I really hope you clearly understand the concept behind each question, not learn by heart the answers.

+ Composite Quiz 1
+ Composite Quiz 2
+ Composite Quiz 3
+ Composite Quiz 4
+ Composite Quiz 5
+ Composite Quiz 6
+ Composite Quiz 7
+ Composite Quiz 8
+ Composite Quiz 9
+ Composite Quiz 10
+ Composite Quiz 11
+ Composite Quiz 12
+ Composite Quiz 13
+ Composite Quiz 14
+ Composite Quiz 15

Note: These quizzes included all New Updated Questions.

Welcome Premium Member!

January 26th, 2013 voicetut 10 comments

Welcome to Premium Member zone. Now you can access to all the resources for learning CCNA Voice – ICOMM on this site:

+ Flash-based questions to check your knowledge before each topic of CCNA Voice. You will find a link to each test before each topic at the right-side menu.
+ Composite Flash-based questions to test your whole CCNA Voice knowledge. This helps you fully prepare for the CCNA exam. You can find all the composite quizzes here.

Also if you have any questions please feel free to comment here or send us an email to support@voicetut.com.

Note: The CCNA Voice ICOMM exam does not have any simulation questions.

New Updated Questions 6

January 26th, 2013 voicetut 2 comments

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

Question 1

When you configure a shared extension in Cisco Configuration Professional, which two characteristics should be the same on both phones, so that the phones can successfully use a shared extension? (Choose two)

A. monitor mode
B. MAC address
C. ephone-dn
D. number
E. user ID

 

Answer: C D

Explanation

A shared ephone-dn has the following characteristics:
+ Appears on two different phones but uses the same ephone-dn and number
+ Can make one call at a time between the two phones, and that call appears on both phones
+ Should be used when you want the capability to answer or pick up a call at more than one phone
+ Only one ephone can pick up the call, ensuring privacy
+ When the call is placed on hold, either ephone can retrieve the call

If the ephone-dn is connected to a call on one phone, that ephone-dn is unavailable for other calls on the second phone because these phones share the same ephone-dn. If a call is placed on hold on one phone, it can be retrieved on the second phone.

shared_ephone-dn.jpg

The simplest configuration (although not optimal) of this example is shown below:

CME(config)#ephone-dn 1
CME(config-ephone-dn)#number 1001
CME(config)#ephone-dn 2
CME(config-ephone-dn)#number 1001
CME(config)#ephone 1
CME(config-ephone)#mac-address 000F.4356.AA13
CME(config-ephone)#button 1:1
CME(config)#ephone 2
CME(config-ephone)#mac-address 000F.4356.AA14
CME(config-ephone)#button 1:1

You can download a good Ephone and Ephone-dn guide here: http://www.voicetut.com/download/shared extension_Ephonedn_and_Ephone.pdf.

Question 2

An engineer is configuring a new Cisco Unified Communications Manager server. However, when the engineer tries to register the IP phones, the registrations are unsuccessful. When the engineer checks one of the phones, there is no status that is shown along with the Cisco Unified Communications Manager server IP address.
What is the probable cause?

A. The server connection is established, but the information is encrypted.
B. There is no current connection with the Cisco Unified Communications Manager server.
C. The Cisco Unified Communications Manager server is currently available.
D. The currently receiving call-processing services are running on the phone.

 

Answer: B

Question 3

Which option should you use in the Cisco Unified Communications Manager End Users Configuration page to ensure that a user can use a desk phone both for calls and for Cisco Unified Presence?

A. Enable Cisco Unified Presence Communicator
B. Allow Control of Device from Cisco Computer Telephony Integration
C. Allow Cisco Unified Personal Communicator Integration
D. Allow Cisco Unified Presence Control over IP Phone

 

Answer: B

Explanation

The CTI (Computer Telephony Integration) interface handles all the CTI communication for users on the Cisco Unified Presence server to control phones on Cisco Unified Communications Manager. The CTI functionality allows users of the Cisco Unified Personal Communicator client to run the application in desk phone control mode.

To allow the phone to interoperate with the Cisco Unified Personal Communicator client, check the option “Allow Control of Device” from CTI (Menu path: Cisco Unified Communications Manager Administration > Device > Phone).

Allow_Control_of_Device_ CTI.jpg

Question 4

Which Cisco Unity Connection report provides a summary view of the current size, last error condition, and status of the mailbox store?

A. Users
B. Message Traffic
C. Mailbox Store
D. System Configuration

 

Answer: C

Explanation

The Mailbox Store report includes the following information:

+ Mail database name
+ Display name
+ Server name
+ Whether access is enabled
+ Mailbox store size
+ Last error
+ Status
+ Whether the mail database can be deleted

(Reference: http://www.cisco.com/en/US/docs/voice_ip_comm/connection/8x/administration/guide/8xcucsag300.html)

Question 5

Which user parameter is used as a password when connecting through the telephony user interface?

A. PIN
B. Passphrase
C. PassKey
D. Key ID

 

Answer: A

Explanation

Unity Express users, known as subscribers, are individual accounts that are created to provide personal mailbox accounts for voice mail storage. These individual mailbox accounts are known as subscriber accounts. Owners of these accounts can customize their mailboxes to suit their needs.

Each subscriber can be assigned a username, PIN, and password. These credentials allow a user to manage their account. The personal identification number (PIN) is used when the subscriber manages their mailbox using the telephone user interface (TUI). The username and password are used when the subscriber manages their account using the web GUI interface or other email access protocols.

(Reference: CCNA Voice Study Guide)

Question 6

What is the quickest way to test the Cisco Unified Communications Manager configuration part of MWI to see if MWI On and MWI Off is working?

A. Dial into Cisco Unity Connection from an IP phone. Enter the MWI On numbers, then enter the MWI Off numbers.
B. Call a voice-mail user and ask them if their MWI light is on, and then disconnect the call. Call the user back and ask if the MWI light is off.
C. In Unity Connection, issue the MWI Flash command to turn all MWI lights on, then off.
D. If MWI numbers are dialable from an IP phone, dial the MWI On number. If the light comes on, then dial the MWI Off number to see if the light goes off.
E. MWI cannot be tested directly from the Cisco Unified Communications Manager or an IP phone.

 

Answer: D

Explanation

An Message Waiting Indicator (MWI) is a lamp, flashing LCD panel, or special dial tone on user phones that lets users know a voice message is waiting. The type of indicator depends on the phone system and the user phones. Phone systems that support message counts may also display the number of messages that the user has.

MWIs are not the same as message notification, which is the feature that notifies a user of new voice messages by calling a phone, pager, or other device, or by sending an email message.

The following events trigger Cisco Unity Connection to turn MWIs on and off:
+ When a message for a user arrives on the Connection message store, Connection notifies the phone system to turn on an MWI on the phone for that user.
+ Any message that arrives on the Connection message store (for example, voice messages, emails, and faxes) trigger turning MWIs on and off.
+ When the user listens to the message, Connection notifies the phone system to turn off the MWI on the phone.
+ When the user saves a listened-to message as a new message, Connection notifies the phone system to turn on the MWI on the phone for that user.
+ When a user deletes a new message without listening to it, Connection notifies the phone system to turn off the MWI on the phone.
+ When MWIs are synchronized, Connection queries the message store to determine the status of MWIs on all phones, and resets the applicable MWIs.

We can configure numbers to turn MWI On or Off. To configure the MWI On and Off Extensions for Port Groups (SCCP Integrations Only), following these steps:
Step 1: In Cisco Unity Connection Administration, expand Telephony Integrations, then click Port Group.
Step 2: On the Search Port Groups page, click the name of the first port group for the SCCP integration.
Step 3: On the Port Group Basics page, under Message Waiting Indicator Settings, in the MWI On Extension field, confirm that the extension for turning on MWIs is entered. If the field is blank, enter the MWI On extension.
Step 4: In the MWI Off Extension field, confirm that the extension for turning off MWIs is entered. If the field is blank, enter the MWI Off extension.
Step 5: Click Save.
Step 6: Click Next.
Step 7: Repeat Step 3 through Step 5 for the remaining port groups in the SCCP integration.

(Reference: http://www.cisco.com/en/US/docs/voice_ip_comm/connection/7x/troubleshooting/guide/7xcuctsg050.html)

Question 7

Which report can you use to display all currently unassigned directory numbers?

A. Route Plan Report
B. Directory Number Assignment
C. Unassigned Objects Report
D. IP Phone Number Assignment

 

Answer: A

Explanation

The route plan report lists all assigned and unassigned directory numbers (DN), call park numbers, call pickup numbers, conference numbers, route patterns, translation patterns, message-waiting indicators, voice mail ports, and Cisco CallManager Attendant Console pilot numbers in the system. The route plan report allows you to view either a partial or full list and to go directly to the associated configuration windows by clicking the Pattern/Directory Number, Partition, or Route Detail fields.

In addition, the route plan report allows you to save report data into a .csv file that you can import into other applications. The .csv file contains more detailed information than the web pages, including directory numbers for phones, route patterns, pattern usage, device name, and device description.

(Reference: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/4_1_3/ccmcfg/b03rtrep.html)

Question 8

Which statement about Cisco Unified Presence and LDAP is true?

A. Cisco Unified Presence must be integrated with LDAP.
B. LDAP integration is mandatory if Cisco Unified Personal Communicator is used as a client.
C. LDAP integration with Cisco Unified Presence is optional. However, user search functionality in Cisco Unified Personal Communicator will not be available.
D. LDAP integration with Cisco Unified Presence is optional. However, instant messaging functionality in Cisco Unified Personal Communicator will not be available.

 

Answer: C

Explanation

LDAP is a standards-based (with some significant vendor-specific exceptions) system that allows an organization to create a single, centralized directory information store. LDAP holds information about user accounts, passwords, and user privileges. The information centralized in LDAP is available to other applications, so that separate directories do not need to be maintained for each application. Using LDAP simplifies user administration, and makes using systems slightly easier for users because they only need to maintain their information and passwords in one place. Some popular LDAP systems are Microsoft Active Directory, Microsoft Lightweight Directory Services, iPlanet Directory Server, Oen LDAP, Sun ONE Directory Server…

CUCM can interact with LDAP in two ways: LDAP Synchronization and LDAP Authentication. Implementing LDAP Synchronization (LDAP Sync) means that some user data (but not all) is maintained in LDAP and replicated to the CUCM database. An advantage of LDAP Synchronization is to help administrator import all users in the LDAP Server (like Active Directory) instead of manually adding every single user in CUCM.

LDAP Authentication redirects password authentication requests from CUCM to the LDAP system. End-User account passwords are maintained in the LDAP system and are not configured, stored, or replicated to CUCM. Because one of the benefits (particularly to the End User) of LDAP is a centralized password system (making single sign-on possible), it is typical and desirable to implement LDAP Authentication with LDAP Sync.

The search functionality in Cisco Unified Personal Communicator is only available when LDAP is integrated with Cisco Unified Presence.

(Reference: CCNA Voice 640-461 Official Certification Guide – Page 258)

For more information please read: http://www.cisco.com/en/US/docs/voice_ip_comm/cups/8_0/english/install_upgrade/deployment/guide/dgldap.html#wp1099163 and http://docwiki.cisco.com/wiki/Cisco_Unified_Presence,_Release_7.x_–_Prerequisites_for_Integrating_the_LDAP_Directory.

———————————– This is the end of this update wave ———————————–

New Updated Questions 5

January 26th, 2013 voicetut 5 comments

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

Question 1

Which license capability must be enabled for Cisco Unified Presence to work with a specific user?

A. Enable Cisco Unified Presence
B. Enable Cisco Unified Presence Server
C. Enable Cisco Unified Presence Communicator
D. Enable Extensible Messaging and Presence Protocol

 

Answer: A

Explanation

Immediately following a fresh installation of Cisco Unified Presence, a 90-day trial evaluation period starts by default. During this time,
+ an organization can use or “run” a Cisco Unified Presence server without requiring a server license, and
+ users in that organization, who you have already configured on Cisco Unified Communications Manager, can access Cisco Unified Presence and you can configure these users to use Cisco Unified Personal Communicator, without requiring the necessary user licenses (DLUs)
After the trial evaluation period ends, users no longer have access to Cisco Unified Presence functionality. You must upload the server license file, and the required user licenses, to enable Cisco Unified Presence in permanent Production mode.

(Reference: http://www.cisco.com/en/US/docs/voice_ip_comm/cups/8_0/english/install_upgrade/deployment/guide/dglic.html)

Question 2

Can an IP phone be configured so that calls to that IP phone can be rerouted if the IP phone become unregistered?

A. Yes, configuring call forward All provides this coverage
B. No, if a phone is unregistered, the re-order tone is played to the caller
C. Yes, configuring Call Forward Busy Internal provides this coverage
D. Yes, configuring Call Forward Unregistered provides this coverage
E. No. the call is dropped at the gateway

 

Answer: D

Explanation

The Call Forward Unregistered (CFU) feature allows you to forward a call to a different number if the directory number (DN) is not associated with a phone or if the associated phone is not registered to Cisco Unified CME.

Note: Call Forward Busy allows incoming calls to be rerouted automatically to another phone when your line is busy.

(Reference: http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmetrans.html)

Question 3

When you reset an IP phone via the Cisco Unified Communications Manager Administration page, which method briefly shuts down a registered phone and brings it back up?

A. drop
B. restart
C. reset
D. shutdown
E. shut and no shut

 

Answer: B

Explanation

The “restart” method causes the phone to perform a warm reboot and redownload its configuration file from the TFTP server.

Note: The “reset” method is used to perform a complete reboot of an IP phone.

Question 4

Which level of users in Cisco Unified Communications Manager CDR Analysis and Reporting an generate reports for quality of service?

A. administrators
B. managers
C. auditors
D. individual users

 

Answer: B

Explanation

CAR provides reporting capabilities for three levels of users:
+ Administrators – Generate system reports to help with load balancing, system performance, and troubleshooting.
+ Managers – Generate reports for users, departments, and QoS to help with call monitoring for budgeting or security purposes and for determining the voice quality of the calls.
+ Individual users – Generate a billing report for their calls.

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

Question 5

When creating a Cisco Unity Connection user template, which element should you configure to automatically play a “This department is closed” message at specific hours?

A. greeting schedule
B. extension greetings
C. schedule
D. active schedule

 

Answer: D

Explanation

Active schedule can be used to play a greeting as well as specify the action that Cisco Unity Connection takes after the greeting. We should choose “Closed” greeting to play during the closed hours. A closed greeting overrides the standard greeting, and thus limits the standard greeting to the open hours defined for the active schedule.

Question 6

Which type of ephone-dn is typically used for intercoms and paging?

A. dual-line
B. multiline
C. single-line
D. special-line

 

Answer: C

Explanation

You can configure each ephone-dn you create as either a single- or dual-line mode ephone-dn. Here’s the difference:
+ Single-line ephone-dn: In single-line mode, the ephone-dn is able to make or receive only one call at a time. If a call arrives on an ephone-dn where there is already an active call, the caller receives a busy signal.
+ Dual-line ephone-dn:In dual-line mode, the ephone-dn is able to handle two simultaneous calls. This is useful for supporting features like call waiting, conference calling, and consultative transfers.

In most network environments, dual-line configurations are useful for user IP phones, whereas single-line configurations are useful for network functions (such as intercom or paging).

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

Question 7

Which four characteristics are associated with video? (Choose four)

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

 

Answer: A C D G

Explanation

The picture below summarizes voice, data and video characteristics:

voice_data_video_characeristics.jpg

Question 8

Which protocol is being used when you are utilizing the chat feature of Cisco Unified Personal Communicator?

A. SIMPLE
B. SIP
C. XMPP
D. HTTPS

 

Answer: C

Explanation

The Extensible Messaging and Presence Protocol (XMPP) provides instant messaging, availability and roster management services.

Question 9

An administrator wants to import users using the Bulk Administration menu in Cisco Unified Communications Manager Administration. Which file format is valid for this operation?

A. .PDF
B. .DOC
C. .DOCX
D. .CSV
E. .XLS

 

Answer: D

Explanation

The Cisco Unity Connection Bulk Administration Tool (BAT) allows you to create, update, and delete multiple user accounts or system contacts by importing information contained in a comma separated value (CSV) file. In addition, it allows you to export information about users or system contacts from Cisco Unity Connection to a CSV file. When Cisco Unity Connection is running as part of Cisco Unified Communications Manager Business Edition (CMBE), you cannot create, update, or delete users with BAT. Modifications to users must be done in Cisco Unified Communications Manager Administration.

CSV is a common text file format for moving data from one data store to another. For example, importing from a CSV file can be useful for transferring information from a corporate directory to Cisco Unity Connection. Transferring the information allows users with voice mailboxes to add corporate directory users who are not Connection users to their address books and to then create call-routing rules based on calls from such contacts.

(Reference: http://www.cisco.com/en/US/docs/voice_ip_comm/connection/7x/user_mac/guide/7xcucmacappa.html)

Question 10

Refer to the exhibit.

call_fail_troubleshooting.jpg

The user of IP phone A has opened a trouble ticket stating that he cannot call IP phone B. Where is the best place to start troubleshooting this issue?

A. IP phone B
B. the Cisco Unified Communication Manager system in site A
C. IP phone A
D. IP phone A user
E. the local VoIP voice gateway of IP phone B
F. IP phone B user

 

Answer: D

New Updated Questions 4

January 26th, 2013 voicetut 1 comment

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

Question 1

Which command is used to determine if an MGCP gateway is registered with a Cisco Unified Communications Manager server?

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

 

Answer: C

Explanation

The “show ccm-manager” command is used to indicate if the gateway is currently registered with Cisco CallManager (Cisco Unified Communications Manager). The output of this command is shown below:

Router# show ccm-manager
MGCP Domain Name: Router
Total number of host: 2
Priority Status Host
============================================================
Primary Registered 10.89.129.210
First backup Backup ready 10.89.129.211
Second backup Undefined 
Current active Call Manager: 10.89.129.210
Current backup Call Manager: 10.89.129.211
Redundant link port: 2428
Failover Interval: 30 seconds
Keepalive Interval: 15 seconds
Last keepalive sent: 1d00h (elapsed time: 00:00:03)
Last MGCP traffic time: 1d00h (elapsed time: 00:00:03)
Last switchover time: 04:49:39 from (10.89.129.211)
Switchback mode: Graceful

Question 2

Which type of packet-oriented network has the characteristic of being drop-tolerant and delay-insensitive?

A. data
B. voice
C. video
D. converged
E. All packet-oriented networks share these characteristics.

 

Answer: A

Explanation

The picture below summarizes voice, data and video characteristics:

voice_data_video_characeristics.jpg

Question 3

To monitor the service health and performance, which service should you activate in Cisco Unity Connection?

A. CUC Performance Service
B. CUC System Auditing
C. Real-Time Monitoring Tool
D. Cisco Serviceability Reporter

 

Answer: D

Explanation

CUC includes a built-in reporting system to monitor server health and performance. These reports are accessed through the Unified Service ability web application (not to be confused with the CUC Serviceability application). To access these reports, you must first activate the Cisco Serviceability Reporter service -> D is correct.

Note: The Real-Time Monitoring Tool (RTMT), which runs as a client-side application, uses HTTPS and TCP to monitor system performance, device status, device discovery, and CTI applications for Cisco Unity Connection. Therefore the correct answer may be C. Real-Time Monitoring Tool but in fact the Real-Time Monitoring Tool also requires Cisco Serviceability Reporter to be activated first.

Question 4

What will happen if an end user is deleted from the Cisco Unified Communications Manager system?

A. The user will be removed, but the associated device and directory number will remain in the system.
B. The user and the associated device will be removed, but the directory number will become orphaned.
C. The user, the associated device, and the directory number will be removed.
D. The user will be removed, and the associated device and directory number will be automatically assigned to the administrator.
E. The user will be removed, and the associated device and directory number will be allocated to the next user added to the system.

 

Answer: A

Question 5

Which statement accurately describes a calling search space?

A. a group of object with similar reach ability characteristics.
B. a calling feature that finds mobile users.
C. a toll that is used to track calls to certain numbers.
D. a feature that defines which partitions are reachable from a device.

 

Answer: D

Explanation

A calling search space comprises an ordered list of route partitions that are typically assigned to devices. Calling search spaces determine the partitions that calling devices, including IP phones, softphones, and gateways, can search when attempting to complete a call.

Note: A partition comprises a logical grouping of directory numbers (DNs) and route patterns with similar reachability characteristics.

(Reference: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/5_0_1/ccmsys/a03ptcss.html)

Question 6

What is the preferred analog signaling method to reduce glare?

A. loop start
B. ground start
C. on-hook
D. off-hook

 

Answer: B

Explanation

Glare is a phenomenon in which a user picks up a phone and connects unexpectedly to an incoming call. The best way to prevent glare is to use ground-start signaling. If you wish to learn more about loop start and ground start please read: http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a00800a6210.shtml.

Question 7

Which three rules are valid transfer rules in Cisco Unity Connection? (Choose three)

A. standard
B. alternate
C. closed
D. holiday
E. nonstandard

 

Answer: A B C

Explanation

Call transfer rules control how Cisco Unity Connection handles incoming indirect calls, which are from callers who do not dial you directly (for example, callers who use the directory to reach you). For direct calls – when outside callers or other users dial your personal phone number to reach you – your Connection transfer settings do not apply.

Standard Transfer Rule
This transfer rule is active during the business hours that your Connection administrator specified for your organization. If no other transfer rules are enabled, the standard transfer rule is active for nonbusiness hours as well.

By design, the standard transfer rule cannot be disabled.

Alternate Transfer Rule
Enable this transfer rule for a specific time period when you want to override the other transfer rules. For example, you may want to route all your calls immediately to voicemail while you are out of the office on vacation or you may want to transfer your calls to a different extension if you are temporarily working from another location.

When it is enabled, the alternate transfer rule is always active. It overrides all other transfer rules.

Closed Transfer Rule
Enable this transfer rule when you want Connection to perform different transfer actions during the nonbusiness hours that your Connection administrator specified for your organization. For example, you may want to route all your calls immediately to voicemail during nonbusiness hours.

When it is enabled, the closed transfer rule is active during nonbusiness hours.

(Reference: http://www.cisco.com/en/US/docs/voice_ip_comm/connection/7x/user/guide/phone/7xcucugphone150.html)

Question 8

Which Cisco Unified Personal Communicator mode should you use when connecting via an existing IP phone?

A. office mode
B. deskphone mode
C. softphone mode
D. IP phone mode

 

Answer: B

Explanation

The softphone mode should be used only when the IP phone is not available or the user is away from his desk. Otherwise deskphone mode should be used.

Note: Softphone is a software that functions as a telephone. Cisco Unified Personal Communicator includes a softphone.

Question 9

Which protocol should you use to securely access Cisco Configuration Professional?

A. HTTPS
B. Telnet
C. SCP
D. TLS

 

Answer: A

Question 10

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

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

 

Answer: B E G

Explanation

Same as Question 2.

New Updated Questions 3

January 26th, 2013 voicetut 3 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

New Updated Questions 2

January 26th, 2013 voicetut 1 comment

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

Question 1

Which utility should you use when you need to add a large number of users into Cisco Unified Communications Manager?

A. Cisco Unified User Administration
B. Application User CAPF Profile
C. Cisco Unified Communications Manager Bulk Administration Tool
D. Cisco Unified Telephony User Administration

 

Answer: C

Explanation

The Bulk Administration Tool (BAT) is a very powerful web-based tool of Cisco Unified Communications Manager. BAT lets you add, update, or delete a large number of similar phones, users, or ports at the same time. When you use Cisco CallManager Administration, each database transaction requires an individual manual operation, while BAT automates the process and achieves faster add, update, and delete operations.

(Reference: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/bat/5_1_4/bat_ovr.html)

Question 2

An organization is using a centralized DHCP server for all IP phones. However, when the IP phones are powered up, the phones are unable to obtain an IP address. Which CLI command should be in the router configuration to ensure that the IP phones are able to reach the DHCP server?

A. router(config)#helper-address
B. router(config-if)#ip helper-address
C. router(config-if)#helper-address
D. router(config)#ip helper-address

 

Answer: B

Explanation

The full syntax of this command is:

router(config-if)#ip helper-address {address-of-DHCP-server}

Let’s consider an example of how to use this command with the topology below:

IP_helper-address.jpg

Suppose there is only one PC on the subnet attached with E0 interface of the router. First, PC A needs an IP address so it will send a broadcast DHCP Request. When the router receives the DHCP Request, it changes the destination IP address of the packet to the value set with the ip helper-address command. Because the PC does not yet have an IP address, the DHCP request (as sent by PC A) has an source IP address of 0.0.0.0. The router then changes the source IP address so that the DHCP Response packet can be routed back to the original subnet, and then broadcast back onto that subnet.

Question 3

With GETVPN, if a key server is configured to use multicast as the rekey transport mechanism, then under which of these conditions will the key server retransmit the rekey messages?

A. it never retransmit the rekey messages
B. it only retransmit the rekey message when it does not receive the rekey acknowledgment from at least one group member
C. it only retransmit the rekey message when it does not receive the rekey acknowledgment from all group member
D. it only retransmit the rekey message when DPD to the group members fails
E. it always retransmit the rekey message

 

Answer: E

Explanation

GETVPN combines the keying protocol Group Domain of Interpretation (GDOI) with IP security (IPsec) encryption to provide users with an efficient method to secure IP multicast traffic or unicast traffic. A GETVPN deployment has primarily three components: Key Server, Group Member and Group Domain of Interpretation (GDOI) protocol.
+ Group Members do encrypt/decrypt the traffic.
+ The responsibilities of the Key Server include maintaining the policy and creating and maintaining the keys for the group and distributing the encryption key to all the group members.. When a group member registers, the key server downloads this policy and the keys to the group member. The Key Server also rekeys the group before existing keys expire.
+ GDOI protocol is used between the Group Member and Key Server for group key and group security association (SA, also mean IPSec) management. Minimum one KS is required for a GETVPN deployment.

Since all GMs use the same key, any GM can decrypt the traffic encrypted by any other GM.

Rekey messages are sent without the use of any reliable transport. There is no efficient feedback mechanism by which receivers can indicate that they did not receive a rekey message. After key server sends out rekey messages, it does not wait to receive the rekey acknowledgment -> B and C are not correct.

Dead Peer Detection (DPD) is only used to keep track of the state of other Key Servers and it has no related with Group members or rekey message -> D is not correct.

Rekey messages are sent in advance of the SA expiration time to ensure that valid group keys are always available. Also after the registration is successful, the key server sends a multicast rekey to all the group members that have registered within a group -> E is correct but A is not.

(Good resource and reference:
+ http://www.cisco.com/en/US/prod/collateral/iosswrel/ps6537/ps6586/ps6635/ps7180/deployment_guide_c07_554713.html
+ http://www.cisco.com/en/US/docs/ios-xml/ios/sec_conn_getvpn/configuration/15-2mt/sec-get-vpn.html#GUID-A2442E2A-4B03-4692-9EBE-98BEB273BCEC)

Question 4

Which two directory services are supported by Cisco Unified Communications Manager for Lightweight Directory Access Protocol integration? (Choose two)

A. Windows Active Directory 2008
B. Novell eDirectory
C. iPlanet Directory Server 4.0
D. Sun ONE Directory Server
E. Open Lightweight Directory Access Protocol 2.1

 

Answer: A D

Explanation

CUCM supports LDAP integration with several widely used LDAP systems, including the following:
+ Microsoft Active Directory (2000, 2003, 2008)
+ Microsoft Active Directory Application Mode 2003
+ Microsoft Lightweight Directory Services 2008
+ iPlanet Directory Server 5.1
+ Sun ONE Directory Server (5.2, 6.x)
+ Open LDAP (2.3.39, 2.4)

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

Question 5

Which generating QoS reports CAR tool, what two parameters are valid for report generation? (Choose two)

A. route lists
B. route patterns/hunt pilots
C. route groups
D. gateway types
E. partitions
F. IP phone directory numbers

 

Answer: B D

Explanation

+ Route Pattern: matches a set of dialed digits and triggers a call-routing process that can include one or more potential paths, providing a hierarchical set of call-routing options.
+ Hunt Pilot: a specific pattern of digits that, when matched, triggers a customizable call-coverage system.

The samples of configuring of Route Pattern and Hunt Pilot are shown below:

Route Patterns:

QoS_CAR_route_patterns_hunt_pilots.jpg

Gateway Types:

QoS_CAR_gateway_types.jpg

(Good reference: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/service/3_3_3/ccmsrva/sacar4.html)

Question 6

Which protocol should you use to configure Cisco Unified Personal Communicator for secure voice messaging with Cisco Unity Connection?

A. TCP
B. SSL
C. TLS
D. UDP

Answer: C

Explanation

Transport Layer Security (TLS) is a successor to Secure Sockets Layer (SSL) protocol. TLS provides secure communications on the Internet for such things as e-mail, Internet faxing, and other data transfers. We should use IMAP (TCP/TLS) protocol to configure CUPC for secure voice messaging with Cisco Unity Connection.

Question 7

An engineer is installing an IP phone in a remote location. When the engineer plugs the IP phone into the network, the phone does not power up. What is the first thing that should be checked?

A. Power over Ethernet switch
B. Cisco Unified Communications Manager Server
C. Cisco Unified Presence
D. DHCP server

 

Answer: A

Question 8

When performing backups within Cisco Unified Communications Manager, which component stores backups on a remote SFTP server?

A. Local Agent
B. Master Agent
C. Scheduler
D. Backup Controller

 

Answer: B

Explanation

One of the responsibilities of a Master Agent is to store backups of CUCM and CDR/CAR databasess on a local tape drive or a remote Secure FTP (SFTP) server.

Question 9

By default, how many failed attempts at signing into Cisco Unity Connection is a user allowed before their account is locked out?

A. 2
B. 3
C. 4
D. 5
E. 6
F. no limit

 

Answer: B

Explanation

By default, three filed attempts are allowed to sign into Cisco Unity Connection before their account is locked out:

CUC_failed_login.JPG

Question 10

Which Cisco IOS CLI command should you use to perform an IP phone cold reboot?

A. router(config-ephone)#reset
B. router(config-ephone)#restart
C. router(config-ephone-dn)#restart
D. router(config-phone)#reset

 

Answer: A

Explanation

The “reset” command in ephone configuration mode is used to perform a complete reboot of an IP phone.

Note: The “restart” command under ephone configuration mode causes the phone to perform a warm reboot and redownload its configuration file from the TFTP server.

New Updated Questions 1

January 26th, 2013 voicetut 4 comments

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

Question 1

When creating a new softphone for a Cisco Unified Presence user, which phone type should you select if you are using Cisco Unified Personal Communicator Release 7.1?

A. Cisco Unified Personal Communicator
B. Cisco Unified Client Services Communicator
C. Cisco Unified Client Services Framework
D. Cisco Unified Personal Communicator Framework

 

Answer: A

Explanation

Cisco Unified Personal Communicator is a multimedia application that provides your most frequently used communications applications and services: easily access soft phone, presence, instant messaging (IM), voicemail, video, and multiparty conferencing to effectively communicate and collaborate from anywhere.

Cisco Unified Presence: Provides status and reachability information for the users of the voice network. For example, Joe might check the status for Samantha and find that she is available on an instant messenger client but is currently engaged in a video call.

Note: Cisco Unified Presence is required to support the core functions for Cisco Unified Personal Communicator.

(Reference: http://www.cisco.com/en/US/docs/voice_ip_comm/cupc/8_5/english/user/guide/win/CUPC85_FAQ_chapter1.html)

Question 2

If two phones share a line between them, what happens if the user of the first phone is using the line and the user of the second phone uses the Barge feature?

A. The user of the first phone is notified of the Barge request.
B. A three-way conference begins.
C. The call is discontinued, but the first phone user and the second phone user are able to talk.
D. The user on the first phone is knocked off the line, and the second phone user takes over the line.

 

Answer: B

Explanation

Barge allows a user to get added to a remotely active call that is on a shared line. Pressing a softkey automatically adds the user (initiator) to the shared-line call (target), and the users currently on the call receive a tone (if configured). Barge supports built-in conference and shared conference bridges. If two phones have a shared line configured and one of the phones is using that line, the second phone can force a three-way conference with the first phone by using the Barge feature.

Phones support Barge in two conference modes:
+ Built-in conference bridge at the target device (the phone that is being barged). This mode uses the Barge softkey.
+ Shared conference bridge. This mode uses the cBarge softkey.

(Reference: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/4_1_2/ccmfeat/fsbarge.html and CCNA Voice 640-461 Official Certification Guide)

Question 3

Users report that when they press the Messages button on their phones, they often get a busy tone.
Which option can rectify this issue?

A. Disable the Callers Can Edit Messages checkbox under the Message settings
B. The CSS for the phones does not contain the voice-mail port partitions
C. The CSS for the phone does not contain the voice-mail pilot partition.
D. Precede all Cisco Unity Connection greetings to announce that each message is limited to 90 seconds long to free up voice-mail ports.

 

Answer: A

Explanation

By default, the option “Callers Can Edit Messages” is checked. This option allows caller the option to rerecord, append to or delete their message after they record it but it also extends the time messaging port tied up -> user may get a busy tone when they press the Messages button.

Question 4

Which web-based tool is used to access Cisco Unified Communications Manager CDR Analysis and Reporting?

A. Cisco Unified CDR Administration
B. Cisco Unified CM Administration
C. Cisco Unified Reporting
D. Cisco Unified Serviceability

 

Answer: D

Explanation

Cisco Unified Serviceability is a web-based troubleshooting tool for Cisco Unified Communications Manager. Cisco Unified Serviceability provides the following reporting tools:

+ Cisco Unified Communications Manager CDR Analysis and Reporting (CAR) – Generates Cisco Unified Communications Manager reports for Quality of Service, traffic, and billing information through Cisco Unified Communications Manager CDR Analysis and Reporting (CAR). For more information, refer to the Cisco Unified Communications Manager CDR Analysis and Reporting Administration Guide.
+ Cisco Unified Communications Manager Real-Time Monitoring Tool (RTMT) – Monitors real-time behavior of components through RTMT; creates daily reports that you can access through the Serviceability Reports Archive. For more information, refer to the Cisco Unified Communications Manager Real-Time Monitoring Tool Administration Guide.
+ Serviceability Reports Archive – Archives reports that the Cisco Serviceability Reporter service generates.
+ Cisco Unified Communications Manager Dialed Number Analyzer – Allows you to test and diagnose a deployed Cisco Unified Communications Manager dial plan configuration, analyze the test results, and use the results to tune the dial plan. For more information on how to access and use Dialed Number Analyzer, refer to the Cisco Unified Communications Manager Dialed Number Analyzer Guide.

(Reference: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/service/6_0_1/admin/saintdes.html#wp1046652)

Question 5

When deploying Cisco Unified Presence, which service is a basic service that can be considered as optional?

A. Cisco Unified Presence Engine
B. Cisco Unified Presence Sync Agent
C. Cisco Unified Presence Session Initiation Protocol Proxy
D. Cisco Unified Protocol Extensible Messaging Presence and Protocol Agent

 

Answer: D

Explanation

XMPP is an XML-based application layer protocol designed solely for instant messaging and presence management. This is only an optional service when deploying Cisco Unified Presence (CUP).

Question 6

What is the accepted maximum limit for good-quality voice connection delay?

A. 100 ms
B. 200 ms
C. 250 ms
D. 300 ms

 

Answer: C

Explanation

The generally-accepted limit for good-quality voice connection delay is 200 ms one-way (or 250 ms as a limit). As delays rise over this figure, talkers and listeners become un-synchronized, and often they speak at the same time, or both wait for the other to speak. This condition is commonly called talker overlap. While the overall voice quality is acceptable, users sometimes find the stilted nature of the conversation unacceptably annoying.

(Reference: http://www.cisco.com/en/US/tech/tk652/tk698/technologies_white_paper09186a00800a8993.shtml)

Question 7

In which mode is CTI used with Cisco Unified Personal Communicator?

A. soft-phone mode
B. desk-phone mode
C. IP communicator mode
D. IP phone mode
E. CTI mode

 

Answer: B

Explanation

CUPC operates in two modes:
+ Deskphone mode: CUPC can control the user’s deskphone to place calls.The IP phone must be registered with CUCM and associated with the user. CUPC uses Computer Telephone Integration Quick Buffer Encoding (CTIQBE) for IP phone control. In this mode, Cisco Computer Telephone Interface (CTI) is used between Cisco Unified Communications Manager and Cisco Unified Personal Communicator.

+ Softphone mode: When the IP phone is not available or the user is away from his desk, CUPC activates the associated softphone based on the Cisco Unified Client Services Framework (CSF), which registers the softphone with CUCM as a SIP device. The CUCM administrator must create the CSF device in order to enable this functionality. In this mode, SIP is used between Cisco Unified Communications Manager and Cisco Unified Personal Communicator.

(Reference: CCNA Voice 640-461 Official Certification Guide. Also you can see the picture described these two modes on page 381 of this book.)

Question 8

What are the two differences between the IntServe and DiffServ models of QoS? (Choose two)

A. DiffServ provides absolute QoS guarantees.
B. IntServ is the default QoS mechanism for all routers, because applications signal the router with the QoS level they require.
C. DiffServ inherits the connection-oriented approach from telephony network design. Every individual communication must explicitly specify its traffic descriptor and requested resources to the network.
D. With IntServ, packet delivery is guaranteed. However, the use of IntServ can severely limit the scalability of a network.
E. DiffServ was designed to overcome the limitations of both the best-effort and IntServ models and can provide an “almost guaranteed” QoS.

 

Answer: D E

Explanation

+ Integrated Services (IntServ): The IntServ model works through a method of reservations. For example, if a user wanted to make an 80Kbps VoIP call over the data network, the network designed purely to the IntServ model would reserve 80Kbps on every network device between the two VoIP endpoints using the Resource Reservation Protocol (RSVP). For the duration of the call, 80Kbps of bandwidth would not be available for any otheruse otherthan the VoIP call. Although the IntServ model is the only model that provides guaranteed bandwidth, it also has scalability issues. If enough reservations are made, the network simply runs out of bandwidth.

+ Differentiated Services (DiffServ): The DiffServ model is the most popular and flexible model to use for implementing QoS. In this model, you can configure every device to respond with a variety of QoS methods based on different traffic classes. You can specify what network traffic goes into each class and how each class is treated. Unlike the IntServ model, the traffic is not absolutely guaranteed (since the network devices do not completely reserve the bandwidth). However, DiffServ gets so close to guaranteed bandwidth (some Cisco documentation refers to it as “almost guaranteed” bandwidth), while at the same time addressing the scalability concerns of IntServ, that it has become the standard QoS model used by most organizations around the world.

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

Question 9

Which type of server is used to deliver the configuration to an IP phone?

A. TFTP
B. DHCP
C. FTP
D. Cisco Discovery Protocol

 

Answer: A

Explanation

IP Phones can download their configurations from a TFTP server.

Question 10

Which type of user in Cisco Unified Communications Manager has an interactive login?

A. administrator
B. end user
C. application user
D. phone user

 

Answer: B

Explanation

By default, on a non-integrated Cisco Unified Communications Manager (CUCM), there are two types of users: end users and application users.
+ End users – All users associated with a physical person and an interactive login. This category includes all IP Telephony users, as well as Unified CM administrators when you use the User Groups and Roles configuration (equivalent to the Cisco Multilevel Administration feature in prior Unified CM versions).
+ Application users – All users associated with other Cisco IP Communications features or applications, such as Cisco Attendant Console, Cisco IP Contact Center Express, or Cisco Unified Communications Manager Assistant. These applications need to authenticate with Unified CM, but these internal users do not have an interactive login. This serves purely for internal communications between applications, for example, CCMAdministrator, AC, JTAPI, RM, CCMQRTSecureSysUser, CCMQRTSysUser, CCMSysUser, IPMASecureSysUser, IPMASysUser, WDSecureSysUser, and WDSysUser.