Please share with us your experience after taking the CAPPS v8.0 exam, your materials, the way you learned, your recommendations…

Please share with us your experience after taking the TVoice v8.0 exam, your materials, the way you learned, your recommendations…

Please share with us your experience after taking the CVoice v8.0 exam, your materials, the way you learned, your recommendations…

Please share with us your experience after taking the CIPT2 v8.0 exam, your materials, the way you learned, your recommendations…

Please share with us your experience after taking the CIPT1 v8.0 exam, your materials, the way you learned, your recommendations…

Question 1
Which issue does CAS signaling on a T1 circuit create?

A. Signaling bits are subtracted from each frame, which causes a significant loss of voice quality.
B. An extra signaling bit is added to the sixth frame to carry signaling information.
C. A signaling bit is subtracted from every sixth frame to carry signaling information.
D. Signaling bits are added to the signaling stream to create extended super frames.

Answer: C

Question 2

Which network component would cause variable network delay?

A. dejitter buffer
B. DSP delay
C. processing delay
D. serialization delay
E. propagation delay

Answer: E

Question 3

Which headers are compressed when cRTP is used?

A. the UDP and RTP headers
B. the IP header only
C. the TCP header only
D. the RTP header only
E. the IP, UDP, RTP headers and the first byte of the payload
F. the IP, UDP, and RTP headers

Answer: F

Question 1

Click and drag the items on the left into the correct order on the right for a systematic troubleshooting method

Answer:

Start -> Define Problem -> Gather Facts -> Consider Possibilities -> Create Action Plan -> Implement Action Plan -> Observe Results -> Ultilize Process

Do Problem Stop? -> Problem Resolved -> Document Facts -> Finished

Question 1

Which description describes the low latency queuing algorithm?

A. Empty queue 1. If queue 1 is empty, empty queue 2, then empty queue 3, unless a packet for a higher queue arrives.
B. An administrator defines the traffic classes based on match criteria, including protocols, access control lists, and input interfaces.
C. A flow-based algorithm that simultaneously schedules interactive traffic to the front of a queue to reduce response time and fairly shares the remaining bandwidth among high-bandwidth flows.
D. This feature brings strict priority queuing to CBWFQ.
E. Packets are placed into a single queue and serviced in the order they were received.

Answer: D

Question 2

Which description describes the weighted fair queuing algorithm?

A. Empty queue 1. If queue 1 is empty, empty queue 2, then empty queue 3, unless a packet for a higher queue arrives.
B. An administrator defines the traffic classes based on match criteria, including protocols, access control lists, and input interfaces.
C. A flow-based algorithm that simultaneously schedules interactive traffic to the front of a queue to reduce response time and fairly shares the remaining bandwidth among high-bandwidth flows.
D. This feature brings strict priority queuing to CBWFQ.
E. Packets are placed into a single queue and serviced in the order they were received.

Answer: C

Question 1
Users report that all external callers are leaving urgent voice-mail messages. Where can this behavior be changed?

A. under the Phone Menu Configuration > Unidentified Callers Message Urgency
B. under the Opening Greeting > Unidentified Callers Message Urgency
C. under the Message Settings > Unidentified Callers Message Urgency
D. under the System Call Handlers > Unidentified Callers Message Urgency
E. under the Voice-mail Box Settings > Unidentified Callers Message Urgency

Answer: C

Question 2
Which two fields are required parameters when manually creating users on Cisco Unity Connection with predefined templates? (Choose two)

A. username (alias)
B. extension
C. first name and last name
D. employee ID
E. title

Answer: A B

Question 3
Which two options allow the maximum message length to be adjusted in Cisco Unity Connection? (Choose two)

A. Message Settings under individual users
B. User Templates >VoiceMailUserTemplate > Message Settings
C. Contacts > Message Settings
D. Enterprise Parameters > Maximum Message length
E. Service Parameters > Voicemail Settings

Answer: A B

Question 4

Refer to the exhibit. Where can the phone menu be accessed from?

A. from the individual users or user templates
B. from the Class of Service configuration screen
C. from the user contacts
D. from the Interview Handler configuration screen
E. from the Message Store configuration screen

Answer: A

Question 5
Which tools allow the administrator to migrate users from Cisco Unity to Cisco Unity Connection?

A. Cisco Object Backup and Restore Application Suite
B. Cisco Disaster Recovery Framework Tool
C. Cisco Real Time Monitoring Tool
D. Cisco Unity Serviceability Tool

Answer: A

Question 6
Which three options are valid for creating users in Cisco Unity Connection? (Choose three)

A. manual creation
B. bulk using *.csv file
C. bulk using enterprise parameters
D. Cisco Unity Connection Serviceability
E. automatic creation through TUI by users dialing into voice mail
F. import through Active Directory

Answer: A B F

Question 7
Which statement about Cisco Unity Connection user templates is true?

A. Changes in user templates affect only new users to be created.
B. Changes in user templates affect only existing users.
C. Changes in user templates affect new and existing users.
D. Changes in user templates have no impact on users unless those users are imported through Active Directory.

Answer: A

Question 1
When troubleshooting a phone that is unable to get an IP address from a DHCP server, what is the first thing to check for on the phone?

A. Make sure that DHCP Enabled is disabled on the phone.
B. Make sure that the phone is getting the proper VLAN information.
C. Make sure that the TFTP server address is correct on the phone.
D. Make sure that the DHCP scope has enough addresses left in the range.
E. Make sure the phone has the correct phone load ID.

Answer: B

Question 2

An IP phone has a line calling search space and a device calling search space. If a call is made from the IP phone, which calling search space is used?

A. Neither calling search space is used.
B. The line calling search space takes precedence and is used.
C. The device calling search space takes precedence and is used.
D. The line and device calling search spaces are combined and the line calling search space has precedence.

Answer: D

Question 3
In which two ways can an administrator reset an IP phone that is registered with Cisco Unified Communications Manager? (Choose two)

A. Phone can be reset in Cisco Unified Communications Manager Administration.
B. Phone can be reset in the Cisco Unified Communications Manager Express CLI.
C. Press the * * #* * key combination on the IP phone keypad.
D. Enter the reset ephone command in the switch.
E. Press the * * #* key combination on the IP phone keypad.
F. Press the ##**# key combination on the IP phone keypad.

Answer: A C

Question 4
After changes are made to an IP phone, which reset method is the fastest to bring the phone back into service?

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

Answer: B

Question 5
How can an administrator determine which codec is being used between two endpoints while a call is in progress?
A. Run the codec trace in Cisco Unified Communication Manager.

B. Use Cisco Unified Serviceability network trace.
C. Can only be seen in Cisco SDI traces.
D. Can only be seen in a sniffer trace.
E. Pressthe ? button twice on one of the IP phones.

Answer: E

Question 6

An administrator wants to see how long a specific IP phone extension is in use during a given period of time. What CAR tool feature would the administrator use?

A. System Reports > Traffic > Summary by Phone Number
B. CDR > Search > By Call Precedence Level
C. Device Reports > Route Patterns/Hunt Groups > Route and Line Group Utilization
D. User Reports > Top N > By Duration

Answer: A

Question 7
After an IP phone has been added to a Cisco Unified Communications Manager server, the administrator notices that the phone has a directory number that is not in the number range in use by the organization. He also notes that the directory number is 1000. What is the most likely cause?

A. The phone number was misconfigured.
B. The phone may have auto-registered.
C. The phone is configured on another server.
D. DHCP gave the phone the wrong directory number
E. TFTP server is misconfigured

Answer: B

Question 8
All users report that when they press the Messages button on their IP phone, nothing happens. What is the most likely cause of this issue?

A. Cisco Unity Connection is not configured.
B. The default Voicemail Profile does not have a Pilot number configured.
C. The Voicemail Pilot does not have the VoiceMail Profile configured.
D. The Integrated Service Engine is offline.
E. The Voicemail Pilot is incorrect.

Answer: B

Question 9
Which type of single switch port can support data and voice VLANs and is recommended for Cisco Unified IP phones?

A. multiflex port
B. trunk port
C. access port
D. ISL trunking port

Answer: C

Question 10
When adding an IP phone to a Cisco Unified Communications Manager system, what two choices does an administrator have to add the phone to the system? (Choose two)

A. auto-registration
B. FHSS provisioning
C. IP phone configuration assistant
D. manual provisioning
E. Cisco Unified Serviceability

Answer: A D

Question 11
On an IP phone line appearance, for which two purposes is the Display parameter designed to be used? (Choose two)

A. display a number other than the directory number of the IP phone
B. display a name instead of the directory number of the IP phone
C. internal caller ID
D. external caller ID
E. full directory number ID for outgoing calls

Answer: B C

Question 1
An administrator wants to locate and remove all unassigned directory numbers on the Cisco Unified Communications Manager system. Which method is the best method to accomplish this task?

A. Choose Device > Phone. Search all phones and remove the undesired directory numbers.
B. Use the Dial Plan Installer to remove the directory numbers.
C. Use the Disaster Recovery System to restore only valid directory numbers.
D. Choose Call Routing > Route Plan Report, choose the Unassigned DN drop-down menu, and then remove all orphaned directory numbers.
E. Choose Device > Device Settings > Device Defaults and use the wizard to locate and remove the orphaned directory numbers.

Answer: D

Question 2
How does the Cisco Unified Communications Manager Express match an outbound VoIP dial peer?

A. Outbound dial-peer matching uses the entire string of digits en bloc to match the dial peer with the longest match.
B. Outbound dial-peer matching is completed on a digit-by-digit basis.
C. It matches outbound dial peers by placing all the dial peers into a hunt group and then uses the entire dialed number en bloc to match the first dial peer.
D. Cisco Unified Communications Manager Express creates a hunt group that contains all the configured dial peers and then applies the dial digits in a digit-by-digit manner to match a dial peer.

Answer: A

Question 3
In which situation are user PINs maintained by the local Cisco Unified Communications servers?

A. Only when the system is using LDAP synchronization.
B. Only when the system is using LDAP authentication.
C. Only when the system is using Global Directory.
D. PINs are always maintained by the local Cisco Unified Communications Manager servers
E. PINs are never maintained by the local Cisco Unified Communications Manager servers.

Answer: D

Question 4
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. showccm-manager
D. show isdn status
E. show isdn q921

Answer: C

Question 5
An administrator is attempting to add a new user in Cisco Unified Communications Manager Administration but does not see the option to add a new user. What is the most likely cause of this issue?

A. The SQL User database is not running.
B. The system is synchronized with an LDAP server.
C. BAT is not enabled.
D. The administrator hasthe aceno user adda rights box checked.
E. The SIP Realm is not defined in User Management.

Answer: B

Question 6
Which call processing agent is based on Cisco IOS software and works with ISR platforms?

A. Cisco Unified Presence Server
B. Cisco Unity Connection
C. Cisco Unified Communications Manager Express
D. Cisco Unified Communications Manager
E. Cisco UnifiedContact Center Express

Answer: C

Question 7

A user in Cisco Unified Communications Manager Administration has been added to the Standard CCM Admin Users group, which includes the Standard CCMADMIN Administration role, but the user cannot add new users. What is the cause of this issue?

A. The add user capability has been disabled for the group
B. The incorrect group and role were assigned.
C. The add user capability has been disabled for the role.
D. Only theCCMAdmin user can add users.
E. Users can be added only via LDAP

Answer: C

Question 8
A new phone has been added to the Cisco Unified Communications Manager server. The phone display shows Your Current Options, but when the New Call softkey is pressed, no dial tone is heard and the call cannot be placed. What could be the cause of this issue?

A. An incorrect MAC address has been entered for the new phone.
B. No directory number has been assigned to a line.
C. The end user is not associated with the device.
D. No calling search space has been configured on the line.
E. An incorrect device pool has been configured on the phone

Answer: B

Question 9
What Cisco client application allows administrators to interact with performance monitoring counters to assist in determining the overall health of the Cisco Unified Communications Manager server?

A. Cisco Unified Communications Manager Administration
B. Cisco Unified Real-Time Monitoring Tool
C. Cisco Unified OS Administration
D. CAR Tool
E. BAT Tool

Answer: B

Question 10

Which report can be generated by using the User Reports feature of the CAR tool?

A. Traffic
B. Top N
C. Malicious Call Details
D. CDR Error
E. FAC/CMC

Answer: B

Question 1

In which two locations can an end user configure their Call Forward All settings? (Choose two)

A. Cisco Unified Serviceability
B. Cisco Unified Communications Manager User Options Interface
C. Directly on the Cisco Unified IP phone
D. Cisco Unified Communications Manager Administration
E. Cisco Unified User Serviceability

Answer: B C

Question 2
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 aredialable 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

Question 3

To obtain CDR information, which application is used?

A. Cisco Unified Communications Manager Administration
B. Cisco Unified Serviceability
C. Cisco Unified Operating System Administration
D. Disaster Recovery System
E. Cisco Unified Communications Manager Call Detail Record Analysis and Reporting tool
F. Cisco Unified Reporting

Answer: E

Question 4
What component does Cisco Unified Communications Manager Express use to match outbound dial peers?

A. destination pattern
B. incoming called-number
C. calling number ANI
D. answer-address
E. port or session target

Answer: A

Question 5
In Cisco Unified Communications Manager Native Presence, what two things is a watcher monitoring in real time? (Choose two.)

A. registration status of a specific IP phone
B. registration status of the hunt group
C. registration status of the MGCP gateway
D. registration status of Cisco Extension Mobility of the IP phone
E. status of a registered directory number

Answer: A C

Question 6
Which protocol is used for communication between Cisco Unity Express and Cisco Unified Communications Manager Express?

A. H.323
B. G.711
C. MGCP
D. Q.931
E. SIP

Answer: E

Question 7

What is the correct URL to use to add a user to Cisco Unified Communications Manager Express?

A. http://ipaddress/cucme.html
B. https://ipaddress/cue.html
C. httpd://www.ipaddress/cme.html
D. http://ipaddress/ccme.html
E. http://ipaddress/cme.html

Answer: D

Question 8
What is the Cisco Unified Communications Manager implementation of one-way intercom referred to as?

A. One-Way Intercom monitor
B. Whisper Intercom
C. Secure Intercom
D. Silent monitor

Answer: B

Question 9
Which configuration must be configured on the Cisco Unified Communications Manager Express
router prior to Graphical Administration?

A.
ip http server

telephony-service
web admin system name admin password cisco
dn-webedit
time-webedit

B.
ip http server
ip http path flash:path

telephony-service
web admin system name admin password cisco
dn-webedit
time-webedit

C.
ip https server
ip https path flash:path

telephony-service
web admin system name admin password cisco
dn-webedit
time-webedit

D.

ip http server
ip http path flash:path

telephony-service
dn-webedit
time-webedit

A. Exhibit A
B. Exhibit B
C. Exhibit C
D. Exhibit D

Answer: B

Question 1

An end user reports that they are unable to control their Cisco IP phone using Cisco Unified Personal Communicator and cannot make any calls. Which situation can cause this issue?

A. The Cisco Unified Personal Communicator is not registered in the Cisco Unified Presence server.
B. The LDAP integration is incorrect.
C. The Cisco Unified Personal Communicator is configured in desk-phone mode.
D. The Allow Control of Device from CTI checkbox in the device configuration on Cisco Unified Communications Manager is disabled.
E. The Allow Control of Device from CTI checkbox in the device configuration on the Cisco Unified Presence is disabled.

Answer: D

Question 2

Refer to the exhibit. The exhibit shows a partial screen shot for a Cisco Unified Personal Communicator device. If the username that is associated with this device is jdoe, what should the device name be?

A. The device name should be JDOEUPC.
B. The device name should be UPCJDOE.
C. The device name should be JDOE.
D. The device name should be UPCCUPC.
E. The device name should be UPCCSF.
F. The device name has no naming convention.

Answer: B

Question 1

Which two options are features in Cisco Unified Presence? (Choose two)

A. IP Phone Messenger
B. Native Presence
C. BLF speed-dial
D. Enterprise Instant Messaging
E. BLF speed-dial pickup

Answer: A B

Question 2

Which two protocols are used by Cisco Unified Presence? (Choose two)

A. SIP/SIMPLE
B. XMPP
C. SCCP
D. PPPoX
E. IMPP

Answer: A B

Question 3

Refer to the exhibit. What protocol is being used to send and receive instant messaging between the Cisco Unified Personal Communicator and Cisco Unified Presence?

A. Enterprise Instant Messaging Protocol
B. Extensible Messaging and Presence Protocol
C. SIP
D. SCCP
E. CTI

Answer: B

Question 4

In which location is Cisco Unified Presence enabled for a specific user in Cisco Unified Communications Manager?

A. User Administration
B. Application
C. Advanced Features
D. Capabilities Assignment
E. on the IP phone

Answer: D

Question 5

Which action must be taken for Cisco Unified Personal Communicator clients to access Cisco Unity Connection voice mail?

A. Cisco Unity Connection must be integrated with LDAP.
B. Cisco Unity Connection must be integrated with Cisco Unified Communications Manager using SIP integration.
C. A Microsoft Exchange mailbox store must be configured in Cisco Unified Presence.
D. IMAP must be enabled on Cisco Unity Connection for users that need to access voice mail through Cisco Unified Personal Communicator clients.
E. Voice mail is automatically enabled for users who log in through Cisco Unified Personal Communicator clients.

Answer: C

Question 6

What application uses the Cisco Unified Operating System for administration and configuration?

A. Cisco Unity Express
B. Cisco Unified Messaging Gateway
C. Cisco Unified Communications Manager Express
D. Cisco Unified Presence

Answer: D

Question 1

Refer to the exhibit. The exhibit shows a partial screen shot for a Cisco Unified Client Services Framework device. When should this device be configured?

A. when configuring the Service Advertisement Framework feature for Call Control Discovery
B. when Cisco Unified Personal Communicator is used in desk-phone mode.
C. when Cisco Unified Personal Communicator version 7.0 is used in soft-phone mode
D. when Cisco Unified Personal Communicator version 8.0 is used in soft-phone mode
E. when Cisco Unified Personal Communicator version 8.0 is used in desk-phone mode

Answer: A

Question 2

When implementing a plan of action, what should you do?
A. Make all of the changes at once to minimize the impact to users.
B. Limit the impact of the changes to users.
C. Completely remove access lists to ensure that they will not impact the changes.
D. Even if a change adversely affects the users, keep moving forward with the plan of action.

Answer: B

Question 3

Refer to the exhibit. The error message was displayed when the administrator attempted to set the default user password in the user template to four digits long. Which option can rectify this issue?

A. The minimum password length in Cisco Unity Connection should be at least five digits long.
B. The password length needs to be configured under the Authentication Rules settings using the Minimum Credential Length configuration field.
C. The password length needs to be configured under the user template settings using the Minimum Credential Length configuration field.
D. The password length cannot be adjusted under the user template; the password length can only be adjusted under the individual users.

Answer: B

Question 4

Refer to the exhibit. The exhibit shows a partial screen shot for a Cisco Unified Client Services Framework device. If the username that is associated with this device is jdoe, what should the device name be?

A. The device name should be JDOEUPC.
B. The device name should be UPCJDOE.
C. The device name should be JDOE.
D. The device name should be UPCCUPC

Answer: C D

Question 5

Which two types of device can the Cisco Unified Disaster Recovery use as a backup target? (Choose two.)

A. DVD RAM drive
B. FTP server
C. tape device
D. TFTP server
E. WebDAV server
F. SFTP server
G. CD RAM

Answer: C F

Question 6

What is a benefit of using FRF.12 in a Frame Relay network?

A. provides a Layer 3 mechanism for reducing latency in the network
B. fragments packets into equal sizes to reduce fixed-network delay
C. reduces delay and jitter by expediting the transfer of smaller frames through the hardware transmit queue
D. eliminates the need for prioritization of delay-sensitive traffic

Answer: B

Question 7

In which scenario is it possible to have the same directory number configured on two different lines or phones and not be a shared line?

A. directory number assigned to different partitions
B. directory number assigned to the same partition
C. directory number assigned to different calling search spaces
D. directory number assigned to the same calling search space

Answer: A

Question 8

card type t1 0 0
enbale password cisco
!
isdn switch-type primary-ni
!
controller T1 0/0/0
cablelength short 110
pri-group timeslots 1-24 service mgcp
!
interface Serial0/0/0:23
no ip address
encapsulation hdlc
isdn swtich-type primary-ni
isdn incoming-voice voice
no cdp enable
!

Refer to the exhibit. Is this configuration correct and is the T1 operational? If not, what is the cause?

A. Yes, the configuration is correct and it is operational.
B. No, the configuration is incorrect and the T1 is not operational because MGCP is not the proper service statement.
C. No, the configuration is incorrect and the T1 is not operational because, the isdn switch-type is incorrect.
D. No, the configuration is incorrect and the T1 is not operational because the isdn-bind-13ccmmanager command is missing from the serial0/0/0:23 interface.
E. No, the configuration is incorrect and the T1 is not operational because the noip address command is applied to the serial0/0/0:23 interface.

Answer: D

Question 9

What would be the result if a user logs into Cisco Extension Mobility using a device profile that is not subscribed to the Cisco Extension Mobility Service?

A. Normal Cisco Extension Mobility operation will occur.
B. The user will not be able to log into Cisco Extension Mobility on the phone.
C. The phone will not use the correct device profile.
D. The user will not be able to log out of Cisco Extension Mobility on the phone.
E. The phone will reboot continuously.

Answer: D

Exam Description

The Introducing Cisco Voice and Unified Communications Administration (ICOMM v8.0) 640-461 exam is associated with the CCNA Voice certification. This exam tests a candidate’s knowledge of the architecture, components, functionalities, and features of Cisco Unified Communications solutions. It also tests the knowledge needed to perform tasks such as system monitoring, moves, additions and changes on Cisco Unified Communications Manager, Cisco Unified Communications Manager Express, Cisco Unity Connection, and Cisco Unified Presence.

ICOMM Exam Duration: 75 minutes
Number of Questions: 60-70

Notice: The last day to test the old IIUC 640-460 exam or CVoice 6.0 exam is Feb-28-2011.

Please share with us your experience after taking the ICOMM exam, your materials, the way you learned, your recommendations…

Please share with us your experience after taking the TUC exam, your materials, the way you learned, your recommendations…

Please share with us your experience after taking the CIPT2 exam, your materials, the way you learned, your recommendations…

Please share with us your experience after taking the CIPT1 exam, your materials, the way you learned, your recommendations…

Please share with us your experience after taking the QoS exam, your materials, the way you learned, your recommendations…

First you should learn about the difference between them. The CCNA Voice 640-460 exam, sometimes called the Implementing Cisco IOS Unified Communications (IIUC) 640-460 exam, is the exam for those who are working with the Cisco CallManager Express (CME), Cisco Unity Express (CUE), Unified Communications 500 (UC500) Smart Business Communications System solutions and implementing EPhone, IP Phone. These devices are often used in medium & small organizations (with less than 2000 employees).

In the other hand, the CVoice 6.0 642-436 exam is dedicated for ones who use the Cisco Unified Communications Manager (CUCM) and Cisco Gateways/Gatekeepers, including dial-peer & call-leg concept, H.323 & SIP protocols. It is typically employed by large organizations such as governments, large companies. In general, this path is easier to learn than the previous path and it is similar to CCNA and CCNP with command-line-interface (CLI). Moreover, you can use GNS3 to simulate a Voice Gateway or Gatekeeper so you will have more fun (in the case you don’t have real devices).

(Note: A Voice Gateway or Gatekeeper is just like a normal router with a special IOS for voice)

If you are still confused, please have a look at the menu on the right of this website, it gives you information about what are covered in these two exams.

Some Cisco IP Phones (for CCNA Voice 640-460)

Cisco Unified Communications 500 Series (for CCNA Voice 640-460)

Now we will go into the main part of this article: Which path should you choose? Well, it really depends on your job. If you often see devices shown above then you should go with CCNA Voice 640-460 to gain knowledge about Cisco Voice devices (CME, CUE, UC500, Ephone…). Otherwise, go with CVoice 642-436. Notice that a company with more than 2000 employees or customers seldom uses above devices because they are not suitable for a big company and not cheap (this is a secret thing Cisco never tells you while learning their certification ^_^), so if you work (or want to work) with a big company, go with CVoice 642-436 definitely.

One big thing needs to be considered is the certification you will get after complete the exam. You can get the CCNA Voice Certificate by either passing the CCNA Voice IIUC 640-460 exam or the CVoice 6.0 642-436 exam. CCNA Voice is the fundamental certification in the Cisco Voice System, just like CCNA (but you will need a valid CCNA Certification to get it). But the problem occurs if you want to continue pursuing CCVP, a higher voice certification of Cisco (just like CCNP). CCVP requires passing 5 exams: the CVoice 642-436, QoS 642-642, CIPT1 642-446, CIPT2 642-456, TUC 642-426 exams. Yes, as you can see, you need the CVoice 6.0 642-436 to be a CCVP. It means if you learned 640-460 and want to get CCVP, you need to learn CVoice 642-436 as a prerequisite.

Some materials for learning CCNA Voice 640-460:

• CCNA Voice Official Exam Certification Guide by Jeremy Cioara, Michael J. Cavanaugh, Kris A. Krake
• Video Training: CBT Nuggets – Cisco CCNA Voice – Exam-Pack 640-460: IIUC by Jeremy Cioara

Some materials for learning CVoice 642-436:

• CVOICE Student Guide v6.0 (this material and the Cisco Press Authorized Self Study Guide CVOICE book are very identical, up to 95% so you should use CVOICE Student Guide v6.0 if available)
• Video Training: Cisco CCVP – Exam-Pack: 642-436 CVOICE by Jeremy Cioara

If you have any questions or suggestions, please don’t hesitate to comment. It surely will help all of us very much!

Note about the new Voice exams:

Cisco is officially changing the name of its Cisco Certified Voice Professional (CCVP) certification to Cisco Certified Network Professional (CCNP) Voice. The name change is intended to provide a clear career path for Cisco voice and unified communications IT professionals aligning to the entire Cisco Voice certification track.  Effective Tuesday, October 19, 2010, candidates studying for the previous CCVP exams or new CCNP Voice exams will receive a CCNP Voice certificate. The last day to test using current CCVP exams is Monday, February 28, 2011. For more information please read https://learningnetwork.cisco.com/docs/DOC-9016.

Question

This task requires you to access the Cisco Unified Communications 500 Series GUI to configure it per the given requirements as stated in the scenario.

From the choices on the left you may select Scenario, Cisco Unified Communications Server (CUC) Device, or Topology. The Scenario link provides the details of the task to be accomplished. The Topology link displays the network. To get access to UC500, click on the PC shown in the Topology or you can click on the CUC Device button on the left.

Scenario:

You have been asked to implement an SBCA solution for a startup company that has two technical support staff. The lP Phones use 4 digit extensions starting with 5001. Using the information in tables 1 and 2, configure the following:

1. Configure the AA & Voicemail features

2. Add the two IP Phones with extensions/users given in table 2 including Voicemail.

3. 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.

Table 1

Table 2

Answer and Explanation

Click on the PC to connect to the Cisco Unified Communications 500 Series

Pressing OK to enter the main screen of CCA. Notice that you will need to use username:Cisco / password:Cisco to login. In the Configure Panel (on the left) click on Telephony and select Voice to see all the tabs we need to configure.

It is the place where we have to solve all the requirement of this lab. So let’s start!

Notice: Don’t press OK or Apply until all the tabs are configured!

Requirement 1: Configure the AA & Voicemail features

UC520, by default, only uses 3-digit extensions but all the required extensions in this lab are 4-digit so we have to adjust it by clicking on the Dial Plan tab. In the “Number of Digits Per Extension” type 4, just leave other configurations unchanged because we will return to this tab later.

The first requirement of this lab-sim is to configure the AA & Voicemail features so click on this tab to see its content

Assign extensions to this tab as described in table 1 (Auto Attendant: 5000; Voicemail: 5111)

Requirement 2: Add the two IP Phones with extensions/users given in table 2 including Voicemail.

To add two IP Phones, select Users tab and assign corresponding values to these boxes

Requirement 3: 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.

+ Configure Hunt Group

The last parameter in table 1 is the Hunt Group Pilot number but we haven’t configured it yet. To configure this parameter, click on the Voice Features tab and you will see a section for configuring Hunt Group, use these settings (Enable Hunt Groups: 1; Pilot#1: 5010; Hunt Type: sequential; Forward to: Voicemail)

Next we need to assign Phone 5001 & 5002 to this group. Return to Users tab and click on the “More” text on Phone 1

In the “Primary Extension” we can assign Phone 1 to Hunt Group 1 by setting Hunt Group of Phone 1 to 5010:

Also set the “Call Forward Busy” and “Call Forward No Answer” to 5002 (Phone 2)

Do the same thing with Phone 2, but in the “Call Forward Busy” and “Call Forward No Answer” set the extensions to 5111 (Voicemail)

(Notice: In theory, we don’t need to configure “Call Forward Busy” and “Call Forward No Answer” but in the exam we should configure them to make sure we get full mark.)

Most things are done, but when a call coming how can we forward it to Hunt Group 1? This is the last step we need to do.

Select Dial Plan Tab again, in the Incoming Call handling – FXO Trunks set the text to Hunt Group. This will make another box, “Available Hunt Groups”, appears; set it to Hunt Group:1 (5010)

For your information:

Below is some information about Hunt Group:

Hunt groups create a shared line appearance on specified phones that hunts for members of the hunt group based on configured parameters. Phones in a hunt group can ring based on a configured sequence, based on longest idle time or in a peer fashion. If no member of the hunt group is available to take the call, behaviors can be configured for forwarding the call or sending the caller to voicemail.

—————————–

Note: As system configuration is done color highlighting is used both on the tab label and throughout the tab pages. To indicate if steps were completed correctly (green) highlighting is used. If additional configuration is required or if the configuration is incorrect (red) highlighting is used. The configuration cannot be saved until all items that are critical to the system are configured. Once all configuration is correct it can be saved to memory.

Question

To configure the Cisco Unified Communications Manager Express click on the console host icon that is connected to a Cisco Unified Communications Manager Express by a serial console cable (shown in the diagram as a dashed black line). You can click on the buttons below to view the different windows. Each of the windows can be minimized by clicking on the [-]. You can also reposition a window by dragging it by the title bar.

The “Tab” key and most commands that use the “Control” or “Escape” keys are not supported and not necessary to complete this simulation. The help command does not display all commands of the help system.

Scenario:
You are required to manually configure a Cisco Unified Unified Communications Manager Express to support two IP phones with directory numbers starting with 5001.
The phones need to be configured as dual-line phones as per the following table:

The Cisco Unified Communications Manager Express should be configured as a DHCP and TFTP server.
The IP phones should obtain their IP addresses via DHCP in the range 10.3.130.10 – 10.3.130.254. 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.

Topology:

Answer and Explanation

Configure DHCP and TFTP Server:

CME#configure terminal
CME(config)#ip dhcp excluded-address 10.3.130.1 10.3.130.9
CME(config)#ip dhcp pool MyPool
CME(dhcp-config)#network 10.3.130.0 255.255.255.0
CME(dhcp-config)#default-router 10.3.130.1
CME(dhcp-config)#option 150 ip 10.3.130.1
CME(dhcp-config)#exit

The default-router sets the default gateway that will handed out to the DCHP clients and the option 150 tells the IP Phone where the TFTP server is. In this case, the CME takes the role of DHCP server and TFTP server so we use the same IP address 10.3.130.1.

CME(config)#telephony-service
CME(config-telephony)#max-dn 3
CME(config-telephony)#max-ephone 2
CME(config-telephony)#ip source-address 10.3.130.1
CME(config-telephony)#exit

As the question required, we have to configure these phones as dual-line ones. 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.

CME(config)#ephone-dn 1 dual-line
CME(config-ephone-dn)#number 5001
CME(config-ephone-dn)#exit

CME(config)#ephone-dn 2 dual-line
CME(config-ephone-dn)#number 5002
CME(config-ephone-dn)#exit

CME(config)#ephone-dn 3 dual-line
CME(config-ephone-dn)#number 5010
CME(config-ephone-dn)#exit

CME(config)#ephone 1
CME(config-ephone)#mac-address 0014.1cbc.e179
CME(config-ephone)#button 1:1 2:3
CME(config-ephone)#restart
CME(config-ephone)#exit

CME(config)#ephone 2
CME(config-ephone)#mac-address 0003.e3c4.463c
CME(config-ephone)#button 1:2 2:3
CME(config-ephone)#restart
CME(config-ephone-dn)#exit
CME(config)#exit

The button 1:2 2:3 syntax assigns ephone-dn 2 (number 5002) to button 1 and assigns ephone-dn 3 (number 5010) to button 2. The colon (:) separator designates that this is a “normal ring” button assignment.

Notice that the ephone-dn 3 is assigned to both ephones with the button … 2:3 commands. With this configuration, incoming calls to DN-3 (5010) will ring on both phones, and the call will be transferred to whoever answers the call first.

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

Finally, save the configuration

CME#copy running-config startup-config

Question 1

Which two codes together make up the number that follows the E.164 recommendation numbering scheme? (Choose two)

A. country code
B. subscriber code
C. national destination code
D. provider code

Answer: A B

Explanation

E.164 is an international numbering plan created by the International Telecommunication Union (ITU). Each number in the E.164 numbering plan

contains the following components:

• Country code (CC)
• National destination code (NDC – optional)
• Subscriber number (SN)

The CC consists of one, two or three digits. It is what we add in order to access different countries and often prefixed with a +

The NDC is the code we often call the area code.

The SN is for telephone numbering. It is given by your phone operator.

E.164 numbers are limited to a maximum length of 15 digits.

For example, the North American Numbering Plan E.164 is as follows:

1-602-555-1212

+ 1: Country code

+ 602555: National destination code (for North American Numbering Plan, 602 is called the area code while 555 is called Central Office Code)

+ 1212: Subscribe Number

Answer C is also correct but just optional. E.164 Numbering Plan must have Country Code and Subscriber Code so A & B are the correct answers.

Question 2

Which statement is true about only out-of-band signaling?

A. A signaling bit is robbed from each frame.

B. Signaling bits are sent in a special order in a dedicated signaling frame.

C. All signaling is directly associated with its corresponding voice frame.

D. All voice packets carry their own signaling.

Answer: B

Explanation

Out-of-Band signaling is telecommunication signaling exchange of information in order to control a telephone call. Out-of-Band signaling uses common channel signaling (CCS), that means signaling information is transmitted using a separate, dedicated signaling channel.

Answers A C D are characteristics of Channel associated signaling (CAS) so they are not correct.

Question 3

The D channel in ISDN is an example of which two signaling methods? (Choose two.)

A. CCS signaling

B. out-of-band signaling

C. in-band signaling

D. CAS signaling

Answer: A B

Explanation

There are two types of ISDN lines: Basic Rate ISDN (BRI) and Primary Rate ISDN (PRI). Both BRI and PRI types have the same 64kbps D channel that is used for call supervision. This D channel is dedicated for signaling only and contains all the necessary signaling for establishing call between two end-points so it is a kind of CCS signaling and out-of-band signaling.

Question 4

In North America, which E&M signaling type is used most often for geographically separated equipment?

A. Type I

B. Type II

C. Type III

D. Type IV

E. Type V

Answer: B

Explanation

This information is quoted from http://www.cisco.com/en/US/tech/tk1077/technologies_tech_note09186a0080093f60.shtml

E&M Type I – This is the most common interface in North America.

• Type I uses two leads for supervisor signaling: E, and M.
• During inactivity, the E-lead is open and the M-lead is connected to the ground.
• The PBX (that acts as trunk circuit side) connects the M-lead to the battery in order to indicate the off-hook condition.
• The Cisco router/gateway (signaling unit) connects the E-lead to the ground in order to indicate the off-hook condition.

E&M Type II – Two signaling nodes can be connected back-to-back.

• Type II uses four leads for supervision signaling: E, M, SB, and SG.
• During inactivity both the E-lead and M-lead are open.
• The PBX (that acts as trunk circuit side) connects the M-lead to the signal battery (SB) lead connected to the battery of the signaling side in order to indicate the off-hook condition.
• The Cisco router / gateway (signaling unit) connects the E-lead to the signal ground (SG) lead connected to the ground of the trunk circuit side in order to indicate the off-hook condition.

E&M Type III – This is not commonly used in modern systems.

• Type III uses four leads for supervision signaling: E, M, SB, and SG.
• During inactivity, the E-lead is open and the M-lead is set to the ground connected to the SG lead of the signaling side.
• The PBX (that acts as trunk circuit side) disconnects the M-lead from the SG lead and connects it to the SB lead of the signaling side in order to indicate the off-hook condition.
• The Cisco router / gateway (signaling unit) connects the E-lead to the ground in order to indicate the off-hook condition.

E&M Type IV – This is not supported by Cisco routers / gateways.

E&M Type V – Type V is symmetrical and allows two signaling nodes to be connected back-to-back. This is the most common interface type used outside of North America.

• Type V uses two leads for supervisor signaling: E, and M.
• During inactivity the E-lead and M-lead are open.
• The PBX ( that acts as trunk circuit side) connects the M-lead to the ground in order to indicate the off-hook condition.
• The Cisco router / gateway (signaling unit) connects the E-lead to the ground in order to indicate off-hook condition.

Although above information specifies E&M Type 1 is the most commonly used interface in North America but this type generates significant delay in the signaling operation when transmitting between geographically separated equipment and affects voice signal quality (because of significant inductance and capacitance of the long wires) so Type 2 is often used instead.

Question 5

Which three are supervisory signals? (Choose three)

A. busy

B. on hook

C. off hook

D. call waiting

E. ring

Answer: B C E

Explanation

Supervisory signals involves the detection of changes to the status of a circuit. In other words, supervisory signaling is used to indicate the state of a circuit. Once these changes are detected, the supervisory circuit generates a predetermined response. There are three different types of supervisory signals, which are:

1. On-hook

2. Off-hook

3. Ring

When a telephone handset is in the cradle, the circuit is said to be on-hook. In on-hook state, the circuit is said to be open, thus preventing the current from flowing through the telephone.

When the telephone handset is removed from the cradle, the circuit transitions to an off-hook state and there is a current flowing through the electrical loop. When the telephone network senses the off-hook state via the current flow, it provides a signal in the form of dial-tone that it is ready to accept the call. When making a call, the caller receives a ringback tone from the telephone switch, which alerts the caller that the telephone switch is sending ringing voltage to the called party. It is important to know that only the ringing that the recipient (the called party) hears is the supervisory signal; the ringback tone that the caller hears is simply a call-progress indicator and is not a supervisory signal.

Question 6

What is the approximate frequency range of human speech?

A. 20 Hz to 20,000 Hz

B. 40 Hz to 15,000 Hz

C. 200 Hz to 9000 Hz

D. 600 Hz to 5400 Hz

Answer: C

Explanation

Human speech ranges from 200 Hz to 9000 Hz but Nyquist cut the sampling frequency range to 4000 Hz to save bandwidth although this cut down the quality of voice too.

Question 7

What is the process of assigning audio amplitude to a unique digital code word?

A. linear prediction

B. encoding

C. sampling

D. quantization

Answer: D

Explanation

Quantization is the process of assigning a value from the voltage range based on the amplitude of each audio sample. Notice that the Voltage values are not evenly spaced. The spaces near the horizontal line are much closer than the ones at the two ends. This helps our ears distinguish common sounds more easily.

Question 8

What is the E.164 standard?

A. private numbering plan

B. national numbering plan

C. dial plan

D. international public telecommunications numbering plan

Answer: D

Explanation

E.164 is an ITU-T recommendation which defines the international public telecommunication numbering plan used in the PSTN and some other data networks.

Question 9

For the following items, which is the most common E&M type used outside North America?

A. Type IV

B. Type I

C. Type II

D. Type III

E. Type V

Answer: E

Explanation

Read question 4

Question 10

A new business in Great Britain needs to have a PSTN connection that will handle a maximum of 30 inbound and outbound calls at any given time. The customer only has one slot available on the designated PSTN router. Which digital line type will you recommend?

A. ISDN T1 PRI

B. QSIG

C. ISDN E1 PRI

D. ISDN BRI

Answer: C

Explanation

The ISDN E1 PRI has 32 timeslots (channels). Each timeslot is 8 bits and has a data rate of 64,000 bits/second. Timeslot 0 is used for frame synchronization and alarms. Timeslot 16 is used for signaling so we can use 30 timeslots to carry calls.

ISDN T1 PRI only has 24 timeslots and can not support 30 simultaneous calls.

QSIG is just an ISDN based signaling protocol for signaling.

The installation of Wireshark is easy so I will not mention here, you can find newest Wireshark version at http://www.wireshark.org/download.html

Using of Wireshark/Ethereal

1. Capturing
Normally it is possible to use Ethernet hub with ethereal or some better switch on which one Ethernet port can be configured as monitoring portTo capture Ethernet traffic start Wireshark/Ethereal, select Capture menu and click to Options. Following screen will appear:

Capture Options

In interface selection select Ethernet interface from which you would like to capture traffic. In some configurations default selection can be for example Generic NdisWan Adapter – which is not physical network card from which Wireshark/Ethereal is able to capture. This adapter can be founded in configurations with enabled terminal services. If capture for some specific host is needed it is possible to define filter. Examples of some filters related to hosts:

It is possible to capture just some low-level protocol. Here a few examples:

2. Filtering (during capture session)

It is possible, during capture session, to define another filter which will apply to captured information. See following example

(Wireshark/Ethereal in action)

In filter field is string: “ldap” which means that Wireshark/Ethereal will show just transactions which are using ldap protocol. It is possible to change value of this filter during capturing session.
Some simple examples:

More complicated examples:

Note: Filtering is case sensitive!

]]> http://voicetut.com/knowledge-base/wireshark-beginner-guide/feed 4 http://voicetut.com/cvoice-642-436/analog-voice-ports http://voicetut.com/cvoice-642-436/analog-voice-ports#comments Mon, 14 Jun 2010 06:28:44 +0000 voicetut http://voicetut.com/?p=58 Here you will find answers to CVoice – Analog Voice Port Questions

Question 1

Refer to the exhibit for IP addresses and telephone numbers. You are working with a customer opening a small sales office in Atlanta. You want the user in Atlanta to be able to dial into the PBX in New York over the IP WAN. The New York PBX uses ground start, a two-wire operation, and DTMF dialing. Choose the correct FXO port configuration commands for New York.

A.
voice-port 1/0/0
signal ground-start
operation 2-wire
dial-type dtmf

B.
voice-port 1/1/1
destination 2015551212
signal ground-start
operation 2-wire
type 1
dial-type dtmf

C.
voice port 1/0/0
session target ipv4:172.16.1.1
destination 2015551212
signal ground-start
operation 2-wire
dial-type dtmf

D.
voice port 1/0/0
session target ipv4:172.16.1.1
source 2015551212
signal wink-start
operation 2-wire
dial-type dtmf

Answer: A

Question 2

Refer to the exhibit. Which configuration option will allow communication between a voice-enabled router and a PBX?

A. voice port 1/0/0
signaling wink-start
operation 4-wire
auto-cut-through
type 1

B. voice port 1/0/0
signaling immediate-start
operation 4-wire
type 5

C. voice port 1/0/0
signaling delay-start
auto-cut-through
operation 4-wire
type 3

D. voice port 1/0/0
signaling wink-start
operation 4-wire
type 4

Answer: A

Question 3

Examine the following PBX system parameters:

• The calling side seizes the line by going off-hook on its E-lead and sends information as DTMF digits.
• The voice path is 4-wires, and the voice enabled router is in another building from the PBX.

Select the correct set of commands to allow communication between a voice enabled router and a PBX.

A.
voice port 1/0/0
signal immediate-start
operation 4-wire
type 2

B.
voice-port 1/0/0
signal delay-dial
operation 4-wire
type 1

C.
voice port 1/0/0
signal wink-start
operation 4-wire
type 3

D.
voice port 1/0/0
signal immediate-start
operation 4-wire
type 4

Answer: A

Question 1

Which option is true concerning the MGCP call agent?

A. acts only as a recorder of call details

B. provides only call signaling and call setup

C. manages all aspects of the call and voice stream

D. monitors the quality of each call after setup

Answer: B

Explanation

MGCP Call Agent is a central control component to remotely control various devices. When the MGCP call agent exists in the network, calls are routed via route patterns on the Call Agent (Cisco Unified Communications Manager), not by dial peers on the gateway.

The messages sent between the voice gateway and the MGCP Call Agent are just used for call signaling and call setup only. In summary, the Call Agent will instruct the gateways what to do in each stage: receive dialed digits, find the destination gateway, send connection request… Finally, the Call Agent will allow gateways to establish RTP Streams with each other. Notice that the voice streams only flow between the two voice gateways, not to the Call Agent.

At the conversation finishs (one of the endpoints goes on-hook), that gateway notifies the Call Agent and the Call Agent sends Delete Connection (DLCX) Requests for both gateways.

Question 2

At what point does the MGCP call agent release the setup of the call path to the residential gateways?

A. after the call agent has been notified that an event occurred at the source residential gateway

B. after the call agent has been notified of an event and has instructed the source residential gateway to create a connection

C. does not release call path setup

D. after the call agent has sent a connection request to both the source and destination and has relayed a modify-connection request to the source so that the source and destination can set up the call path

E. after the call agent has forwarded session description protocol information to the destination from the source and has sent a modify connection to the destination and a create-connection request to the source

Answer: D

Explanation

Below is the call flow between two voice gateway through a MGCP Call Agent

The MGCP call agent releases the setup of the call path to the residential gateways when the conversation begins. After sending the Modify Connection (MDCX), the two gateways have enough information to start the conversation so the duty of the Call Agent finishs.

Question 3

Which three services are supported by CUBE when supporting H323-to-SIP calls? (Choose three)

A. SIP cause codes

B. media flow-around

C. media flow-through

D. codec transparent support

E. Transport Layer Security

F. H.261, H.263, and H.264 video codecs

Answer: C D E

Explanation

Media flow through and media flow around mode is supported on the Cisco Unified Border Element (CUBE). The CUBE is always involved in the call setup (signaling) portion of the call, but the media (RTP bearer stream) may flow through the CUBE or be routed around the platform. Media flow through must be used to support many of the features available like IP address translation and IP address hiding. Media flow around allows the CUBE greater scalability in the number of calls that can be processed by one CUBE router.

For “Media flow through” option, the media packets are passed through the CUBE, they will get terminated and re-originates with CUBE’s IP address and port number, so here we cannot find the original gateway’s ip address. This is one of the security feature in the CUBE. The default option is “media flow-through”.

Use the “codec transparent” command to configure codec pass-through. Use this command to enable endpoint-to-endpoint codec negotiation

without a Cisco UBE router -> D is correct.

Transport Layer Security (TLS) is a security protocol that enables encrypted network communications. TLS on CUBE can be configured on a per-leg basis in order to allow a TLS to non-TLS SIP call. For example, CUBE uses IPSec in order to secure signaling and support calls from H.323 to SIP with the H.323 leg, while the SIP leg uses TLS -> E is correct

The questions only let us choose 3 answers but I think B – media flow-around can be used for H323-to-SIP calls. We just can’t use media flow-around for SIP-to-SIP calls.

For your information, H.323-to-SIP interworking is configured using the allow-connection h323 to SIP command. Then issue the allow-connections sip to h323 command to enable SIP to H.323 calls.

Question 4

Which two are attributes of SCCP? (Choose two)

A. It is Cisco proprietary.

B. It is a supervisory signaling protocol.

C. It is classified as client/server architecture.

D. SCCP devices are considered intelligent endpoints.

Answer: A C

Explanation

SCCP is the only Cisco-proprietary VoIP protocol currently in use. The purpose of SCCP protocol is to provide a signaling protocol between the Cisco Unified Communications Manager and Cisco IP phones. Similar to MGCP, SCCP is a client/server protocol -> A & C are correct.

Supervisory signals involves the detection of changes to the status of a circuit (on-hook, off-hook, ringing). Any event causes a message to be sent to a Cisco UCM -> We can say SCCP is more than a supervisory signaling protocol because it tells the phone exactly what to do. From the on-hook, off-hook, buttons pressed, lamp on/off, through the prompt, key settings, and even the dialtone -> B is not correct.

The beauty of SCCP is that it makes the endpoints very cheap comparing to the H.323 devices. The end stations (telephones) that use SCCP are

called Skinny clients, which consume less processing overhead and they do not contain call control intelligence -> D is not correct.

SCCP devices (in this case the Cisco IP Phones) become “dump” devices when using this protocol because they have to ask the Unified Communications Manager for every action they need to do -> D is not correct.

Question 5

Refer to the exhibit. All IP phones are SCCP phones. Phone D makes an internal call to phone G. Which call setup signaling statement is true?

A. Phone D signals phone G directly. Call setup is handled by the phones.

B. Phone D signals gateway A, which processes the call and signals phone G.

C. Phone D signals gateway B, which processes the call and signals phone G.

D. Phone D signals gatekeeper. The gatekeeper processes the call and signals phone G.

E. Phone D signals the call agent. The call agent processes the call and signals phone G.

Answer: E

Explanation

This is a …weird and wrong question. Maybe the phone they want to ask here is Phone A, B or C because only these phones can use SCCP to communicate with the Call Agent. Phones D and E can’t use SCCP to talk with a H.323 Gateway.

Phone A, B or C are SCCP Phones so they hand over the call control intelligence to the Call Agent and the Call Agent need to process the call and signals phone G before these phones can talk with each other.

Question 6

Which statement is true about MGCP?

A. Call completion is always shared, with some intelligence on the endpoint, some on the call agent.

B. Endpoints always take all actions to complete calls.

C. Endpoints may act alone or cooperate with call agent to complete calls.

D. Call agents order and direct each step of call completion for the endpoints.

Answer: D

Question 1

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

A.

service voice voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to sip

allow-connections sip to h323

B.

service voice voip

allow-connections h323 to h323

allow-connections h323 to sip

C.

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

D.

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

Answer: C

Explanation

The Acme Corp connects to the ITSP via SIP Trunk and connects to RR industries via H.323. The Acme Corp itself uses H.323 so we have to enable protocol interworking with allow-connections commands:

allow-connections h323 to h323: allow Acme Corp to communicate with RR industries (in both ways)

allow-connections h323 to sip: allow Acme Corp to talk with ITSP (Acme Corp can talk and ITSP can hear but not vice versa)

allow-connections sip to h323: allow ITSP to talk with Acme Corp (Acme Corp can hear and ITSP can talk but not vice versa)

Notice that the configuration for H.323 and SIP interworking is unidirectional, thus if bidirectional interworking is required, you need to configure the mirror-matching statement as well.

Acme Corp doesn’t use SIP so we don’t need to configure “allow-connections sip to sip”.

Question 2

H.323 is an umbrella Recommendation from the ITU Telecommunication Standardization Sector (ITU-T) that defines the protocols to provide audio-visual communication sessions on any packet network. Which CUBE configuration will support H.323 protocol interworking and address hiding?

A.

voice services voip

h323 interworking

media flow-around

B.

voice services h323 to h323

h323 interworking

media flow-through

C.

voice services voip

allow-connections h323 to h323

media flow-around

D.

voice service voip

allow-connections h323 to h323

Answer: D

Explanation

Address hiding is a security feature of the CUBE which will hide the IP address of the originating gateway. This feature is turn on by default so we don’t need to set it.

A and B are not correct because the command “h323 interworking” doesn’t exist (moreover A uses “media flow-around” feature which will turn off the address hiding feature).

C is not correct because it uses “media flow-around” feature too.

Question 3

Refer to the exhibit. The Acme Corp. is deploying a CUBE. As a component of protocol interworking between RR Industries and the ITSP, they need to configure at least two dial peers. When the IP WAN is functional, Acme Corp. wants to use 5-digit dialing to RR Industries. Which three dial peers will complete the configuration for Acme Corp.? (Choose three)

A. dial-peer voice 50 voip
destination-pattern 50…
session protocol sipv2
session-target ipv4:192.168.100.100

B. dial-peer voice 1000 voip
destination-pattern 51…
session-target ipv4:192.168.100.100

C. dial-peer voice 91 voip
session protocol sipv2
destination-pattern 91T
session-target ipv4:10.1.100.1
dtmf-relay rtp-nte digit-drop h245-alphanumeric

D. dial-peer voice 91 voip
destination-pattern 91T
session-target ipv4:10.1.100.1
dtmf-relay rtp-nte digit-drop h245-alphanumeric

E. dial-peer voice 1000 voip
destination-pattern 51…
session-target ipv4:10.1.100.1

F. dial-peer voice 50 voip
destination-pattern 50…
session-target ipv4:172.16.14.6

Answer: B C F

Question 1

When setting up a VoIP call, what is the first thing a gateway router tries to match to a dialed number?

A. call leg
B. IP route
C. session target
D. destination pattern

Answer: D

Explanation

First, the gateway attempts to match the called number with the incoming called-number. If no match is found, the router or gateway attempts to match the calling number of the call set-up request with the answer-address of each dial-peers. If no match is found, it attempts to match the calling number of the call set-up request to the destination-pattern of each dial-peer.

Notice that these steps are just applied for inbound dial peer.

Question 2

Refer to the exhibit. Highland Park Property Development is integrating a Cisco Unified Communications Manager Express system with the existing PBX via an E1 QSIG trunk. After the initial configuration, no calls can be placed from IP phones to PBX phones. How can this problem be resolved?

A. Increase the ISDN T302 timer to allow more time for call setup.
B. Add the command isdn negotiate-bchan to the serial interface.
C. Add the command isdn contiguous-bchan to the serial interface.
D. Change the channel selection order from descending to ascending.

Answer: B

Question 3

Refer to the exhibit. The Carmichael caller dials the site access code for Merrimack (6) followed by the four digit extension number of the destination phone (0124). If the call is going to go across the IP WAN, which action will have to be taken?

A. Translate 60124 to 5125550124.
B. Strip the site access code and send four digits.
C. Strip the site access code and prepend 1512555.
D. Do nothing because the site access code matches the last five digits of the target number.
E. Strip the site access code, send four digits, then prepend the access code when it reaches the Merrimack gateway.

Answer: B

Explanation

The site access code (6) is just used to inform the originating gateway which gateway it needs to send traffic to. Therefore, after learning the traffic should be sent to Merrimack gateway, it trips off the site access code. Notice that the receiving gateway will receive “0124″, which is enough information to ring the phone plugged into it.

Question 4

Which path selection mechanism lets you choose either the even or odd channels first?

A. hunt groups
B. trunk groups
C. tailend hopoff
D. Call Admission Control

Answer: B

Explanation

By using trunk groups, we can choose to use either the even or odd channels first with the command:

hunt-scheme …. [even | odd ....] (notice: the full command is very long so I shorten it to the simplest form)

Question 5

When using CUBE, which two statements describe how media flow-through differs from media flow-around? (Choose two.)

A. Media flow-around provides address hiding by terminating both signaling and RTP streams.
B. Media flow-through terminates the signaling channel and the RTP streams flow directly between endpoints.
C. Media flow-around and media flow-through function in a similar manner, but media flow-around supports NAT traversal.
D. Media flow-through terminates the RTP streams but allows signaling to flow directly between endpoints.
E. Media flow-around terminates the signaling stream and allows RTP streams to flow directly between endpoints.
F. Media flow-through provides address hiding by terminating both signaling and RTP streams.

Answer: E F

Explanation

Media flow through and media flow around mode is supported on the Cisco Unified Border Element (CUBE). The CUBE is always involved in the call setup (signaling) portion of the call, but the media (RTP bearer stream) may flow through the CUBE or be routed around the platform.

Media flow around allows the CUBE greater scalability in the number of calls that can be processed by one CUBE router.

For Media flow through option, the media packets are passed through the CUBE, they will get terminated and re-originates with CUBE’s IP address and port number, so here we cannot find the original gateway’s ip address. This is one of the security feature in the CUBE. The default option is “media flow-through”.

Question 6

Refer to the IOS configuration in the exhibit. How will the next incoming call be routed?

A. The call will be routed to the longest idle channel.
B. The call will be routed to the least used channel.
C. The call will be routed to a random available channel.
D. The call will be routed to the next available channel, starting from channel 1, hunting up toward channel
E. The call will be routed to the next available channel, starting from channel 24, hunting down toward channel 1.

Answer: E

Explanation

In the configuration, we learn that the hunt-scheme sequential is used. It specifies the sequential search method for finding an available channel in a trunk group for outgoing calls. The syntax of this command is shown below:

hunt-scheme sequential [both | even | odd [up | down] ]

Description:

+ both: Searches both even and odd numbered channels.
+ even: Searches for an idle even numbered channel. If no idle even numbered channel is available, an odd-numbered channel is sought.
+ odd: Searches for an idle odd numbered channel. If no idle odd numbered channel is available, an even-numbered channel is sought.

+ up: Searches channels in ascending order based within a trunk group member.
+ down: Searches channels in descending order within a trunk group member.

Notice that up & down parameters are used with both, even or odd.

Therefore the command hunt-scheme sequential even up searches in ascending order for an even numbered idle channel starting with the trunk group member of highest precedence. I am not so sure but channel 24 will have highest precedence so the “hunt” begins from channel 24 down to channel 1. Therefore, E is the most suitable solution for this question.

The cas-custom command is used to customize T1/CAS signaling parameters for a particular T1 channel group on a channelized T1 line.

Question 7

Site A uses three-digit internal numbers and remote Site B uses four-digit internal numbers. All calls to the PSTN are routed through Site B. What dial plan below best represents provision simplicity, assuming the NANP numbering plan?

A. Translate all called numbers within Site A to four digits.
B. Translate all called numbers within Site B to three digits.
C. Translate all called numbers leaving Site A to ten digits.
D. Translate all called numbers at either site to ten digits.

Answer: C

Explanation

North American Numbering Plan (NANP) is designed around a 10-digit numbering plan:

(Sometimes you will see it as NXX – NXXX – XXXX, which means that the first and fourth digits can’t be zero or one)

It consists of 3-digit area codes and 7-digit telephone. For telephone numbers that are located within an area code, the PSTN uses a seven-digit dial plan numbers.

Notice that “Site B uses four-digit internal numbers” means we need ten digits to access site B from an outside PSTN. Therefore, if people from Site A want to call people at site B and sometimes they just press 4 digits then the administrators should translate the called numbers to ten digits before leaving Site A.

Question 1

Refer to the exhibit. Choose the correct configuration command set that will allow the gateway in zone BR to register with the gatekeeper in the same zone.

A.
interface fastethernet 0/0
ip address 10.1.110.1
h323-gateway voip interface
h323-gateway voip id BR ipaddr 10.2.120.1
h323-gateway voip h323-id BRgw
!
gateway
B.
interface fastethernet 0/0
ip address 10.2.120.1
h323-gateway voip interface
h323-gateway voip id BR ipaddr 10.1.120.2
h323-gateway voip h323-id BRgw
!
gateway
C.
interface fastethernet 0/0
ip address 10.2.120.1
h323-gateway voip interface
h323-gateway voip id BR ipaddr 10.2.120.1
h323-gateway voip h323-id BRgw
!
gateway
D.
interface fastethernet 0/0
ip address 10.1.110.1
h323-gateway voip interface
h323-gateway voip id BR ipaddr 10.1.110.1
h323-gateway voip h323-id BRgw
!
gateway
E.
interface fastethernet 0/0
ip address 10.2.120.1
h323-gateway voip interface
h323-gateway voip id HQ ipaddr 10.1.110.1
h323-gateway voip h323-id BRgw
!
gateway

Answer: C

Explanation

Notice that the router at zone Br is functioned as both gateway and gatekeeper and it uses the IP address of 10.2.120.1 as the “zone local BR”. Therefore if we want “the gateway in zone BR to register with the gatekeeper in the same zone” we must use 10.2.120.1 in the command:

h323-gateway voip id BR ipaddr 10.2.120.1

In which, BR is the zone name defined in the “zone local BR” command (of the gatekeeper) and the is the IP address of an interface of the gatekeeper and it should be 10.2.120.1-> B, D and E are not correct.

A can be correct but it is not as clear as answer C.

Notice that last command h323-gateway voip h323-id BRgw specifies the BRgw is the name of the gateway to communicate with the gatekeeper.

Question 2

Examine the example output.

Choose the command that will restore communication with gatekeeper functionality to this device.

A. h323-gateway voip h323-id GK1
B. gateway
C. h323-gateway voip bind srcaddr 172.16.2.2
D. h323-gateway voip GW1-zone2.abc.com abc.com ipaddr 172.16.2.1

Answer: B

Explanation

The gateway command enables the H.323 VoIP gateway to register with the gatekeeper. This is the first command you should enter when configuring a voice gateway.

Question 3

Which item correctly describes the relationships between the feature and the category it belongs?

A. Gateway – 1 and 2
CUBE – 3 and 4
B. Gateway – 1 and 3
CUBE – 2 and 4
C. Gateway – 2 and 3
CUBE – 1 and 4
D. Gateway – 2 and 4
CUBE – 1 and 3

Answer: A

Question 1

A customer needs to configure a CAS E&M circuit that will support inbound and outbound DNIS and inbound ANI. Which configuration will accomplish this task?

A. pri-group timeslots 1-24
B. ds0-group 0 timeslots 1-24 type none
C. ds0-group 0 timeslots 1-24 type e&m-fgd
D. ds0-group 0 timeslots 1-24 type fgd-eana
E. ds0-group 0 timeslots 1-31 type r2-digital r2-compelled ani

Answer: C

Explanation

To define T1 channels for compressed voice calls and the channel-associated signaling (CAS) method by which the router connects to the PBX or PSTN, enter the ds0-group controller configuration command. Below is the syntax of this command:

ds0-group ds0-group-no timeslots timeslot-list type {e&m-immediate | e&m-delay | e&m-wink | e&m-fgd | fgd-eana}

Description

(Reference: http://www.ciscosecure.net/en/US/docs/ios/12_1/12_1xd/feature/guide/hdv_fgd.html)

T1 CAS always provides the ANI/DNIS delimiter on incoming T1/CAS trunk lines. The customer wants E&M circuit so the answer should be C.

Notice:

+ CAS signaling main feature is its use of user bandwidth to perform signaling functions. CAS signaling is often referred to as robbed-bit-signaling because user bandwidth is being “robbed” by the network for other purposes.

+ E&M signaling is typically used for trunks. It is normally the only way that a central office (CO) switch can provide two-way dialing with direct inward dialing.

+ ANI – Automatic number identification. SS7 (signaling system 7) feature in which a series of digits, either analog or digital, are included in the call, identifying the telephone number of the calling device. In other words, ANI identifies the number of the calling party.

+ DNIS – Dialed number identification service, also known as the called party number. The telephone number of the called party after translation occurs in the Public Switched Telephone Network. A given destination may have a different DNIS number based on how the call is placed (for example, 800 or direct dial).

Question 2

In T1 CAS, where are the signaling states and control features carried for Super Frame robbed-bit signaling?

A. 6th and 12th frame
B. 6th, 12th, 18th, and 24th frame
C. the first and seventeenth time slot
D. the first and sixteenth time slot

Answer: A

Explanation

Each T1 has 24 channels ( or 24 DS0 – digital signal level 0) that can transmit 8 bits per channel each. This give us a total of 192 bits. One more bit is used for framing, bringing the total to 193 bits. Super Frame bundles 12 of these 193-bit frames for transport. The picture below shows the structure of a T1 Super Frame

The T1 CAS signaling looks at every 6th & 12th frames for signaling information, these bits are referred to as the A and B bits. The A and B bits can represent different signaling states or control features (on hook or off hook, idle, busy, ringing, and addressing)

According to Nyquist theory, we sample voice 8000 times per second, that means we need to send 8000 of these 193-bit frames every second. So 8000 x 193 = 1,544 Mbps.

Extended super frame (ESF), due to grouping the frames in sets of twenty-four, has four signaling bits per channel or timeslot. These occur in frames 6, 12, 18, and 24 and are called the A-, B-, C-, and D-bits respectively. So if the question asks about ESF, the answer should be B.

Question 1

Which mechanism do you use to implement calling privileges on Cisco Unified Communications Manager Express?

A. CoS
B. QoS
C. CAC
D. COR
E. SRST

Answer: D

Explanation

Calling privileges define the destination a user is allowed to dial and they are implemented on Cisco IOS gateway using Class of Restriction.

Class of Restriction (COR) is the feature that determines which numbers might not be dialed on the system. COR is required only when you want to restrict the ability of some phones to make certain types of calls but allow other phones to place those calls. COR functionality provides the ability to deny certain call attempts on the basis of the incoming and outgoing CORs that are provisioned on the dial-peers. This functionality provides flexibility in network design, allows users to block calls (for example, calls to 900 numbers), and applies different restrictions to call attempts from different originators.

(Reference: http://www.cisco.com/en/US/tech/tk652/tk90/technologies_configuration_example09186a008019d649.shtml)

Question 2

Using Cisco Unified Communications Manager Express, what four steps are necessary to implement COR? (Choose four)

A. Configure SRST.
B. Define COR labels.
C. Configure COR lists.
D. Assign COR list to ephone-DN.
E. Configure COR lists on voice ports.
F. Configure dial peers and assign COR lists.

Answer: B C D F

Explanation

Four steps to configure COR on Cisco IOS gateway using Cisco Unified Communications Manager Express:

1) Define COR labels.
2) Configure COR lists.
3) Configure dial peers and assign COR lists.
4) Assign COR lists to ephone-dn.

For example, we will define three calling privilege classes:

Local: This class should allow emergency and local calls.
Long Distance: This class should allow emergency, local, and long distance calls.
International: This class should allow emergency, local, long distance, and international calls.

Step 1: Define the four COR labels to be used as COR list members with the command dial-peer cor custom.

Router(config)#dial-peer cor custom
Router(config-dp-cor)#name 911
Router(config-dp-cor)#name local
Router(config-dp-cor)#name ld
Router(config-dp-cor)#name intl

Description

• 911: Allows calls to emergency 911
• local: Allows local calls only
• ld: Allows long distance calls
• intl: Allows international calls

Step 2: Define the COR lists that will be assigned as “outgoing” to the PSTN dial peers with the command dial-peer cor list.

Router(config-dp-corlist)#dial-peer cor list 911call
Router(config-dp-corlist)#member 911
Router(config-dp-corlist)#dial-peer cor list localcall
Router(config-dp-corlist)#member local
Router(config-dp-corlist)#dial-peer cor list ldcall
Router(config-dp-corlist)#member ld
Router(config-dp-corlist)#dial-peer cor list intlcall
Router(config-dp-corlist)#member intl

Define the COR lists that will be assigned as “incoming” from the local dial peers with the command dial-peer cor list .

Router(config)#dial-peer cor list local
Router(config-dp-corlist)#member 911
Router(config-dp-corlist)#member local

Router(config)#dial-peer cor list ld
Router(config-dp-corlist)#member 911
Router(config-dp-corlist)#member local
Router(config-dp-corlist)#member ld

Router(config)#dial-peer cor list intl
Router(config-dp-corlist)#member 911
Router(config-dp-corlist)#member local
Router(config-dp-corlist)#member ld
Router(config-dp-corlist)#member intl

Step 4: Assign Outbound COR Lists to PSTN Dial Peers

• Dial peer 911 has the outgoing 911call COR list
• Dial peer 9911 has the outgoing 911call COR list.
• Dial peer 9 has the outgoing localcall COR list.
• Dial peer 91 has the outgoing ldcall COR list.
• Dial peer 9011 has the outgoing intlcall COR list.

Router(config)#dial-peer voice 911 pots
Router(config-dial-peer)#destination-pattern 911
Router(config-dial-peer)#forward-digits all
Router(config-dial-peer)#corlist outgoing 911call
Router(config-dial-peer)#port 0/0/0:23

Router(config)#dial-peer voice 9911 pots
Router(config-dial-peer)#destination-pattern 9911
Router(config-dial-peer)#forward-digits 3
Router(config-dial-peer)#corlist outgoing 911call
Router(config-dial-peer)#port 0/0/0:23

Router(config)#dial-peer voice 9 pots
Router(config-dial-peer)#destination-pattern 9[2-9]……
Router(config-dial-peer)#corlist outgoing localcall
Router(config-dial-peer)#port 0/0/0:23

Router(config)#dial-peer voice 91 pots
Router(config-dial-peer)#destination-pattern 91[2-9]..[2-9]……
Router(config-dial-peer)#prefix 1
Router(config-dial-peer)#corlist outgoing ldcall
Router(config-dial-peer)#port 0/0/0:23

Router(config)#dial-peer voice 9011 pots
Router(config-dial-peer)#destination-pattern 9011T
Router(config-dial-peer)#prefix 011
Router(config-dial-peer)#corlist outgoing intlcall
Router(config-dial-peer)#port 0/0/0:23

Reference: CVoice Student Guide v6.0 (Page 4-165)

Question 3

Refer to the exhibit. Which dial peer configuration will block phone A from making long distance calls?

Partial configuration on Gateway-A: dial-peer cor custom name Emergency name Local name LD name Intl dial-peer cor list Em01 member Emergency dial-peer cor list Local01 member Local dial-peer cor list LD01 member LD dial-peer cor list Intl01 member Intl dial-peer cor list LocalLst member Emergency member Local dial-peer cor list LDLst member Emergency member Local member LD dial-peer cor list IntlLst member Emergency member Local member LD member Intl

A.
dial-peer voice 1374 pots
destination-pattern 1374
port1/0/0
dial-peer voice 100 voip
corlist incoming Intl01
destination-pattern 9011T
session target ipv4:192.168.101.254

B.
dial-peer voice 1374 pots
destination-pattern 1374
port1/0/0
dial-peer voice 100 voip
corlist outgoing Intl01
destination-pattern 9011T
session target ipv4:192.168.101.254

C.
dial-peer voice 1374 pots
corlist incoming LDLst
destination-pattern 1374
port1/0/0
dial-peer voice 100 voip
destination-pattern 9011T
session target ipv4:192.168.101.254

D.
dial-peer voice 1374 pots
corlist outgoing LDLst
destination-pattern 1374
port1/0/0
dial-peer voice 100 voip
destination-pattern 9011T
session target ipv4:192.168.101.254

E.
dial-peer voice 1374 pots
corlist incoming LocalLst
destination-pattern 1374
port1/0/0
dial-peer voice 100 voip
corlist outgoing Intl01
destination-pattern 9011T
session target ipv4:192.168.101.254

F.
dial-peer voice 1374 pots
corlist outgoing LDLst
destination-pattern 1374
port1/0/0
dial-peer voice 100 voip
corlist outgoing Intl01
destination-pattern 9011T
session target ipv4:192.168.101.254

Answer: E

Explanation

To block phone A from making long distance calls, phone A must belong to an “incoming” dial-peer which is not a member of the LD (Long Distance). In three incoming dial-peer (the three last dial-peers), there is only one dial-peer satisfies with this condition, that is the LocalLst dial-peer so the answer should be E.

One tip to quickly recognizes which dial-peer is for “outgoing” dial-peer is that this type of dial-peer usually have only one member. In this question, the outgoing dial-peers are Em01, Local01, LD01, Intl01. Dial-peers which have more than one member are often “incoming” dial-peers.

You can read http://www.cisco.com/en/US/tech/tk652/tk90/technologies_configuration_example09186a008019d649.shtml for another example. [/protect]

Question 4

Where would you assign COR lists in Cisco Unified Communications Manager Express?

A. ephone
B. ephone-dn
C. voice register dn
D. voice register pool

Answer: B

Explanation

For Cisco Unified Communications Manager Express, the COR list is directly assigned to the appropriate Ethernet phone-dn (ephone-directory number)

Question 1

Refer to the exhibit. You have been asked to configure a dial peer on R2 that will match only the extensions of the four telephones attached. Which dial-peer statement will you use?

A. dial-peer voice 1 pots
destination-pattern 5552.[0-5]0
B. dial-peer voice 1 pots
destination pattern 5552[5-6].0
C. dial-peer voice 1 pots
destination-pattern 555[2-5][56]
D. dial-peer voice 1 pots
destination-pattern 5552[5-6][05]0

Answer: D

Explanation

The numbers can be summaried as 5552(5 or 6)(5 or 0)0 so the destination-pattern should be written as 5552[5-6][05]0 or 5552[56][05]0

Question 2

Refer to the exhibit. When extension 201-555-1000 dials 404-555-1200, how are the digits manipulated in R1 so that they are presented correctly at R2?

A. The outbound VoIP dial peer is matched and all digits are sent.
B. The digits 404-555 are stripped off before matching the outbound POTS dial peer.
C. The digits 404-555 are stripped off by the connection trunk and R2 receives only 1200.
D. R1 collects the 1200 and prepends the tie-line digits 404555. That number is matched to a VoIP dial peer and sent to the appropriate address.

Answer: A

Explanation

When 201-555-1000 (Phone A) calls 404-555-1200 (Phone B) the dial-peer voice 1 voip at R1 is matched with the destination-pattern 404555…. But notice that this is a voip dial-peer so digits are not stripped and all digits are sent to R2.

Question 3

Refer to the exhibit. Your customer wants to converge the existing PBX network with the IP network. The three remote offices have various types of PBXs. The customer is using a combination of tie-lines and trunks to connect the PBXs today. Which kind of connection should be implemented to allow calls to be placed from 201-555-1000 to 727-555-1000 so that when the call is completed, network resources are returned for other uses?

A. PLAR
B. trunk
C. tie-line
D. answer-mode

Answer: C

Explanation

E&M signaling supports tie-line type facilities.

Question 4

Which dial plan characteristic shows the most obvious improvement by dropping a number translation step?

A. availability
B. post-dial delay
C. scalability
D. hierarchical design

Answer: B

Explanation

Post-dial delay is the time between when the last digit is dialed and the moment the phone rings at the receiving location. In the PSTN, people expect a short post dial delay and to hear ring back within seconds. The more translations, digit manipulations, and lookups that take place, the longer the post dial delay becomes. Overall network design, translation rules, and alternate paths affect post dial delay. Minimize the amount of dial peers and translations to reduce post-dial delay.

By dropping a number translation step, the post-dial delay time will be obvious improvement.

Question 5

Refer to the exhibit. Users are not able to complete a call from 678-555-1212 to 770-555-1111. What is the correct diagnosis for the problem?

A. incorrect destination-pattern in router 1
B. incorrect POTS dial-peer statement in router 2
C. incorrect session-target statement in router 2
D. incorrect port statement in router 1 pots dial peer
E. missing no digit-strip on the voip dial peer in router 1

Answer: A

Explanation

The dial-peer 2 voip in Router 1 was configured “destination-pattern 770555..”. Notice that there are only two dots (.) in the destination-pattern that means when the user presses 77055511, the voip dial-peer is matched immediately without waiting for two last “11″ pressed. Therefore router R2 only receives the “777055511″ number and it doesn’t match with the destination-pattern in the “dial-peer voice 1 pots” configured in router R2.

Question 6

Refer to the exhibit. Three department managers share the directory number 3000. The Marketing manager’s phone is attached to port 1/1. The Engineering manager’s phone is attached to port 1/2. The Shipping manager’s phone is attached to port 1/3. In which situation would an incoming call ring on the Shipping manager’s phone?

A. The Marketing manager is on the phone.
B. None of the managers are on the phone.
C. The Engineering manager is on the phone.
D. The Shipping manager and Marketing manager are on the phone.
E. The Engineering manager and Marketing manager are on the phone.

Answer: E

Explanation

With the preference 0 configured in dial-peer voice 1 pots, this dial-peer (Marketing) has the highest priority to receive call if it is idle. Dial-peer 2 (Engineering) has the next priority and dial-peer 3 (Shipping) has lowest priority so it only rings when both Marketing and Engineering phones are busy.

It is a bit weird but the router considers lower preferences to be better than higher preferences. One more notice is that the default preference for a dial peer is 0.

Question 7

Refer to the exhibit. Your customers dial in to your company using a local number, and their calls cross the WAN to an IVR system. They are complaining that the IVR system does not always accept their input or may get it wrong. The IVR system has been checked and is working properly. What needs to be added to the dial peer on the incoming H.323 gateway to correct this problem?

A. no vad
B. tech-prefix 1#
C. codec g729ar8 bytes 30
D. dtmf-relay h245-alphanumeric

Answer: D

Explanation

DTMF is the tone generated when you press a button on a touch-tone phone. This tone is compressed at one end of a call; when the tone is decompressed at the other end, it can become distorted, depending on the codec used. The DTMF relay feature transports DTMF tones generated after call establishment out-of-band by using either a standard H.323 out-of-band method or a proprietary RTP-based mechanism. For session initiation protocol (SIP) calls, the most appropriate method to transport DTMF tones is Real-Time Transport Protocol named telephony event (RTP-NTE) or session initiation protocol notify (SIP Notify).

When you press a button on the touch-tone phone, a “high group” frequency is combined with a “low group” frequency and you can hear a generated tone. Notice that you often don’t see the “A B C D” column in most modern DTMF phones nowadays.

Although DTMF is usually transported accurately when using high-bit-rate voice codecs such as G.711, low-bit-rate codecs such as G.729 and G.723.1 are highly optimized for voice patterns and tend to distort DTMF tones. As a result, interactive voice response (IVR) systems may not correctly recognize the tones. Therefore the IVR sometimes can not recognize the DTMF tones and doesn’t accept their input o may get it wrong.

The main advantage of the “dtmf-relay” command is it sends DTMF tones with greater fidelity than is possible in-band for most low-bandwidth codecs, such as G.729 and G.723.

(Reference: CVoice Student Guide v6.0)

Question 8

You have designed a complex dial plan using digit manipulation. Given the following snippet of your configuration file, what action would you expect to result when a call beginning with the digits “612″ is received?

A. A nine digit number beginning with 5501 will be forwarded.
B. A ten digit number beginning with 5501 will be forwarded.
C. A twelve digit number beginning with 5501612 will be forwarded.
D. A thirteen digit number beginning with 5501612 will be forwarded.

Answer: C

Explanation

This dial-peer has the “no digit-strip” command so no digits are stripped when this dial-peer is matched. So the whole number will be transferred with the format of 5501612xxxxx (5501 is prefixed with the command prefix 5501)

Question 9

Which command sets parameters to search a series of dial peers for a destination that is not in use?

A. dial-peer rotary
B. dial-peer circulate
C. dial-peer hunt
D. dial-peer distribute

Answer: C

Explanation

Dial peer hunting is the process used when an originating router tries to establish a call on different dial peers if the originating router receives a user-busy invalid number or an unassigned-number disconnect cause code from a destination router.

Question 10

On the basis of the provided exhibit. Enzo’s Bikes manufactures high end bicycle frames. Until recently they sold only to bicycle shops; however, now they are starting to sell to end users. They need a way to add two additional sales staff and ensure that the senior sales technician always gets the first call. Drew is the senior sales technician. Bob is the newest sales technician. Bob’s phone should always be the last one chosen for incoming sales calls, after Drew and James. Bob’s phone should be chosen first only when Drew and James are busy on calls. Select the correct dial-peer command set for Bob’s phone.

A.
dial-peer voice 3 pots
destination-pattern 5555110
preference 2

B.
dial-peer voice 3 pots
destination-pattern 5555110
preference firstlast

C.
dial-peer voice 3 pots
destination-pattern 5555110
preference 0

D.
dial-peer voice 3 pots
destination-pattern 5555110
preference high

Answer: A

Explanation

The router considers lower preferences to be better than higher preferences and the default preference is 0. Therefore, by setting the preference of Bob’s dial-peer to 2 we guarantee Bob will be the last one to receive the call (while James’ priority is set to 1 and Drew uses the default configuration).

Question 1

Refer to the exhibit.

Lighthorse Equine Management would like to investigate converging voice and data on their existing T1 Frame Relay WAN link between New York and Atlanta. Currently the following list of applications are consuming no more bandwidth than what is listed on this segment of the network.

T1 link 1536 kbps
e-mail 75 kbps
internet 200 kbps
Oracle 500 kbps
FTP 250 kbps
Total 1025 kbps

The customer has allocated 25% of the WAN link for routing updates and other overhead. They would like to increase the number of samples encapsulated in each PDU to 40 ms. You have calculated 6 bytes of overhead for Frame Relay, no cRTP, and the use of the G.711 codec. How many simultaneous calls could be placed on this link?

A. 0 calls
B. 1 call
C. 2 calls
D. no more than 5 calls
E. no more than 10 calls
F. no more than 20 calls

Answer: B

Question 2

Refer to the exhibit. A QoS strategy has already been deployed on the LAN. Choose three WAN QoS best practices that should be used over the WAN link. (Choose three.)

A. Implement NBAR.
B. Implement admission control.
C. Mark voice traffic as EF in DSCP.
D. Mark voice traffic highest priority in 802.1p.
E. Use cRTP to maximize bandwidth utilization.
F. Configure access switches to trust traffic from IP phones.

Answer: B C E

Question 3

Refer to the exhibit. Users are not able to complete a call from 678-555-1212 to 770-555-1111. What is the correct diagnosis for the problem?

A. incorrect destination-pattern in router 1
B. incorrect POTS dial-peer statement in router 2
C. incorrect session-target statement in router 2
D. incorrect port statement in router 1 pots dial peer
E. missing no digit-strip on the voip dial peer in router 1

Answer: A

Question 4

A telemarketing firm needs to use number translation for incoming and outgoing calls. They have defined two translation profiles, one for incoming and one for outgoing calls. What can be used to simplify this task?

A. dial peer
B. voice port
C. hunt group
D. trunk group
E. source IP group

Answer: D

Question 5

Which command parameter specifies that the router should not attempt to initiate a trunk connection but should wait for an incoming call before establishing the trunk?

A. codec clear-channel
B. connection-trunk 404555…. answer-mode
C. voice-port 1/0:1
D. ds0-group timeslots 1-23 type ext-sig

Answer: B

Question 6

In a VoIP environment when speech samples are framed every 20 ms, a payload of 20 bytes is generated. Assuming a total packet length of 60 bytes, what is the length of the packet header if cRTP is deployed without redundancy checks?

A. 1 byte
B. 2 bytes
C. 3 bytes
D. 4 bytes
E. 20 bytes
F. 40 bytes

Answer: B

Question 7

You have set up a complex dial plan using translation rules. The following translation rule has been configured. What output would correspond to the test translation-rule command?

A. test translation-rule 1 512
The replaced number: 21555021512
B. test translation-rule 1 555
The replaced number: 55521555021
C. test translation-rule 1 617
The replaced number: 61721555021
D. test translation-rule 1 910
The replaced number: 21555021910

Answer: A

Question 8

Which device is used to allow an H.323 stream to transit a firewall?

A. gatekeeper
B. gateway
C. proxy
D. MCU

Answer: C

Question 9

To hide its identity when initiating calls, Phone B requests that Server B place its calls for it. What kind of device is Server B?

A. proxy
B. redirect
C. registrar
D. user agent client
E. user agent server

Answer: A

Question 1

The SJ local zone contains a gatekeeper that controls two gateways, SJ1 and SJ2. Both gateways provide access to area code 408. Which two command strings should be entered into the gatekeeper to give the SJ2 gateway priority over the SJ1 gateway? (Choose two.)

A. zone prefix SJ 408 gw-priority 6 SJ1
B. zone prefix SJ 408 gw-priority 6 SJ2
C. zone prefix SJ 408 gw-priority 10 SJ1
D. zone prefix SJ 408 gw-priority 10 SJ2
E. zone prefix SJ 408 gw-priority 0 SJ2, 10 SJ1
F. zone prefix SJ 408 gw-priority 6 SJ1, 10 SJ2

Answer: A D

Explanation

The simple syntax of “zone prefix” command is

zone prefix gatekeeper-name e164-prefix [gw-priority priority gw-alias,...]

For example, the command “zone prefix SJ 408 gw-priority 6 SJ1“

SJ is the gatekeeper-name, 408 is the E164-prefix area code, 6 is the priority and SJ1 is the GW-alias

The [gw-priority priority gw-alias,...] part defines how the gatekeeper selects gateways in its local zone for calls to numbers beginning with prefix e164-prefix. The priority ranges from 0 to 10, where 0 prevents the gatekeeper from using the gateway gw-alias for that prefix and 10 places the highest priority on gateway gw-alias. The default is 5.

By assigning SJ2 a priority value higher than that of SJ1, SJ2 will be the first choice when making call to this zone.

Question 2

Refer to the H.323 message in the exhibit. What is the gateway doing with the gatekeeper?

A. initial registration
B. full registration
C. lightweight registration
D. registration retry

Answer: C

Explanation

For the first time the gateway registers with the gatekeeper, it uses full registration. Prior to H.323 Version 2, Cisco gateways re-registered with the gatekeeper every 30 seconds. Each registration renewal used the same process as the initial registration, even though the gateway was already registered with the gatekeeper. This behavior generated considerable overhead at the gatekeeper. So from H.323 version 2, gateways can re-register with the gatekeeper using lightweight registration (it still requires the full registration process for initial registration, but uses an abbreviated renewal procedure to update the gatekeeper and minimize overhead).

An endpoint’s registration with a gatekeeper may have a limited life span. The gatekeeper specifies the registration duration for an endpoint by including a timeToLive field in the Registration Confirm (RCF) message. After the specified length of time, the registration is considered expired. The endpoint must periodically send a Registration Request (RRQ) having the keepAlive bit set prior to the expiration time. Such a message may include a minimum amount of information as described in H.225.0 and is known as a lightweight RRQ.

In the exhibit above, we can see the keepAlive bit is set to TRUE -> this is a lightweight RRQ.

Question 3

In which three RAS messages is the technology prefix sent? (Choose three.)

A. GRQ
B. RRQ
C. RCF
D. IRR
E. IRQ

Answer: A B E

Explanation

The Cisco gatekeeper uses technology prefixes to group endpoints of the same type together. It uses the technology prefix appended in the called number to select the destination gateway or zone.

This method prepends a technology prefix to the called number matched by the dial-peer. It is not used for registration, but for call setup with the Cisco gatekeeper. For example, called number 5551234 becomes 1#5551234.

The technology prefix registration information is sent to the Cisco gatekeeper in the RAS Registration Request (RRQ) message. For example:

GWY-B1(config)#interface ethernet 0/0
GWY-B1(config-if)#h323-gateway voip tech-prefix 1#

-> B is correct.

Question 4

Refer to the output from the debug h225 asn1 command in the exhibit. You have configured a gatekeeper with two local zones, hq and br. You want the gateway at the branch location to register with zone BR. What needs to be corrected in the branch gateway to resolve the issue?

A. Change the IP address in the h323-gateway voip id command.
B. Change the gatekeeper-id in the h323-gateway voip id command.
C. Add a zone remote for zone BR so the gateway can register with the correct zone.
D. Change the gatekeeper-id and the IP address in the h323-gateway voip id command.

Answer: B

Question 5

You have been asked to deploy a gatekeeper to support CUBE that will connect your organizational domain to the domain of an Internet Telephony Service Provider so that callers can reach the 407 area code. Which configuration will support this function?

A.
gatekeeper
zone local GKVIA acme.com 192.168.10.1
zone remote GK407 ITSP.com 10.10.1.100
zone prefix GK407…….
no shutdown

B.
gatekeeper
zone local GKVIA acme.com 192.168.10.1
zone remote GK407 ITSP.com 10.10.1.100 1719 invia outvia GKVIA
zone prefix GK407 407*
no shutdown

C.
gatekeeper
zone local GKVIA acme.com 192.168.10.1
zone remote GK407 ITSP.com 10.10.1.100 1719 invia GK407 outvia GK407
zone prefix GK407…….
no shutdown

D.
gatekeeper
zone local GKVIA acme.com 192.168.10.1
zone remote GK407 ITSP.com 10.10.1.100 1719 invia GKVIA outvia GKVIA
zone prefix GK407 407*
no shutdown

Answer: D

Question 6

Refer to the exhibit. You have configured a gatekeeper and an IP-IP gateway on the same router. When you look at the output from the show gatekeeper endpoint command, the IP-IP gateway is not registered with the gatekeeper. What needs to be configured to resolve this issue?

A. You need to stop and restart the gateway.
B. You need to add a VoIP dial peer to the configuration.
C. The h323-gateway voip id command has an incorrect IP address.
D. The h323-gateway voip id command has an incorrect gatekeeper ID and IP address.

Answer: B

Question 7

Call Admission control (CAC) is a concept that applies to voice traffic only – not data traffic. Which two types are of Call Admission Control? (Choose two.)

A. resource-based
B. gatekeeper-controlled RSVP
C. local
D. QoS-based

Answer: A C

Explanation

There are 3 types of CAC:
+ Local CAC
+ Measurement Based CAC
+ Resource-Based CAC

For more information about these types, please read: http://www.cisco.com/en/US/docs/ios/solutions_docs/voip_solutions/CAC.html

Question 8

The SJ local zone contains a gatekeeper that controls two gateways, SJ1 and SJ2. Both gateways provide access to area code 408. For the following command strings, which two will be entered into the gatekeeper to give the SJ2 gateway priority over the SJ1 gateway? (Choose two.)

A. zone prefix SJ 408 gw -priority 10 SJ2
B. zone prefix SJ 408 gw-priority 10 SJ1
C. zone prefix SJ 408 gw-priority 6 SJ2
D. zone prefix SJ 408 gw-priority 6 SJ1

Answer: A D

Question 9

You are a Acme network administrator, your new task is to deploy a gatekeeper to support CUBE that will connect your organizational domain to the domain of an Internet Telephony Service Provider so that callers can reach the 407 area code. Which configuration will support this function?

A.
gatekeeper
zone local GKVIA acme.com 192.168.10.1
zone remote GK407 ITSP.com 10.10.1.100
zone prefix GK407 407
no shutdown

B.
gatekeeper
zone local GKVIA acme.com 192.168.10.1
zone remote GK407 ITSP.com 10.10.1.100 1719 invia outvia GKVIA
zone prefix GK407 407*
no shutdown

C.
gatekeeper
zone local GKVIA acme.com 192.168.10.1
zone remote GK407 ITSP.com 10.10.1.100 1719 invia GK407 outvia GK407
zone prefix GK407 407
no shutdown

D.
gatekeeper
zone local GKVIA acme.com 192.168.10.1
zone remote GK407 ITSP.com 10.10.1.100 1719 invia GKVIA outvia GKVIA
zone prefix GK407 407*
no shutdown

Answer: D

Question 10

You are the director of the Acme VoIP network, based on the exhibit. You have a client that is testing a directory gatekeeper in the lab to provide address resolution between two different zones. Two of the commands in the running-config output are incorrect. Which two changes will correct the configuration? (Choose two.)

A.
replace
zone local GK-A acme.com 172.16.14.44 1719
with
zone remote GK-A acme.com 172.16.14.44 1719

B.
replace
zone local DGK acme.com
with
zone remote DGK acme.com

C.
replace
zone prefix GK-B 404….
with
zone prefix GK-B 404……..

D.
replace
zone prefix GK-A 770…….
with
zone prefix GK-A 770….

Answer: A D

Question 1

The proper call-signaling term to the correct box in the diagram to establish RSVF-based Call Admission Control between the two Cisco Unifield Border Elements: Cisco UBEs. Some option is may be user more than once.

Answer:

Explanation

Here is how the call is established with RSVF-based Call Admission Control

1) The Cisco Unified Communications Manager (at the left-side) sends an H.225 setup to the Cisco UBE.
2) The Cisco UBE processes the call setup information and associates an outbound VoIP dial peer requiring an RSVP reservation. The Cisco UBE sends out an RSVP Reservation request to the remote Cisco UBE.
3) The remote Cisco UBE acknowledges the reservation and initiates the reservation for the return path, which is acknowledged by the local Cisco UBE.
4) The H.225 setup message is routed to the remote Cisco UBE, which then routes the call to the outbound VoIP dial peer pointing to Cisco Unified Communications Manager (at the right-side).
5) H.245 negotiation occurs with media flow-through enabled.
6) The call is established.

Question 2

Click and drag the type of call on the above to the type of voice port it applies to on the below.

Answer:

1) T1 or E1 with CAS or PRI: PBX to PBX
2) FXO: off-net
3) FXS: local
4) FXS or switch: on-net
5) E&M, FXO, FXS: PLAR

Explanation

First let’s have a quick review of these types of calls:

Local calls are calls that occur when both the calling and called phones are attached to the same router.

On-net calls are calls that need more than one router. For example, the calling phone is from one router and the called phone attaches to another router. But notice that these routers are part of the same network.

Off-net calls are calls that originate on a router but terminate on the PSTN.

PLAR calls occur when a caller picks up a phone and the phone automatically dials a preconfigured number.

PBX to PBX calls are on-net calls, where the source and destination are PBXs.

Next we will explain the answers above:

PBX to PBX connections can use T1 or E1 with CAS or PRI: Nowadays, we often make PBX connections to a network through T1 or E1 lines with channel associated signaling (CAS) or Primary Rate Interface (PRI) signaling.

For off-net calls, the typical connection between the router and the PSTN is through FXO port.

A local call just need FXS ports so it is the only choice for this type of call.

We can make on-net calls through FXS port (phone directly connected to the router) or FXO port (phone connected to a PBX). The “switch” here means that we can connect an IP phone through a switch and place on-net calls through Cisco Unified Communications Manager.

A PLAR call can work with any type of signaling, including E&M, FXO, FXS interfaces.

Question 3

Assume a SIP voice network. Drag each characteristic to the type of SIP call setup the characteristics best describes.

Answer:

Direct call setup:
+ Nonscalable
+ UA must keep data on large number of destinations
+ Relies on cached information to resolve addresses

Redirect Server Call Setup:
+ Server reports back to a UA with destination coordinates

Proxy Server Call Setup:
+ Most dynamic address resolution capability
+ All setup messages to through server
+ UA incapable of establishing its own sessions

Question 4

Which item correctly describes the relationships between the feature and the category it belongs?

A. Gateway – 1 and 2
CUBE – 3 and 4

B. Gateway – 1 and 3
CUBE – 2 and 4

C. Gateway – 2 and 3
CUBE  – 1 and 4

D. Gateway – 2 and 4
CUBE – 1 and 3

(Note: In the real exam, this question may be represented as a drag and drop question)

Answer: A

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Question 1

Which type of voice port will allow the gateway to terminate 23 or 30 circuits from the PSTN or a PBX?

A. FXO
B. FXS
C. PRI T1/E1
D. E&M
E. BRI

Answer: C

Explanation

The ISDN E1 PRI has 32 timeslots (channels). Each timeslot is 8 bits and has a data rate of 64,000 bits/second. Timeslot 0 is used for frame synchronization and alarms. Timeslot 16 is used for signaling so we can use 30 timeslots to carry calls.

The T1 PRI only has 24 timeslots (channels). Timeslot 0 is used for frame synchronization. Unlike E1 (which uses a dedicated timeslot for signaling) T1 Channel Associated Signaling (CAS) “steals” the 8th bit of the sixth frame for signaling information so we can use 23 timeslots for voice.

Question 2

Which is the best description of time-division multiplexing?

A. A channel is assigned an exclusive slice of the overall frequency of the circuit for the entire time of its operation.
B. All sources get an interleaved slice of time, which offers the entire frequency range allocated for that timeslot.
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. Technology that increases the transmission capabilities by dividing the medium into multiple channels that are each assigned a wavelength based on statistical analysis.
E. On a T1 circuit, this is the process where 24 DS-1 signals are multiplexed into a single DS-0 channel, while on a T3 circuit 24 DS-0 signals are multiplexed into a single DS-3 signal.

Answer: B

Explanation

Time-division multiplexing (TDM) is a method of putting multiple data streams in a single signal by separating the signal into many segments, each having a very short duration. Each individual data stream is reassembled at the receiving end based on the timing.

The TDM technology is based on the time, not on the frequency like frequency-division multiplexing (FDM) technology so each timeslot can have any frequency. Notice in the telecommunication field, there are technologies that use both TDM and FDM. The most popular example is GSM (in particular, it uses TDMA & FDMA).

Question 3

Refer to the exhibit.

Which step of digitizing analog signals does this represent?

A. Encoding
B. Quantization
C. Signal sample
D. Signal compression

Answer: A

Explanation

There are many definitions about voice encoding but I would like to refer it as the process of translating digital numbers into binary values. It makes us remember the concept more easily. Each time we see a binary-conversion process, we can say it is the encoding step ^^.

Questions 4

Refer to the exhibit.

Which step of digitizing analog signal does this represent?

A. Signal sample
B. Signal compression
C. Sample quantization
D. 8-bit digital encoding

Answer: A

Explanation

As you can see in the picture, an analog waveform is sampled to covert into a numeric value by a device described in question 2′s explanation. This process is called Pulse-Amplitude-Modulation (PAM). The results of this step are often decimal numbers. Later, these decimal numbers are converted to binary numbers by the encoding process.

Question 5

What is a numbering plan?

A. It describes the path from one endpoint to another.
B. It describes the calling privileges of an endpoint.
C. It describes how digits are manipulated in processing a call.
D. It describes call coverage in a dial plan.
E. It describes the endpoint addressing used in a voice system.

Answer: E

Explanation

A numbering plan is a world-wide standard to organize and locate telephones all around the world. In CCNA Voice, you often read about E.164 standard.

Question 6

Which three of these are part of the e.164 number in the ITU-T numbering plan for geographic areas? (Choose three)

A. SANC.SPID
B. Station code
C. Country code
D. Subscriber number
E. Numbering plan area
F. National destination code

Answer: C D F

Explanation

E.164 is an international numbering plan created by the International Telecommunication Union (ITU). Each number in the E.164 numbering plan
contains the following components:

• Country code (CC)
• National destination code (NDC – optional)
• Subscriber number (SN)

The CC consists of one, two or three digits. It is what we add in order to access different countries and often prefixed with a +

The NDC is the code we often call the area code. It is optional but for geographic areas we have to use it.

The SN is for telephone numbering. It is given by your phone operator.

For example, the North American Numbering Plan E.164 is as follows:

1-602-555-1212

+ 1: Country code
+ 602555: National destination code (for North American Numbering Plan, 602 is called the area code while 555 is called Central Office Code)
+ 1212: Subscribe Number

E.164 numbers are limited to a maximum length of 15 digits.

Question 7

Which statement is true concerning time-division multiplexing?

A. Time-division multiplexing transmits one voice signal at a time over a four-wire path.
B. Time-division multiplexing consecutively transmits multiple voice signals across four separate communication mediums.
C. Time-division multiplexing simultaneously transmits multiple separate voice signals over one communication medium by quickly interleaving pieces of each signal, one after another.
D. Time-division multiplexing consecutively 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.

Answer: C

Explanation

Time-division multiplexing (TDM) is the technology of putting multiple data streams in a single signal by separating the signal into many segments, each having a very short duration. Each individual data stream is reassembled at the receiving end based on the timing.

Answer A is incorrect because TDM can transmit multiple voice signals at a time. In fact, it is the most important advantage of TDM technology.

Answer B is incorrect as it says “consecutively”, TDM can transmit multiple voice signals simultaneously.

Answer D is long but not correct when stating “consecutively transmits one voice signal at a time”. Just for your information, D would be correct with this statement:

“Time-division multiplexing simultaneously transmits multiple voice signals 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 (first-in first-out). Information from each data channel is allocated bandwidth fixedly, no matter whether there is data that needs to be transmitted or not”.

In fact, we often use some codecs that support bandwidth saving when there is no sound (silence) in a conversation, like G.729b codec.

Question 8

The Point North Company has a few slow links in its voice and data network. Which two techniques can be used to reduce delay in voice transmission? (Choose two)

A. FIFO queuing
B. Buffering voice packets
C. Fragmentation of large packets
D. Compression of IP, RTP, and UDP headers
E. Increasing priority queue sizes

Answer: C D

Explanation

First In First Out (FIFO) queuing algorithm is the simplest of the congestion management methods. All packets are treated equally, and placed into a single queue and serviced the order they were received; hence the name FIRST-IN FIRST-OUT. But when talking about “queuing”, we also mention about delay because the queuing time (or queuing delay) is the time a job waits in a queue until it can be executed. In conclusion, queuing increases delay -> A is not correct.

Buffering also increases delay as queuing -> B is not correct.

In mixed voice/data IP networks, packets must be fragmented prior to traversing bandwidth-limited (less than 1 Mbps) connections to minimize voice delay and jitter. Two types of fragmentation that are universal and not limited to a specific link layer technology (such as ATM or Frame Relay) are IP and PPP fragmentation.

IP fragmentation adjusts the packet (maximum transmission unit [MTU]) size for all packets traversing the router. PPP fragmentation splits large packets into multiple smaller packets and encapsulates them into PPP frames before queuing and transmission. Recombination is done at the other end of the link -> C is correct.

Compression of IP, RTP, and UDP headers decreases consumption of available bandwidth for voice traffic. A corresponding reduction in delay is realized -> D is correct. But notice that this method should only be used for slow connection as it will take more resources of the receiving device to decompress them.

One thing I want to notice is that RTCP only provides feedback on the quality of the transmission link. It does not make any guarantees concerning quality of service.

Congestion management features allow you to control congestion by determining the order in which packets are sent out an interface based on priorities assigned to those packets.

With Priority queueing (PQ), packets belonging to one priority class of traffic are sent before all lower priority traffic to ensure timely delivery of those packets.

Increasing priority queue sizes will increase the maximum number of packets the queue can hold for a class policy configured in a policy map. This can reduce delay in high priority class but suffer other lower priority classes.

(Read more about congestion management at http://www.cisco.com/en/US/docs/ios/12_2/qos/configuration/guide/qcfconmg_ps1835_TSD_Products_Configuration_Guide_Chapter.html)

Question 9

What is required to convert a G711ulaw call to G729?

A. Voice Termination resources
B. Conferencing resources
C. Converter resources
D. Transcoding resources

Answer: D

Explanation

Transcoding compresses and decompresses voice streams to match endpoint-device capabilities. Transcoding is required when an incoming voice stream is digitized and compressed (by means of a codec) to save bandwidth, and the local device does not support that type of compression -> D is the correct answer.

Question 10

Which codec is less processor intensive?

A. G729
B. G729a

Answer: B

Explanation

G.729a is a compatible extension of G.729, but requires less computational power. This lower complexity, however, bears the cost of marginally reduced speech quality.

Question 11

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

Answer: B C D

Question 1

Refer to the exhibit.

Which identifies the amplitude of an analog signal stream?

A. A
B. B
C. C
D. A & C
E. Voltage/Time
F. Voltage/Time x B/C

Answer: C

Question 2

Which three characteristics apply to ISDN PRI? (Choose three)

A. The D channel is 16 kbps
B. The D channel is 64 kbps
C. Can carry data, voice, or video
D. Cannot support call forwarding
E. Commonly used only in Europe
F. Can carry vendor-specific PBX features

Answer: B C F

Question 3

What are two benefits of using an ITSP for long distance telephony services? (Choose two)

A. Connection to an ITSP is very granular and can provision from 1 to hundreds of simultaneous calls.
B. Connection to an ITSP is only available in full T1/E1/PRI circuit quantities.
C. The circuits are dedicated only to voice.
D. Connection to an ITSP is easier to configure because transcoders can seamlessly convert H.323 or SIP to SCCP.
E. 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.

Answer: A E

Question 4

Which type of voice port will terminate a loop start or ground start line from the PSTN or a PBX?

A. L-FXO
B. FXS/FXO
C. PRI T1/E1
D. E&M
E. BRI
F. CAS T1/E1

Answer: B

Question 5

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

Answer: D

Question 6

Which four types of high-density trunk can be used to connect to an ITSP? (Choose four)

A. T1/E1 CAS
B. FXS
C. T1/E1 PRI
D. E&M
E. SIP
F. H.323

Answer: A B C D

Question 7

What are three functions provided by a switching system in a traditional telephony network? (Choose three)

A. Call setup
B. Call supervision
C. Quality of service
D. Codec processing
E. Voice compression
F. Customer IDs and telephone numbers

Answer: A B F

Question 8

An analog telephone is connected to a ________ port on a router?

A. FXO
B. E1
C. T1
D. FXS

Answer: D

Question 1

In which of the following scenarios would a Cisco switch supply PoE to an IP phone?

A. In any cases as long as the switch and IP phone are Cisco products.
B. If the Cisco switch supports prestandard PoE and the IP phone supports 802.3af.
C. If the Cisco switch and IP phone both support a common PoE method.
D. Only if both the switch and IP phone use 802.3af PoE.

Answer: C

Question 2

Refer to the exhibit. What does the parameter Overlay on button number 2 mean?

A. Overlay means that this IP phone has more than one extension configured on line 2.
B. Overlay means any calls to line 1 when busy will be automatically forwarded to line 2.
C. Overlay means that the user can dial using any Overlay application, for example, Click to dial application.
D. Overlay 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.

Answer: A

Question 3

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

Answer: B

Question 4

What is the key command to use when deploying the partially automated telephone setup process?

A. Auto qos
B. Auto assign
C. Auto start-dn
D. Auto phone-type
E. Auto start-ephone-dn
F. Auto telephony-service

Answer: B

Question 5

Which two commands need to be added to this configuration to allow IP Phones in subnet 10.100.1.0 to register with Cisco Unified Communications Manager Express at address 10.10.1.1? (Choose two)

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
E. Encapsulation dot1q 10 native under interface FastEthernet0/0.10

Answer: B C

Question 1

Refer to the exhibit.
Which Cisco Unified Communications layer is the call processing layer?

A. A
B. B
C. C
D. D

Answer: C

Question 2

Which value does an administrator assign to option 150 for DHCP to operate correctly in a Cisco Unified Communications Manager Express environment?

A. IP address of the DNS server
B. IP address of the TFTP server
C. MAC address of the TFTP server
D. MAC address of the DHCP server
E. MAC address of the Cisco Unified Communications Manager Express router
F. IP address of the PC on which the Cisco Unity Express module is installed

Answer: B

Question 3

Which two are considered endpoints in a Cisco Unified Communications solution? (Choose two)

A. Gateway
B. Call agent
C. IP telephone
D. Analog phone
E. H.323 gatekeeper
F. Cisco Unified Communications Manager

Answer: A C

Question 4

Refer to the exhibit.

Which statement about the Cisco Unified Communications Manager Express configuration is correct?

A. 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.
B. MoH will not work because the multicast address is not correct.
C. MoH will work as is. Nothing needs to be added.
D. The configuration will work only for calls that are being transferred because the transfersystem command has been added.

Answer: C

Question 5

A customer is in the planning stages of deploying a Cisco Unified Communications solution for their company. Previously, they were leasing a traditional PBX system from the telco and they have very little experience with voice. The customer wants to know what two signaling methods between the IP phone and the Cisco Unified Communications Manager Express are available for their use. (Choose two)

A. H.323
B. SCCP
C. MGCP
D. SIP
E. RTP

Answer: B D

Questions 6

Which best describes the auto-attendant in a Cisco Unified Communications environment?

A. A set of call processing instructions that automatically tell the system what to do when it reaches a particular system ID
B. A system that automatically allows inside or outside callers to leave voice-mail messages 24 hours, 7 days a week, even when no operator is on duty
C. A function that 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. A function that 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

Answer: C

Questions 7

Which statement about the Cisco Unity Express default AutoAttendant is true?

A. The default AutoAttendant must be enabled during the Initialization Wizard process otherwise voice-mail services will not function.
B. Enabling the default AutoAttendant is not mandatory during the Initialization Wizard process.
C. The default AutoAttendant is enabled by default with the exception of the prompts that must be recorded via AvT.
D. The default AutoAttendant cannot be used as is, and it must be customized for the particular environment it will be used in.

Answer: B

Question 8

You have been tasked 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. 1
B. 2
C. 3
D. 4
E. 5

Answer: A

Question 9

Refer to the exhibit.

What will happen to a call that is transferred when no second line is available?

A. It will fall back to a full-blind transfer.
B. It will be transferred without consultation using H.450.2 standard methods.
C. It will fall back to a blind transfer for nonlocal consultation or for a nonlocal transfer target.
D. It will be transferred without consultation using the proprietary Cisco transfer method.

Answer: A

Explanation

The full-consult parameter will perform call transfers with consultation using second phone line if available and fallback to full-blind if second line unavailable (full-blind: Perform call transfers without consultation). For example, you are talking to Mr.A but you want him to speak with another person, Mr.B, then you can press the “Trnsfer” softkey (1st time). After hearing another dial tone, you can dial Mr.B’s phone number waiting him to answer the phone.

At this time, if your configuration is “full-consult” then you can continue speaking with Mr.B until you press “Trnsfer” softkey a second time. This will drop you out of the conversation and let Mr.A and Mr.B speak with each other.

If your configuration is “full-blind”, the phone will automatically transfer the call right after you have dialed Mr.B number and you don’t have a chance to speak with Mr.B before transferring your conversation. This configuration does not require you press the “Trnsfer” softkey a second time.

But the “full-consult” method can only work when the second line is available, if not, it will fall back to “full-blind” method.

(Notice: “Trnsfer” is the text on your phone, not “Transfer”)

Question 10

Which three commands does Cisco recommend to use first when setting up phones for Cisco Unified Communications Manager Express? (Choose three)

A. Load
B. Max-dn
C. Keepalive
D. Max-ephones
E. Create cnf files
F. Telephony-service

Answer: B D F

Question 11

The XYZ Corporation is migrating from a traditional PBX to a Cisco Unified Communications system. While the migration is taking place they need to provide common channel signaling connectivity to the PBX from the Cisco Unified Communications system. Which type of gateway would be required?

A. Analog station gateway
B. Digital trunk gateway
C. Analog trunk gateway
D. Digital station gateway

Answer: B

Question 12

What statement about the Cisco Unity Express AIM Module and the Cisco Unity Express Network Module is true?

A. The AIM module takes up a router slot whereas the network module does not.
B. The network module takes up a router slot whereas the AIM module does not.
C. The AIM module runs the Linux OS whereas the network module runs the Cisco IOS software.
D. The network module requires the Cisco Unified Communications Manager Express to be preconfigured whereas the AIM module does not.

Answer: B

Question 13

The Woodhull Ink Company has successfully installed the CUE module in its router’s chassis. Which two configuration items are necessary for the module to function properly? (Choose two)

A. A default gateway must be assigned to the service module.
B. The ip numbered command must be used to save the subnet.
C. A subnet must be created 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.
E. The IP address of the service engine must be static and assigned to the interface to avoid the need for a new subnet.
F. Two virtual IP addresses are needed to represent the two ends of the physical Ethernet connection across the backplane.

Answer: A D

Question 14

Refer to the exhibit.

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)

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.

Answer: C D

Question 15

The SBB Company is setting up Call Transfer for its Cisco Unified Communications Manager Express solution. The company uses five-digit extensions and would like to be able to transfer calls outside the network to the CEO’s home. The CEO’s telephone number is 866-555-2222. Which configuration command will allow this to occur?

A. SBB(config-ephone-dn)# transfer-pattern .T
B. SBB(config-telephony-service)# transfer-pattern .T
C. SBB(config-ephone-dn)# transfer-pattern ……51212
D. SBB(config-ephone-dn)# transfer-pattern 8665552222
E. SBB(config-telephony-service)# transfer-pattern 86655…..
F. SBB(config-telephony-service)# transfer-pattern 8665552222

Answer: F

Question 16

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

Answer: D

Question 17

What is the maximum number of phones are supported on Call Manager Express?

A. 48
B. 28
C. 240
D. 500

Answer: C

Question 18

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
C. MAC address of the TFTP server
D. IP address of the TFTP server

Answer: D

Question 19

What device enables Call Admission Control in a CME environment?

A. Gateway
B. Transcoder
C. Bridge
D. Gatekeeper

Answer: D

Question 20

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

Answer: D

Question 21

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

Answer: B

Question 22

What does this command do “dtmf-relay sip-notify”?

Not sure about the options but the answer should be something like this:

This command sets the SIP DTMF relay mechanism to use Unsolicited-Notify messages to relay incoming and outgoing DTMF signals.

Question 1

Refer to the exhibit.
Which two statements about SIP trunk are true? (Choose two.)

A. A SIP trunk configuration is always needed for a UC500 device.
B. A SIP trunk is needed only to provide internet access for your data users.
C. SIP trunk configuration parameters should be provided to your service provider.
D. A SIP trunk is needed only if you are using voice mail to supply the Message Waiting Indicator value to the Cisco Unity Express module.
E. A SIP trunk is needed only for voice if you are planning on using VoIP through a service provider.
F. A SIP trunk is not supported in a keyswitch configuration.

Answer: E F

Question 2

A customer wants to use a signaling protocol on the voice gateways that require registration with the Cisco Unified Communications Manager. What protocol should be recommended?

A. SIP
B. H.323
C. SRTP
D. MGCP

Answer: D

Question 3

Refer to the exhibit.

After deploying 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 have configuration parameters that are most likely going to resolve this issue? (Choose two)

A. Device
B. System
C. Network
D. SIP Trunk
E. Voice Features
F. Dial Plan

Answer: E F

Question 4

Refer to the exhibit.

You have been asked to verify the network configuration that was performed by a colleague. Which of the following conclusions would be accurate?

A. The voice VLAN should always start with VLAN 1.
B. The subnet mask for the network address should be class A instead of class C.
C. The CME IP address should be in a different subnet to that of the IP phones.
D. The CME IP address should be in the exclusion range.

Answer: D

Question 5

Refer to the exhibit.

What is the purpose of access code 9?

A. This access code connects a caller to the operator when they press 9 on their phone keypad.
B. This is a level 1 security to ensure that only users that know the access code can make calls.
C. This is the access code that internal users must use to make PSTN calls.
D. This access code is required when interfacing with a traditional PBX system.

Answer: C

Question 6

Refer to the exhibit.

Which statement is true about the VLAN Number field?

A. You can create a new voice VLAN as long as it falls in the range 1 to 1001.
B. You can create a new voice and data VLAN as long as it falls within the range 1 to 1001.
C. You can create a voice VLAN as long as it falls in the range of 1 to 100.
D. You can only assign a voice VLAN here in the range of 1 to 1001.
E. You can create and assign a voice VLAN in the range of 1 to 1001.

Answer: D

Question 7

Refer to the exhibit.

Based on this configuration, how will an incoming call to 503 be routed?

A. The call will be routed to extension 503 and then sequentially to hunt groups 1 and 2.
B. The call will be routed to extension 503 and then sequentially to hunt groups 2 and 1.
C. The call will be routed sequentially to users that belong to hunt group 3.
D. The call will be routed sequentially to users that belong to hunt group 3 and then sequentially to hunt groups 1 and 2.

Answer: C

Question 8

Refer to the exhibit.

What is the difference between a PBX and a keysystem under the Device tab?

A. A PBX is where each phone has nearly identical configuration whereas a keysystem has a unique extension for each phone.
B. A PBX is where each phone has a unique extension whereas a keysystem has nearly identical configurations for each phone.
C. A keysystem can be activated only by the operator by turning the Systemkey. A PBX operates 24×7 and requires no activation.
D. Incoming PSTN calls to keysystem users have to be routed through an auto-attendant or an operator. Incoming PSTN calls to PBX users can be answered by any user on any line.

Answer: B

Question 9

Which three options can be configured using Cisco Configuration Assistant? (Choose three)

A. Voice VLAN only
B. Voice and data VLANs
C. Script selection for the AutoAttendant
D. Sip trunk to an ITSP
E. Voice-mail archive

Answer: B C D

Question 10

Benson’s Able Messenger Service wants to install a new UC500 solution. Which four of these provide information from the existing numbering plan that helps you design an effective VoIP dial plan? (Choose four)

A. Station number
B. Subscriber code
C. Direct inward dial ranges
D. Length of internal number extensions
E. Number of digits forwarded by the local CO
F. Cisco Unified Communications Manager route pattern used for internal calls

Answer: A C D E

Question 11

Refer to the exhibit.

You have created a new voice VLAN 110 and would like to reassign the IP phones to the new voice VLAN. The IP phones currently reside in the default voice VLAN 100. Which tab allows you to reassign the IP phones?

A. Device
B. System
C. Network
D. Voice Features

Answer: C

Question 12

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

Answer: A

Question 13

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.

Answer: B

Question 1

Drag the term Gateway and drop it into the locations that require a voice gateway

Answer:

Question 2

Answer:

London, United Kingdom:

+ CAS E1
+ 2.048 Mbps
+ 30 voice channels
+ out-of-ban signaling in timeslot 17

Chicago, United States:

+ CAS T1
+ 1.544 Mbps
+ RBS in-band
+ 24 voice channels

Question 3

Click and drag the protocol on the left to the matching characteristic on the right. Not all options will be used.

Answer:

Peer-to-Peer:

+ SIP
+ H.323

Client/Server:

+ SCCP
+ MGCP

Question 4

Drag the correct length arrow(s) that represent call legs and drop them into the proper position to show how call leg(s) are used in a call. Arrows can be used more than once, and not all may apply.

Answer:

Question 5

Drag the correct Call Leg and drop it in the proper position to provide call setup in both directions. Arrows may be used more than once , and not all may apply.

Answer:

Question 6

Place the steps for inbound dial-peer matching in the correct order

Answer:

1) Look for the incoming called-number command in the dial peer that matches the called number or DNIS string in the inbound call leg.
2) Look for the answer-address command in a dial peer that matches the calling number or ANI string of the inbound call leg.
3) Look for the destination-pattern command in a dial peer that matches the calling number or ANI string of the incoming call leg.
4) Look for the POTS dial peer port configuration that matches the voice port associated with the incoming call (POTS dial peers only).
5) If no matches are found, the system uses the default dial peer.

Question 7

Three tasks are necessary to configure IP addressing for Cisco Unity Express hardware. The phrase that begins the statement for these tasks is on the right. Click and drag the phrase from the left to the box on the right that correctly completes the statement for each task.

Answer:

- Configure the service engine interface: with a static IP address or IP address
– Configure the service-module IP address: to be on the same subnet as the router
– Configure the Cisco Unity Express IP default gateway: to be the same as the service engine

Question 8

Click and drag the description on the left to the signaling type it corresponds to on the right. Not all may apply.

Answer:

1) Address Signaling: digits dialed or called party number that can be system -specific or variant-specific.
2) Supervisory Signaling: events that occur on the trunk, including seizure, wink, and answer.
3) Informational Signaling: tones such as ringing or busy and announcements such as “no longer in service”.

Question 9

Drag the steps for converting analog signals to digital signals.

Answer:

Step 1:  Sampling
Step 2: Quantization
Step 3: Encoding
Step 4: Compression (optional)

Explanation

Below is a summary for each step:

1) Sampling: The sampling rate must be at least twice the highest frequency, to accurately represent the original signal.
2) Quantization: consists of a scale made up of 8 major divisions or chords. Each chord is subdivided into 16 equally spaced steps. The chords are not equally spaced but are actually finest near the origin.
3) Encoding: encode the value into an 8-bit digital form.
4) Compression: compress the samples to reduce bandwidth. Although not required but it is widely used.

Question 1

What is the difference between voice VLAN and native VLAN?

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

Answer: C

Explanation

For 802.1Q trunking, one VLAN is not tagged. This VLAN is called native VLAN. The native VLAN is used for untagged traffic when the port is in 802.1Q trunking mode. For Voice VLAN, we need to specific a VLAN to use for voice traffic.

Question 2

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

Answer: B

Explanation

Cisco IP phones will be able to receive voice VLAN configuration via CDP. Once it has received the voice VLAN idr, the IP phone will begin tagging its own packets. Non-Cisco IP phones will not be able to understand CDP packets.

Question 3

Which command is used to assign voice VLAN 110 on a Cisco IOS switch?

A. switchport access vlan 110
B. switchport access voice vlan 110
C. switchport voice vlan 110
D. switchport voice access vlan 110

Answer: C

Explanation

The syntax of this command is “switchport voice vlan {vlan-id | dot1p | untagged | none}”

Question 1

Refer to the exhibit.

What information can be gleaned from the output of the show ephone command?

A. There are two registered IP phones. Shared number 5010 on line 2. Message waiting on line 1 of phone 1.
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 1.
E. There are two IP phones. Phone 2 is unregistered. Shared number 5010 on line 2 of phone 2. Message waiting on line 2 of phone 1.

Answer: D

Explanation

The first Ephone’s status (ephone-1) is REGISTERED so we can confirm this ephone has been registered. The second one (ephone-2) shows DECEASED, it means that the CME router has lost connectivity with this IP Phone through a TCP keepalive failure.

Notice: The status UNREGISTERED indicates the CME router closed the connection to the IP phone in a normal manner.

Question 2

Refer to the exhibit.

The configuration for Call Park has been performed by a colleague before going on vacation. Users are complaining that the Call Park softkey does not appear on their phones. What can rectify this issue?

A. The park-slot number ephone-dn 5 needs to be added as a line to one of the ephone 1 buttons.
B. The phones need to be restarted.
C. The park-slot number should be outside the range of 5XXX numbers.
D. The call-park command enable needs to be added under telephony-service.

Answer: B

Explanation

The park softkey is displayed on your phone only if you have created at least one park slot for your Cisco CME system. When you press the park softkey, the call is transferred to a park slot that has the number that most closely matches the extension number you used to answer the call. So if you are using extension 5001 and you park a call from that extension, the call is parked in a park slot with a number ending in 001 if available. For example, if you create park slots with numbers 6000, 6001, 6002, and 6003, a call parked from extension 5001 uses park slot 6001 if possible. If no park slot with a matching number is available, any available park slot is used.

When you park a call, the park-slot number selected is displayed on your IP phone’s display. To retrieve a parked call, simply press the pickup softkey followed by the park-slot number. To retrieve the last call parked by your phone (or notified to your phone), simply press the pickup softkey followed by the star (*) key on the phone’s keypad.

In short, the command park-slot timeout 10 limit 3 notify 5001 will create a park slot instance with number 5001. It sets the reminder interval to 10 seconds and configures 3 as the maximum number of reminders. As soon as the maximum number of reminders has been sent (after 3 * 10 = 30 seconds), the call in the park slot is disconnected. Each time the reminder interval is reached, a reminder notification is sent to all IP phones with extension 5001.

After you have created your first park slot within your Cisco CME system, you need to reset or restart your IP phone(s) before the park softkey becomes visible on the phone.

Question 3

Refer to the exhibit.

When a call is placed to 2000, phones 1 and 2 are paged. A call to 2001 pages phones 3 and 4.
What command is missing so that a call to 2002 pages all four phones?

A. Paging group 20 21
B. Paging group 10 20 30 40
C. Paging group all
D. Paging group 2000 2001 2002
E. Paging group ephones-all

Answer: A

Explanation

In fact, the exhibit should show the configuration of ephone-dn 20, something like this:

To page all four phones which belong to group 20 & 21, we just need to assign these two groups to a number. The full command is:

CME_Voice(config-ephone-dn)# paging group 20,21

Notice:

- We don’t need to assign any ephones to paging-dn 22 because this ephone-dn represents a group of both paging-dns 20 and 21.

- The IP address that follows the paging command is a multicast address.

Question 4

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
E. Dual-number ephone-dn
F. Two ephone-dns with one number

Answer: A

Explanation

Shared line (or shared ephone-dn) means that we assign the same ephone-dn to multiple ephones. Incoming calls to number 1006 will ring on both phones.

Question 5

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
E. Dual-number ephone-dn
F. Two ephone-dns with one number

Answer: F

Explanation

We can see in both ephone-dns 13 & 14 the number 1003 command is used. It means that if someone dials 1003, one of the two phones will ring. If we don’t assign different priorities on these 2 phones, the CME system will pick one completely at random so sometimes ephone 4 will get the call and sometimes ephone 5 will get the call.

To gain more control over how the call flows than a random line selection, we can use the preference command. This command dictates which ephone-dn is more preferred than the other by assigning a value from 0 to 10, where the lower preference numbers are better. So in the exhibit, if we ignore the “no huntstop” command (will be explained later), the ephone 4 will always ring first because the default preference value is 0 which is lower than the preference value of ephone-dn 14 (preference 1). The ephone 5 will ring only if the ephone 4 is currently busy. But this system will make the person on phone 4 very busy while the one on phone 5 idle.

A better system might have all the IP phones ring when on a call to 1003 and whoever picks up the phone first will answer the call. This is where the “no huntstop” comes into play.

The “no huntstop” command tells the CME system, “continue hunting for other matches with this ephone-dn”. And the system finds out the ephone-dn 14 has the number 1003 configured -> The system will ring phone 5 too.

(For more information, please read http://www.computerfreetips.com/Cisco_Unified_Communications_Manager_Express/Shared-Line-Overlay-ehphone.html)

Question 1

Which type of delay can lead to jitter in a voice network?

A. Propagation delay
B. Serialization delay
C. CODEC delay
D. Queuing delay

Answer: D

Explanation

Jitter is the variation in the arrival of voice packets. For example, the first voice packet of a conversation might take 50 ms to reach a destination while the second voice packet might take 60 ms. There is 10 ms of delay variation (jitter) between these packets. The varying arrival time of the packets can cause gaps in the re-creation and  playback of the voice signal. These gaps are undesirable and annoy the listener. For example, if the speaker says “Enjoy your life” then the listener will hear “Ennnnnnjoy yooooour liiiiiiife”.

Queuing delay (how long a packet waits in a router’s interface queue) is variable because it depends on how many packets are currently in the queue. Therefore queuing delay is the main reason leading to jitter in a VoIP network.

Other delays (propagation, serialization, CODEC) are fixed and predictable delays.

+ Propagation: The time it takes a packet to traverse a link.
+ Serialization: The insertion of bits onto a link.
+ CODEC: The time for translating the audio signal into a digital signal.

Question 2

IP phone A places a call to IP phone B. How many RTP streams are required for the call to be successfully completed?

A. 1
B. 2
C. 4
D. 6

Answer: B

Explanation

Notice that RTP streams are one-way. If you are having a two-way conversation, the devices will establish dual RTP streams, one in each direction

Question 3

Refer to the exhibit.

The Acme Corporation needs assistance in configuring their PSTN voice gateway. Which two dial peers will correctly route calls to emergency services? (Choose two)

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

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

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

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

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

Answer: E F

Explanation

The first time I read this question, I think the Local PSAP (Public Service Answering Point) can accept both 911 and 9911 but it is not true. The Local PSAP only accept 911 so the duty of the administrator is to configure the gateway in order to support both 911 and 9911 numbers. To do this, we need two dial-peers, one for 911 and another for 9911.

But keep in mind that our outgoing dial-peer (port FXO 1/0/0) is a POTS dial-peer so the matched digits of this dial-peer will get stripped so we need to use the forward-digits all or forward-digits 3 (for 911 pattern) or prefix 911 (for 9911 pattern) to keep the called number. Therefore only E and F are correct.

Question 4

Approximately what percentage of voice packets can be dropped before voice quality becomes poor?

A. 1 to 2%
B. 15%
C. 5 to 10%
D. Less than or equal to 1%

Answer: D

Explanation

Packet loss describes an error condition in which data packets appear to be transmitted correctly at one end of a connection, but never arrive at the other. This might be because:

• network conditions are poor and the packet became damaged in transit
• The packet was deliberately dropped at a router because of internet congestion.

The 1% threshold is just an estimate. Some documents say that even with 1% packet loss can “significantly degrade” a VOIP call using G.711 or G.729 codec. But “1% or less” is the best answer for this question.

Question 5

How does LLQ help ensure that voice quality is maintained in a converged network?

A. LLQ allocates minimum bandwidth guaranteed to voice traffic.
B. LLQ allocates a priority queue to voice traffic at a guaranteed rate.
C. LLQ allocates a priority queue and a minimum guaranteed bandwidth queue for voice.
D. LLQ ensures that all traffic is treated fairly and hence voice traffic is not severely impacted.

Answer: B

Explanation

Low-latency queuing (LLQ) is used to give specific traffic classes higher priority when transmitting on the router’s WAN interface. Low Latency Queuing allows delay-sensitive data such as voice to be dequeued and sent first (before packets in other queues are dequeued), giving delay-sensitive data preferential treatment over other traffic.

The bandwidth given to an LLQ priority queue (PQ) is both the guaranteed minimum and policed maximum. This helps prevent the queue starvation that occurs with PQ.

Question 6

In which two situations would a voice gateway be required? (Choose two)

A. To connect a corporate or branch location to an IP WAN
B. To connect a corporate or branch location using VoIP to the PSTN
C. To connect a Cisco Unified Communications Manager to a LAN
D. To connect a Cisco Unified Communications network to a PBX
E. To connect a corporate or branch location to a MAN

Answer: B D

Question 7

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

Answer: D

Explanation

RTCP is used to monitor and provide control information about the quality of RTP streams but notice that RTCP only provides feedback on the quality of the transmission link. It does not make any guarantees concerning quality of service.

Question 8

Which three are components of a dial plan? (Choose three)

A. Call legs
B. Endpoint addressing
C. centralized control
D. Call coverage
E. Digit manipulation
F. Decentralized control

Answer: B D E

Explanation

We should understand what a dial plan is before talking about its components. In short, a dial plan is a collection of rules the call-processing agent uses to route calls. Below lists the components of a dial plan:

Endpoint addressing is the addressing scheme that is used to reach voice endpoints. For example, a company numbering plan might use four-digit extensions at each location and a three-digit site code. To call a phone at your own location, you would dial the four-digit extension. To call a phone at a remote company location, you would dial the site code and the extension.

Call coverage: Special groups of devices can be created to handle incoming calls for a certain service according to different rules, avoiding dropped calls. For example: top-down, circular hunt, longest idle, or broadcast groups are popular ones that you will see while learning CCNA Voice.

Digit manipulation: Digits can be manipulated prior to or after a routing decision has been made. In some cases, it is necessary to manipulate the dialed string before routing the call, for example, when you are rerouting over the PSTN a call originally dialed using the on-net access code, or when you are expanding an abbreviated code (such as 0 for the operator) to an extension.

Other two components of a dial peer are:

Calling privileges (or COR – class of service): Different groups of devices can be assigned to different classes of service, by granting or denying access to certain destinations or resources. For example, Employee phones might be allowed to reach only internal and local PSTN destinations, while Executive phones could have unrestricted PSTN access. The calling privileges assigned to a device are typically called class of service. In a Cisco voice
gateway, class of service is implemented by assigning Class of Restrictions (COR) to dial peers.

Path selection: Depending on the calling device, different paths can be selected to reach the same destination. Moreover, a secondary path can be used when the primary path is not available (for example, a call can be transparently rerouted over the PSTN during an IP WAN failure).

(Reference: CCVP – Implementing Cisco Voice Gateways and Gatekeeper & CVoice v6.0 Module 4 Lesson 1)

Question 9

Refer to the exhibit.

The configuration shows a dial peer that points to Cisco Unity Express. Which line of configuration is incorrect?

A. B
B. C
C. D
D. E
E. F
F. G

Answer: E

Explanation

We don’t have G.729ulaw, just G.711ulaw. G729 has 3 annexes that are G.729a, G.729b and G.729ab.

Question 10

Refer to the exhibit.

How many discrete call legs are needed to set up a call between the POTS phone attached to router 1 and the phone in the PSTN?

A. 3
B. 4
C. 6
D. 7
E. 8
F. 10

Answer: B

Explanation

We need four call legs as shown below

Question 11

Refer to the exhibit.

Which inbound dial peer on CMERouter1 will be matched when phone extension 1234 places a call to 2010?

A. Voip dial peer 30
B. Default dial peer 30
C. None, which will cause the call to drop
D. Default dial peer

Answer: D

Explanation

For CMERouter1 the “dial-peer voice 30 voip” will be matched for the outbound dial peer, not inbound one. When there is no dial-peer matched, the router will use the default dial peer.

Question 12

Refer to the exhibit.

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

A. Serial dial peer
B. PSTN dial peer
C. POTS dial peer
D. Network dial peer
E. VoIP dial peer

Answer: C E

Question 13

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.

Answer: B

Question 14

Which type of voice port will be most cost effective to allow the gateway to terminate two circuits from the PSTN or a PBX?

A. FXO
B. FXS
C. PRI T1
D. E1
E. E&M
F. BRI
G. CAS T1

Answer: E

Explanation

For PSTN and PBX connection, we need to use an analog interface type. E&M signaling is designed to connect directly to a PBX system that also supports E&M interfaces. Many PBX brands have E&M analog trunk cards that can operate as either the trunk circuit side or the signaling unit side and Cisco gateway does support E&M interfaces.

Question 15

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

Answer: B

Explanation

First, the gateway attempts to match the called number with the incoming called-number. If no match is found, the router or gateway attempts to match the calling number of the call set-up request with the answer-address of each dial-peers. If no match is found, it attempts to match the calling number of the call set-up request to the destination-pattern of each dial-peer.

Notice that these steps are just applied for inbound dial peer.

Question 16

Which protocol provides VoIP packet sequence numbering?

A. IP
B. TCP
C. UDP
D. RTP

Answer: D

Explanation

The RTP protocol provides end-to-end network functions and delivery services for delay-sensitive, real-time data, such as voice and video. It runs on top of UDP and provides these services:

• Payload-type identification
• Sequence numbering
• Time stamping
• Delivery monitoring

Question 17

Identify the VoIP network component that provides CAC, bandwidth control and management, and address translation.

A. Gateway
B. Gatekeeper
C. MCU
D. Call agent

Answer: B

Explanation

A gatekeeper can perform these tasks:

Address translation: The gatekeeper translates alias addresses (e.g., E.164 telephone numbers) to Transport Addresses, using a translation table that is updated using Registration messages and other means.

Bandwidth control: The gatekeeper controls how much bandwidth a terminal may use. The gatekeeper provides the above functions for terminals and gateways that have registered with it.

Bandwidth management: Limits the number of concurrent accesses to IP internetwork resources (gatekeeper-based CAC for bandwidth management) (CAC: Call Admission Control).

Question 18

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

Answer: B C D

Explanation

Notice that buffers give smoother audio playout but they does increase latency in VoIP network.

Question 19

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

A. Data link
B. IP
C. UDP
D. RTP

Answer: B C D

Explanation

Compressed Real-Time Transport Protocol (cRTP) compresses IP/UDP/RTP headers on low-speed serial links. We shouldn’t use cRTP on any high-speed interfaces as the price of CPU utilization is higher than the bandwidth savings.

Question 20

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 packet’s Layer 3 and Layer 4 headers in the routers at each end of a link, resulting in lower bandwidth demand for subsequent RTP packets

Answer: C

Explanation

While the Real-time Transport Control Protocol (RTCP) sounds very inportant, its primary job is just statistics reporting, which includes

• Packet count
• Packet Delay
• Packet Loss
• Jitter (delay variations)

These types of information are useful but not as important as the actual RTP audio streams. Keep this in mind to configure RTCP & RTP streams correctly in the future.

Question 21

Which two of the following VoIP gateway platforms are considered to be Integrated Services Routers (ISRs)? (Choose two)

A. Cisco 2600XM Series
B. Cisco 2800 Series
C. Cisco 3700 Series
D. Cisco 3800 Series

Answer: B D

Explanation

We can hardly find a complete definition for the Integrated Services Routers but you can understand ISR as following:

“An ISR integrates other network features into the router other than just routing features. Used mostly in small offices on ADSL lines, they offer things like VPN, firewall, and encryption services.”

or another definition:

“First, the ISR routers are devices with a low-performance CPU when comparing them to the usual workstation/server processors from Intel or AMD.

Second, they are, as their name suggests it, “integrated services routers”, i.e. universal devices capable of performing many diverse networking functions, and that is true. However, even if a device can provide a particular service, it does not mean that it has unlimited power for providing it, and also if a device supports various features, it does not necessarily mean that you can have all of them turned on and expect that they all will perform well under a high load. The ISR routers are very flexible, however, they are still considered to be, at least from the throughput point of view, low-end routers. Their strength is the versatility, not the raw throughput.”

Some benefits of Integrated Services Routers:

1. The ISRs are more cost effective than their legacy equivalents, particularly when the network requirements map to an existing bundle.
2. The ISRs are faster (up to five times) and can handle quite a bit more memory than the legacy platforms. The base configurations also have more memory.
3. The ISRs are designed with the ability to run multiple concurrent services (FW, NAT, IDS, QoS, etc.) at wire-speed.
4. All the ISRs have TWO built-in LAN connections – FE or GE.
5. All the ISRs have an embedded HW VPN accelerator – It is always included, it is just a matter of buying a VPN enabled image to turn it on. If that is not fast enough, a VPN AIM can be added to further enhance VPN performance.
6. The HWIC enabled slots provide an impressive 400Mbps of dedi-cated bandwidth (the old WICs provided up to 8Mbps). This is great news for LAN uplinks and Ethernet Switch HWICs. The NME slots offer up to 1.2Gbps per module (the standard NM was only 600Mbps).
7. The EVM slot offers high density digital/analog voice ports.
8. All the ISRs with  voice support have on-board DSP slots. There is no  need to use a NM slot for a network module with DSPs for voice applications anymore – the on-board DSP slots can provide enough DSP resources for most common requirements.
9. All the ISRs  with voice support can provide voice mail functionality with CUE (AIM and/or NM). CUE was not supported on the 1700 family.
10. All the ISRs that support voice can provide in-line power to Ethernet switch ports via a HWIC-ESW-POE or a NM-ESW-PWR (optional AC-IP power supply is required for in-line power).
11. Most ISRs provide some option for power supply redundancy. The 2811, 2821, 2851 and 3825 have a RPS connector and the 3845 can take a built-in redundant power supply.
12. For investment protection, the ISRs support most of the existing WICs, VICs, VWICs and NM modules (check the datasheets for de-tails).
13. SDM (Security Device Manager), included on all ISRs.

Question 1

A SIP Trunk is a logical connection between an IP PBX and a Service Provider’s 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 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.

Answer: C D

Question 2

What is the default VTP mode on Cisco switches?

A. Master
B. Client
C. Backup
D. Server

Answer: D

Question 3

What are the three VTP modes on Cisco switches?

A. Master
B. Client
C. Server
D. Transparent

Answer: B C D

Question 4

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

Answer: C

Question 5

When would you require a voice gateway? (Choose two)

A. When you conned 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

Answer: B D

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