300-835 : Automating Cisco Collaboration Solutions (CLAUTO) : Part 09

300-835 : Automating Cisco Collaboration Solutions (CLAUTO) : Part 09

1. You are integrating an existing Cisco Unity Connection user with an existing LDAP user. However, the user’s Unity Connection Alias field does not match the LDAP user ID.

   Which of the following will most likely occur if you attempt to use the BAT to import the Unity Connection user without updating the Alias field? (Select the best answer.)

   • A new Unity Connection user account will be created from the LDAP account.
   • A failed objects report will be created that contains information about the update attempt.
   • The Unity Connection user’s alias will be updated to match the LDAP user’s alias.
   • The LDAP user’s alias will be updated to match the Unity Connection user’s alias.
   Explanation:
   Most likely, a failed objects report will be created that contains information about the update attempt. The Bulk Administration Tool (BAT) can be used to add, update, or remove users from Cisco Unified Communications platforms. When you integrate an existing Cisco Unity Connection account with an existing Lightweight Directory Access Protocol (LDAP) user, you must first export the LDAP user information to a comma-separated values (CSV) file. Next, you must import the CSV file into Unity Connection by using the BAT. The LDAP user ID field that is specified in the CSV is what the BAT will use to search the Unity Connection database. If the BAT finds no matching Unity Connection user alias, the action will fail and a failed objects report will be created.

   If you want to integrate an existing Cisco Unity Connection user with an existing LDAP user, you should first ensure that the Alias field in Unity Connection matches the user ID in the LDAP directory. If the values do not match, you should modify the user’s Alias field in Cisco Unity Connection.
   No new Unity Connection account will be created from the LDAP account. In order to create a new Unity Connection user from a CSV import, you must specify that you want to add the users instead of update existing records.

   The Unity Connection user’s alias will not be updated to match the LDAP user’s alias. Because the Unity Connection user alias does not match the LDAP user ID, the BAT will not be able to locate a user to update.

   The LDAP user’s alias will not be updated to match the Unity Connection user’s alias. The BAT import process updates records in the Unity Connection user database, not the data from the CSV, which was exported from LDAP.

2. You have a Cisco CME router with CUE installed. You want to configure the router so that the MWI light will turn on when a user receives a voice mail message. All of the users have four-digit extensions.

   You issue the following commands on the router:

   ephone-dn 1
   mwi on

   Which of the following commands could you issue to complete the configuration? (Select the best answer.)

   • number *1
   • number ….
   • number 601
   • number 6001….
   • number 6001 ….
   • number *1 ….
   Explanation:
   You could issue the number 6001.…command to complete the configuration so that the message waiting indicator (MWI) light will turn on when a user receives a voice mail message. The number command is used to create the code that Cisco Unity Express (CUE) sends to Cisco Unified Communications Manager Express (CME) whenever a user receives a voice mail message.

   To configure MWI, you must create two ephonedns: one ephonedn to turn the MWI light on when the user receives a message, and one ephonedn to turn the MWI light off when the user retrieves all of his or her messages. When CUE receives a voice mail message for a user, CUE will send a code to the CME router to indicate that the user’s MWI light should be turned on. When the user retrieves the message, CUE will send another code to CME to indicate that the user’s MWI light should be turned off.

   The MWI codes can be any number of any length, as long as they are not the same as any existing extension numbers. To configure the MWI code that CUE will send to CME, you should issue the number command with the MWI code plus a number of periods equal to the number of digits in the users’ extensions. For example, if you want to create MWI code 6001 and your system is configured to use four-digit extensions, you should issue the number 6001…. command in ephonedn configuration mode.

   Finally, the ephonedn that is configured with the MWI code must also be configured with the mwi on or the mwi off command, depending on whether the ephonedn should be responsible for turning the MWI light on or off, respectively. For example, the following command set configures ephonedn 1 to turn on the MWI light for any fourdigit extension that is prefaced by the MWI code 6001: ephonedn 1
   number 6001….
   mwi on

   Similarly, the following command set configures ephonedn 2 to turn off the MWI light for any fourdigit extension that is prefaced by the MWI code 6002:
   ephonedn 2
   number 6002….
   mwi off

   After CME receives a dialed string from CUE that contains the MWI code, CME will match the dialed digits to the ephonedn number pattern, strip off the explicitly matched MWI code, and change the state of the MWI light for the extension that matches the remaining forwarded digits. For example, in this scenario, CUE will send the digit string 60017777 to CME when the user at extension 7777 receives a voice mail message? CME will then strip off MWI code 6001 and turn on the MWI light for extension 7777.

   You should not issue the number *1 command to complete the configuration. Although the number *1 command would configure ephonedn 1 to use the digits *1 to turn on the MWI light for an IP phone, the command does not define an extension number pattern. Therefore, the MWI light would not turn on for any phone extension.

   You should not issue the number 601 command to complete the configuration. Although the number 601 command would configure ephonedn 1 to use the digits 601 to turn on the MWI light for an IP phone, the command does not define an extension number pattern. Therefore, the MWI light would not turn on for any phone extension.

   You should not issue the number …. command to complete the configuration. Although the number ….command would configure ephonedn 1 with a fourdigit extension number pattern, no MWI code would be explicitly defined. Therefore, dialing any fourdigit extension would cause CUE to simply dial the user’s extension? it would not turn on the MWI light on a user’s phone.
   You should issue neither the number *1 …. nor the number 6001 …. command to complete the
   configuration. Unless you separate the extension numbers by using the secondary keyword, you cannot issue the number command with more than one extension number parameter? the MWI code and the extension number pattern should be typed together without a space between them.

3. Which of the following can a user template automatically configure for a new user in Unity Connection? (Select 2 choices.)

   • a user alias
   • an extension number
   • an administrator role
   • the message volume
   • an SMTP address
   Explanation:
   A user template can automatically configure an administrator role and the message volume for a new user in Cisco Unity Connection. Unity Connection user templates enable an administrator to automatically populate certain settings of a new user account based on the settings of a user template. The template settings are divided into two parts: basic settings and additional parameters. Administrator roles and the volume of message playback can be configured as additional parameters.

   Templates are created in Unity Connection by clicking Templates > User Templates and then clicking Add New. Next, you should choose an existing template to use as the base for the new template. The template you choose as the base template can be one of the Unity Connection default templates, such as the Voice-mail User Template or the Administrator Template, or a custom template that you have previously configured. The new template will inherit all the settings of the base template except for settings that are unique to each template, such as the template alias and display name.

   To configure a Unity Connection user template with an administrator role, you should select the administrator role you want to add from the list of options available on the Edit > Roles page of the template creation window. The following are the eight unique administrator levels that can be applied to a user template:

   -Audio Text Administrator
   -Greeting Administrator
   -Help Desk Administrator
   -Remote Administrator
   -Mailbox Access Delegate Account
   -Systems Administrator
   -Technician
   -User Administrator

   The administrative abilities of the user accounts that are created from the new user template depend on the role you select in the template’s configuration. If you do not select an administrator role when creating a new template, the user accounts that are created from the template will inherit the same abilities as those granted by the base template. For example, if you base the new template on Unity Connection’s predefined Administrator Template, the template you create will inherit the Systems Administrator role from the predefined template. If the base template does not have an administrator role, new users created from the template will be created as regular, non administrator end users.

   You can edit the message playback settings of the user template by clicking Edit > Playback Message Settings in the template creation window. In the Playback Settings area of the window, you can change the Message Volume dropdown field to Low, Medium, or High. The configuration of the Message Volume field determines the volume of voice mails and recorded instructions for faxing when the user plays those messages over the phone. You can also configure several other message playback settings in the Playback Message Settings window, such as the number of recorded messages a user is allowed to save, the speed at which the messages play, and what information about a new message is announced to the user when that message is played for the first time.

   A user template cannot automatically configure a user alias, an extension number, and a Simple Message Transfer Protocol (SMTP) address. The Alias field stores a user name, or alias, for the new user. The Extension field stores the telephone extension number that callers must dial to call the new user. The Alias field and the Extension field are required fields when you manually create a new user, even if the rest of that user’s information is based on a user template. In addition, the Unity Connection administrative graphical user interface (GUI), not a user template, can automatically configure the SMTP Address field. The SMTP addresses that are created by the GUI are based on the user alias you assign in the Alias field, unless the Alias field contains characters that are not ASCII characters.

4. Two VoIP users and a mobile phone caller are participating in an active UCM call. No analog gateway is present on the network. During the call, one of the UCM subscriber servers fails.

   Which of the following will occur? (Select the best answer.)

   • The mobile phone caller will be disconnected.
   • All three users will be disconnected.
   • All three users will remain active on the call.
   • The two VoIP users will be disconnected.
   Explanation:
   The mobile phone caller will be disconnected because mobile phones, which place calls through the public switched telephone network (PSTN), are not supported by the Cisco Unified Communications Manager (UCM) call preservation feature. The call preservation feature enables some Voice over IP (VoIP) devices to continue active sessions even if UCM fails or communication between UCM and the device is interrupted.

   When a UCM server fails, other UCM servers and supported devices in a cluster can detect the failure. UCM is then able to ensure that active calls remain active until either the users hang up or media stops streaming between the devices. Similarly, if UCM does not fail but loses connectivity to a device that is involved in an active call, both UCM servers and connected devices will detect the failure. The active call will remain active until the users end the call or media stops streaming between the devices.

   When a supported device other than UCM fails, that device will no longer stream media. Thus the device is no longer able to participate in its active call. However, UCM will detect this failure and any other devices that might have been active on the same call will remain connected and active.

5. Which of the following commands is used to view whether a voice gateway is registered to an H.323 gatekeeper? (Select the best answer.)

   • debug isdn q921
   • debug isdn q931
   • show ccmmanager
   • show gateway
   Explanation:
   The show gateway command is used to view whether a voice gateway is registered to an
   H.323 gatekeeper. A gatekeeper is an optional H.323 Voice over IP (VoIP) network devicethat provides access control, bandwidth management, and other services to H.323 devices on a VoIP network. Although a gatekeeper is not a required component of an H.323 VoIP network, all H.323 endpoints must register with a gatekeeper if a gatekeeper exists on the network.

   The show gateway command takes no parameters and should be issued in privileged EXEC mode on an H.323 voice gateway. For example, if you were to issue the show gateway command on an H.323 gateway named gw1 that is not registered with an H.323 gatekeeper, you would see the following output:

   Gateway gw1 is not registered to any gatekeeper

   By contrast, if you were to issue the show gateway command on an H.323 gateway named gw1 that is registered to an H.323 gatekeeper named gatekeeper1, you would see the following output:

   Gateway gw1 is registered to Gatekeeper gatekeeper1

   The show ccmmanager command is not used to view whether a voice gateway is registered to an H.323 gatekeeper. The show ccmmanager command displays a list of Cisco Unified Communications Manager (UCM) servers, the registration status of the servers, and the availability of the servers. For example, you can issue the show ccmmanager fallbackmgcp command to display the status of a Media Gateway Control Protocol (MGCP) gateway’s fallback feature. In addition, you can issue the show ccmmanager config-download command to display the configuration download status of both MGCP gateways and Skinny Client Control Protocol (SCCP) gateways.

   The debug isdn q921 command is not used to view whether a voice gateway is registered to an H.323 gatekeeper. The debug isdn q921 command displays Layer 2 access information about the D channel on Integrated Services Digital Network (ISDN) connections. The output of the debug isdn q921 command is based on International Telecommunication Union -Telecommunication Standardization Sector (ITUT) Recommendation Q.921, which describes ISDN user network interfaces. The debug isdn q921 command displays Layer 2 information related to transmissions and identity requests. The debug isdn q921 command displays information from only the D channel because the D channel contains the peertopeer messages between the router and the called or calling side of the connection.

   The debug isdn q931 command is not used to view whether a voice gateway is registered to an H.323 gatekeeper. The debug isdn q931 command displays information about call setup and call teardown on Layer 3 ISDN connections. The output of the debug isdn q931 command is based on the ITUT Recommendation Q.931, which describes ISDN switching and signaling. The debug isdn q931 command displays information from only the D channel because the D channel contains the peertopeer messages between the router and the called or calling side of the connection.

6. A user wants to ensure that callers from the voice VLAN and callers from the PSTN are directed to a voice mailbox if the line associated with the user’s dn is in use.

   Which of the following settings in the Call Forward and Pickup Settings section of the UCM Administration Directory Number Configuration page would enable this behavior? (Select 2 choices.)

   • Forward All
   • Forward Busy External
   • Forward Busy Internal
   • Forward No Answer External
   • Forward No Answer Internal
   • Forward Unregistered External
   Explanation:
   Of the available choices, the Forward Busy External setting would forward public switched telephone network (PSTN) callers to a voice mailbox if the line associated with the user’s directory number (dn) is in use. Similarly, the Forward Busy Internal setting would forward callers from the voice virtual LAN (VLAN) to a voice mailbox if the line associated with the user’s dn is in use. The Cisco Unified Communications Manager (UCM) Administration Directory Number Configuration page enables a UCM administrator to configure several settings related to dns, including the following: call forwarding, call pickup, call waiting, line display text, ring settings, and voice mailboxes.

   The Forward Unregistered External setting in the Call Forward and Pickup Setting ssection of the UCM Administration Directory Number Configuration page would direct a caller from the PSTN to voice mail if that caller attempted to connect to an extension that does not exist on your company’s Voice over IP (VoIP) network. In contrast to the Forward Unregistered External setting, the Forward Unregistered Internal setting would forward internal callers to a specific voice mailbox if the internal caller dialed a nonexistent dn.

   The Forward All setting forwards all callers, internal or external, to a specific voice mailbox. This is the same behavior as the CFwdAll softkey that appears on a Cisco IP phone. However, an administrator can configure this behavior for a user by accessing the Directory Number Configuration page if the user for some reason does not have access to the CFwdAll softkey.

   The Forward No Answer External setting forwards any calls from the PSTN that go unanswered by the user. Similarly, the Forward No Answer Internal setting forwards any internal calls that go unanswered by the user.

7. Which of the following can you display by clicking System Reports > Traffic > Summary by Phone Number in the UCM 8.0 CAR GUI? (Select the best answer.)

   • the current number of billing errors
   • the call volume for a given period of time
   • the QoS rating information for inbound calls
   • the top number of users by maximum length of calls
   Explanation:
   You can display information about call volume for a given period of time by using the System Reports > Traffic > Summary by Phone Number report in the Cisco Unified Communications Manager (UCM) Call Detail Records (CDR) Analysis and Reporting (CAR) graphical user interface (GUI). This report enables a CAR administrator to choose a range of time and IP phone extension numbers from which to view call volume information, thereby enabling an administrator to view what extensions were in use at a specific time.

   You can view information about the top number of users by maximum length of calls by using the User Reports menu. The By Duration report can be accessed by clicking User Reports > Top N in the UCM CAR GUI. This report enables a CAR administrator to view users who have made the longest calls over a given period of time, starting with the user who placed the longest call.

   You can view information about the current number of billing errors by using the System Reports > CDR Error report in the UCM CAR GUI. This report enables a CAR administrator to view the number of errors that occurred when CDR data was loaded into the reporting system.

   You can view Quality of Service (QoS) rating information for inbound calls by using the System Reports > QoS > Detail report in the UCM CAR GUI. The Detail report enables a CAR administrator to choose a UCM network and a period of time for which to view QoS ratings for both inbound and outbound calls. The Detail report can be used to monitor QoS at a user level.

8. Which of the following is most likely to cause dropped packets on a VoIP network? (Select the best answer.)

   • congestion
   • jitter
   • endtoend delay
   • variation in delay
   Explanation:
   Of the available choices, congestion is most likely to cause dropped packets on a Voice over IP (VoIP) network. Dropped packets can cause clips, or breaks, in the audio stream. However, voice traffic is more tolerant of dropped packets than of delayed packets, because a small amount of packet loss is not noticeable to the human ear. Packet loss can be mitigated by implementing Quality of Service (QoS) and congestion avoidance mechanisms, increasing bandwidth, and increasing buffer space. In addition, some codecs can correct small amounts of packet loss.

   Jitter is another term for a variation in delay, which causes delayed packets, not dropped packets. Jitter can cause packets to arrive out of sequence or at a different rate than they were sent. VoIP traffic is heavily affected by jitter because voice traffic is time sensitive and requires that the destination host receive the voice traffic in the order, and at the same rate, it was sent. When jitter is present, the end user might experience choppiness in the audio connection. A dejitter buffer at the destination is used to collect packets, sort them into the proper sequence based on Realtime Transport Protocol (RTP) time stamps, and release them to the voice application. Although a dejitter buffer can decrease jitter, it can increase delay as packets sit in the buffer. Jitter can be mitigated by increasing bandwidth, using QoS mechanisms to prioritize time-sensitive traffic, using Compressed RTP (cRTP) to compress headers, and using Stacker and Predictor to compress payloads.

   Endtoend delay causes delayed packets, not dropped packets. Endtoend delay is the sum of the processing, queuing, serialization, and propagation delays in the traffic path between the source of the packet and the destination of the packet. Therefore, the total network delay between the source of the packet and its destination is considered endtoend delay. Endtoend delay can be mitigated by QoS mechanisms.

9. You want to configure Cisco Unified Serviceability to log voice application activity at a specific logging level.

   Which of the following should you do? (Select the best answer.)

   • Click Trace > Configuration.
   • Click Alarm > Configuration.
   • Click Tools > Service Activation.
   • Click Tools > Serviceability Reports Archive.
   Explanation:
   You should click Trace > Configuration if you want to configure Cisco Unified Serviceability to log voice application activity at a specific logging level. The Trace menu can be used to configure parameters for voice application tracing tools that can be used in troubleshooting efforts. The tools available through the Trace menu typically send log output to a given destination for later manual review. Among the parameters you can configure in Trace > Configuration is the level of logging information that you want to see for a given service.

   You should not click Tools > Serviceability Reports Archive if you want to configure Cisco Unified Serviceability to log voice application activity at a specific logging level. You can access the Service Statistics Report by navigating to Tools > Serviceability Reports Archive in Cisco Unified Serviceability. The Cisco Unified Serviceability Reports Archive contains all of the following types of statistical reports:

   -Device Statistics Report
   -Server Statistics Report
   -Service Statistics Report
   -Call Activities Report
   -Alert Summary Report
   -Performance Protection Report

   You should not click Alarm > Configuration if you want to configure Cisco Unified Serviceability to log voice application activity at a specific logging level. The Cisco Unified Serviceability Alarm menu helps identify problems that exist with the Cisco Unified Communications system by using Simple Network Management Protocol (SNMP) and Syslog to generate data that can be used as part of the troubleshooting process. The notification process is automatic. Cisco recommends that administrators not modify the SNMP Trap and Catalog information associated with alarms.

   You should not click Tools > Service Activation if you want to configure Cisco Unified Serviceability to log voice application activity at a specific logging level. The Service Activation option under the Tools menu enables you to select individual services to activate or select all services at once. After you have selected the services you want to enable, you should click the Save button to activate those services.

10. Which of the following commands is not issued in telephonyservice configuration mode on a CME router? (Select the best answer.)

    • dn-webedit
    • max-dn
    • max-ephones
    • mac-address
    Explanation:
    The macaddress command is issued in ephone configuration mode, not telephony-service configuration mode, on a Cisco Unified Communications Manager Express (CME) router. The macaddress macaddress command associates a specific ephone number with the IP phone that has the specified Media Access Control (MAC) address. An ephone is a Cisco IOS representation of a physical IP phone device. However, you cannot configure ephones on a CME device until basic telephony service has been configured. Issuing the telephony-service command puts the router into telephonyservice configuration mode, where you can issue commands that configure telephony settings on the router.

    The dnwebedit command should be issued in telephony-service configuration mode. The dnwebedit command enables the addition of ephone directory numbers (dns), or extensions, by using the browser-based graphical user interface (GUI). By default, you cannot add an extension to CME by using the browserbased GUI.

    The maxdn command should be issued in telephonyservice configuration mode. The maxdn command configures the maximum number of extensions that you can configure on a CME device. The maximum value of the maxdn parameter varies based on the router model, the IOS version, and the amount of memory. If you do not issue the maxdn command, you will not be able to configure any ephonedns on the router.

    The maxephones command should be issued in telephonyservice configuration mode. You should issue the maxephones command to set the maximum number of phones, not the maximum number of extensions, that you can configure on a router. Like the maxdncommand, the maxephones command must be issued in telephonyservice configuration mode. The maximum value of the maxephones parameter varies based on the router model and the IOS version. If you do not issue the maxephones command, you will not be able to configure any ephones on the router.

11. You issue the show running-config command on Router1 and Router2 and receive the following partial output on both routers:
    Dial-peer voice 7 voip
    destinationpattern …….
    session target ipv4:10.11.12.13

    

    How will VoIP dial peer 7 be used when Phone1 calls Phone2? (Select the best answer.)

    • Router1 will use it as an inbound dial peer.
    • Router1 will use it as an outbound dial peer.
    • Router2 will use it as an inbound dial peer.
    • Router2 will use it as an outbound dial peer.
    Explanation:
    Router1 will use Voice over IP (VoIP) dial peer 7 as an outbound dial peer. A dial peer defines a logical route to a telephony endpoint. A voice gateway will match the call information with a dial peer configured on the router and create a corresponding call leg. A call leg is a logical inbound or outbound connection for a voice gateway. Dial peers that match inbound calls create inbound call legs, and dial peers that match outbound calls create outbound call legs.

    A dial peer is a plain old telephone service (POTS) dial peer if the call comes from, or is destined to, an analog phone, the public switched telephone network (PSTN), or an analog public branch exchange (PBX). A dial peer is a VoIP dial peer if the call comes from, or is destined to, an IP phone or another voice gateway across an IP WAN.
    In this scenario, Router1 is the originating voice gateway and Router2 is the terminating voice gateway. When Router 1 receives a call from Phone1, Router1 attempts to match the inbound call information to a POTS dial peer because Phone1 is an analog phone connected to a foreign exchange station (FXS) port on Router1. If no matching dial peer is found, Router1 will use the default dial peer. Dial peer 7 does not match, because it is a VoIP dial peer, not a POTS dial peer. Therefore, Router1 will not use dial peer 7 as an inbound dial peer.
    Router1 must then send the call over an IP WAN network; therefore, Router1 attempts to match the outbound call information to a VoIP dial peer. If no matching dial peer is found, Router1 will drop the call. Dial peer 7 is a VoIP dial peer, and the destinationpattern ……. command matches the destination number 5551234. Therefore, Router1 will use dial peer 7 as an outbound dial peer.
    Router2 receives the inbound call from Router1 and attempts to match the inbound call information to a VoIP dial peer. If no matching dial peer is found, Router2 will use the default dial peer. Dial peer 7 does not match, because the destinationpattern …….command does not match the originating extension 4273. Therefore, Router2 will not use dial peer 7 as an inbound dial peer.
    Finally, Router2 must send the call over the PSTN? therefore, Router2 attempts to match the outbound call information to a POTS dial peer and sends the call to Phone2 over the PSTN. If no matching dial peer is found, Router2 will drop the call. Dial peer 7 does not match, because it is a VoIP dial peer, not a POTS dial peer. Therefore, Router2 will not use dial peer 7 as an outbound dial peer.
    A voice gateway router will perform the following evaluations when it receives an inbound call:

    1. The router will attempt to match the destination Dialed Number Identification Service (DNIS) to an incoming called-number DNIS command.
    2. The router will attempt to match the source Automatic Number Identfication (ANI) to an answer-address ANI command.
    3. The router will attempt to match the source ANI to a destination-pattern string command.
    4. The router will attempt to match the incoming call’s voice port to a port command.
    5. If no match is found, the router will use the default dial peer.

    Once a dial peer match is found, the router will immediately create an inbound call leg without proceeding to the next step. If multiple matches are found for a step, the router will select the longest explicit match. The default dial peer will only be used if no match is found. You cannot configure any of the settings for the default dial peer.
    To create an outbound call leg, a voice gateway router will perform the following evaluations:

    1. The router will attempt to match the destination DNIS to a destination-pattern string command.
    2. If the dial peer is a POTS dial peer, the router will forward the call to the port indicated by the corresponding port port command.
    3. If the dial peer is a VoIP dial peer, the router will forward the call to the IP address indicated by the corresponding session target ipv4: ip-address command.
    4. If no match is found, the call will be dropped.

    As with the inbound call leg, the router will immediately create an outbound call leg as soon as a match is found. If multiple matches are found for a step, the router will select the longest explicit match. The call will be dropped only if no match is found.

12. Users report that voice mail recordings are not loud enough.

    Which of the following is least likely to aid in troubleshooting the problem? (Select the best answer.)

    • adjusting AGC decibels
    • disabling AGC
    • obtaining a sniffer capture at the closest point to Unity Connection
    • verifying the advertised codec settings
    Explanation:
    Of the available choices, obtaining a sniffer capture at the closest point to Cisco Unity Connection is least likely to aid in troubleshooting the problem. However, you might obtain a sniffer capture of an audio stream before Unity Connection receives it as a step in troubleshooting garbled voice mail recordings.

    Disabling Automatic Gain Control (AGC) or adjusting AGC decibels might aid in troubleshooting the problem. AGC enables Unity Connection to automatically adjust the audio volume of calls. You can adjust or disable AGC if users report that the audio volume of voice mail recordings is always too loud or always too soft.

    Verifying the advertised codec settings might aid in troubleshooting the problem, especially if users are reporting no sound at all. An IP phone’s codec must match the voice gateway’s codec to enable a user to successfully place a call. In Unity Connection Administration, you can verify the codec’s settings by examining the list of codecs in Telephony Integrations> Port Group.

13. Which of the following is not a method of deleting an unassigned dn from a UCM database? (Select the best answer.)

    • Call Routing > Route Plan Report > Delete
    • Call Routing > Route Plan Report > Delete Selected
    • Call Routing > Route Plan Report > Delete All Found Items
    • Call Routing > Route Plan Report > Delete Unassigned DN
    • Bulk Administration > Phones > Delete Phones > Delete Unassigned DN
    Explanation:
    You cannot use Call Routing > Route Plan Report > Delete Unassigned DN to delete an unassigned directory number (dn) from a Cisco Unified Communications Manager (UCM) database. An unassigned dn is a dn that is not associated with a specific device, such as an IP phone, but that can still be used to forward calls to voice mail or to another dn that is associated with a device. For UCM to load and use an unassigned dn, the Active check box must be selected for the dn. The Active check box is only displayed for unassigned dns.
    Problems with unassigned dns can cause an IP phone that is attempting to autoregister with UCM to display the following error:

    Registration Rejected: Error DBConfig

    Therefore, you should remove unassigned dns from the auto registration configuration if this error occurs.

    There is no Delete Unassigned DN button in the UCM Call Routing > Route Plan Report window.
    You can use Call Routing > Route Plan Report > Delete, Call Routing > Route Plan Report > Delete Selected, or Call Routing > Route Plan Report > Delete All Found Items to delete an unassigned dn from a UCM database. If you click a dn that is displayed in the Route Plan Report window and then click the Delete button, only that dn will be deleted from the UCM database. If you select the check boxes beside a list of dns in the Route Plan Report window and then click the Delete Selected button, all the dns with selected check boxes will be deleted from the UCM database. If you click Delete All Found Items in the Route Plan Report window, every unassigned dn will be deleted from the UCM database, even if the check box beside a dn is not selected.

    You can use Bulk Administration > Phones > Delete Phones > Delete Unassigned DN to delete an unassigned dn from a UCM database. The UCM Bulk Administration > Phones > Delete Phones > Delete Unassigned DN window automatically searches for and displays a list of unassigned dns in the UCM database. Once the list of unassigned dns is complete, you should select the Run Immediately radio button and then click Submitto immediately delete the unassigned dns from the UCM database.

14. Which of the following Cisco Unity Connection services can be deactivated by using Cisco Unity Connection Serviceability Control Center? (Select the best answer.)

    • Connection DB
    • Connection Server Role Manager
    • Connection Serviceability
    • Connection Message Transfer Agent
    Explanation:
    Of the available choices, only Connection Message Transfer Agent can be deactivated by using Cisco Unity Connection Serviceability Control Center. However, deactivating Connection Message Transfer Agent will cause voice mail messages to not be delivered. The Connection Message Transfer Agent is a critical Unity Connection service that enables voice mail messages to be delivered to the message store. Although Unity Connection will operate without this service, voice mail messages cannot be delivered without it.

    Connection DB cannot be deactivated by using Cisco Unity Connection Serviceability Control Center. Instead, this service must be deactivated by using the command line interface (CLI). Connection DB is a status only service that starts the Unity Connection database.

    Connection Server Role Manager cannot be deactivated by using Cisco Unity Connection Serviceability Control Center. Instead, this service must be deactivated by using the CLI. Connection Server Role Manager is a status only service that enables server status when a Cisco Unity Connection cluster is configured.

    Connection Serviceability cannot be deactivated by using Cisco Unity Connection Serviceability Control Center. Instead, this service must be deactivated by using the CLI. Cisco Serviceability is a status only service that starts the Cisco Unity Connection Serviceability Administration interface.

15. You want to add 90 IP phones to a UCM network.

    You enable autoregistration in UCM and configure the Starting Directory Number field to 5000.

    You also configure the Ending Directory Number field.

    You connect the IP phones to the network. The secondtolast IP phone that you connect displays a dn of 5088. However, the last IP phone that you connect never registers with UCM.

    Which of the following is most likely the problem? (Select the best answer.)

    • A rogue IP phone exhausted the dn pool.
    • You configured the same dn for both the start and the end of the dn pool.
    • You did not configure the last IP phone’s MAC address in UCM.
    • You did not clear the Auto registration Disabled on this Cisco Unified Communications Manager check box in UCM.
    Explanation:
    Of the choices provided, the problem is most likely that a rogue IP phone has exhausted the directory number (dn) pool. Autoregistration enables Cisco Unified Communications Manager (UCM) to automatically add new IP phones to the UCM database as the IP phones are connected to the network. When a new IP phone is connected to the network, UCM will automatically assign an unused dn to the IP phone from a specified pool of dn numbers. The dn pool is assigned in order from the lowest available dn through the highest dn. Once the pool of dn numbers is exhausted, new IP phones will not be able to auto register with UCM.

    Auto registration is a security risk because rogue devices can be connected to the network and registered with UCM by using auto registration. In addition, you could accidentally register a valid IP phone with a dn from the wrong dn pool if you leave auto registration enabled after you have completed an auto registration process. Therefore, Cisco recommends that you enable auto registration only for short periods of time, such as when you need to add fewer than 100 IP phones to the network. In this scenario, you have configured auto registration so that UCM can allocate dns to 90 new IP phones. Because the last IP phone you connect cannot auto register, it is likely that a rogue IP phone auto registered during the period of time that you enabled auto registration to add new IP phones. You could have also encountered this problem if the range of dns you configured was not large enough to accommodate the 90 IP phones that you needed to add to UCM.

    A missing IP phone Media Access Control (MAC) address in UCM cannot cause an IP phone to fail to auto-register. By design, the UCM autoregistration process does not require an administrator to configure IP phone MAC addresses in UCM. However, if you manually provisioned IP phones or if you added new IP phones by using the Bulk Administration Tool (BAT), you would need to configure MAC addresses for the IP phones in UCM. When you are manually provisioning an IP phone in UCM, you must fill in the MAC Address field, the Device Pool field, the Phone Button Template field, and the Device Security Profilefield.

    It is not likely that you configured the same dn for both the start and the end of the dn pool. When you enable autoregistration in UCM, you must configure the Starting Directory Number field with a value that is lower than the value of the Ending Directory Number field. If you do not properly configure the range of dns, UCM automatically disables autoregistration. In this scenario, 89 legitimate new IP phones have already autoregistered with UCM before the last IP phone fails. Therefore, it is not possible that autoregistration was disabled because of an invalid dn range.
    It is likely that the Autoregistration Disabled on this Cisco Unified Communications Manager check box is still selected in UCM. If you provide a valid range of dns in the Starting Directory Number field and the Ending Directory Number field, UCM will automatically clear the Autoregistration Disabled on this Cisco Unified Communications Managercheck box, thus enabling autoregistration. In addition, in this scenario, 89 legitimate new IP phones have already autoregistered with UCM before the last IP phone fails. Therefore, it is not likely that the Autoregistration Disabled on this Cisco Unified Communications Manager check box is still selected in UCM.

16. Which of the following does the auto assign command configure a router to do? (Select the best answer.)

    • automatically assign an ephone to an IP phone
    • automatically assign an ephonedn to an ephone
    • automatically assign an IP address to an IP phone
    • automatically assign an IP phone to a hunt group
    Explanation:
    The auto assign command configures a router to automatically assign an ephonedn to an ephone. When an IP phone registers with a router that is configured with the auto assigncommand, the router will associate button 1 on the IP phone’s ephone with an existing, unused ephonedn. If no ephonedn is available, the phone will register but none of the phone’s buttons will be associated with ephonedn extensions.

    The autoregephone command, not the auto assign command, configures a router to automatically assign an ephone to an IP phone. When an IP phone registers with a router that is configured with the autoreg-ephone command, the router will associate the Media Access Control (MAC) address of the IP phone with the first unassigned ephone on the router. If all the ephones on the router are associated with IP phones, the router will create a new ephone, provided that the number of configured ephones does not exceed the value of the maxephone command. You can also manually assign an IP phone to an ephone by issuing the macaddress macaddress command in ephone configuration mode, where macaddress is the MAC address of the IP phone you want to assign to the ephone.

    The auto assign command does not configure a router to automatically assign an IP address to an IP phone. A router that is configured as a Dynamic Host Configuration Protocol (DHCP) server automatically assigns an IP address to an IP phone. The following partial output from the show runningconfig command displays a DHCP server that is configured to distribute IP addressing information to IP phones:

    ip dhcp excluded-address 192.168.14.1 192.168.14.9
    ip dhcp pool IPPhones
    network 192.168.14.0 255.255.255.0
    default-router 192.168.14.1
    option 150 ip 192.168.14.1

    The auto assign command does not configure a router to automatically assign an IP phone to a hunt group. IP phones are associated with ephones, ephones are associated with ephonedns, and ephonedns are manually assigned to hunt groups.

17. You administer a UCM environment that contains only Cisco IP phones with factoryinstalled firmware.

    Which of the following messages will not be sent or received by the IP phones? (Select 2 choices.)

    • 486 Busy here
    • Connected
    • INVITE
    • Off Hook
    • On Hook
    Explanation:
    Cisco IP phones with factory firmware will not send or receive 486 Busy here messages. In addition, these IP phones will not send INVITE messages. Both the 486 Busy here message and the INVITE message are call signaling messages that are sent and received by Session Initiation Protocol (SIP) endpoints. By default, Cisco IP phones use Skinny Control Client Protocol (SCCP), which is a Ciscoproprietary client/ server call signaling protocol. Although Cisco IP phones can be configured to use SIP when installed with a firmware replacement, the Cisco IP phones in this scenario are using factory installed firmware.

    SIP is an Internet Engineering Task Force (IETF)standard call signaling protocol that is supported by a wide variety of IP telephony vendors. A call signaling protocol is responsible for the setup, maintenance, and teardown of a voice call. For example, call signaling protocols can detect and report when a phone is off-hook. SIP uses a textbased signaling method, which is easier to understand and troubleshoot than the binary method used by other protocols, such as SCCP and H.323.

    SIP is most commonly used by Internet telephony service providers (ITSPs). Therefore, many nonCisco IP phones and video phones are SIP phones. Because Cisco Unified Communications Manager (UCM) supports SIP, it is possible for nonCisco IP phones to be registered with UCM and function on a Voice over IP (VoIP) network that otherwise contains SCCP endpoints.

    Connected, Off Hook, and On Hook are all call signaling state messages that are sent to SCCP endpoints in a UCM environment. SCCP typically represents each of these call states with a callstate identifier number. For example, a callstate of 1 indicates a call state of Off Hook. A callstate of 2 indicates a call state of On Hook.

18. You are upgrading the firmware on all Cisco IP Phone 7961s that are connected to your company’s network. All IP phones are configured to their default upgrade settings.

    Which of the following upgrade methods will require the most administrative overhead? (Select 2 choices.)

    • individual IP phone upgrades
    • load server download
    • peer firmware sharing
    • traditional TFTP server download
    Explanation:
    Of the available choices, individual IP phone upgrades and peer firmware sharing will require the most administrative overhead if the IP phones are configured to their default upgrade settings. By default, Cisco IP phones are configured to use the traditional Trivial File Transfer Protocol (TFTP) upgrade method. In order to enable peer firmware sharing, the administrator must enable it on each IP phone. Similarly, individual IP phone upgrades require the administrator to address each IP phone’s upgrade individually.

    When using the traditional TFTP server download method, each IP phone independently downloads the new image from the TFTP server in an “every man for himself” style strategy. When firmware images were small, this strategy was acceptable even when the IP phones were on a network at a separate location from Cisco Unified Communications Manager (UCM). Over time, IP phone firmware sizes have increased, which could cause slow upgrades over WAN links. In addition, the traditional TFTP download method could create high CPU usage on the UCM TFTP server.

    You can also update the firmware on an individual IP phone by using the traditional TFTP method. First, you should make a note of the existing Phone Load Name value for the phone model that you want to upgrade by navigating to Device > Device Settings > Device Defaults in UCM Administrator. This is important because installing the new firmware image will automatically overwrite the value of the Phone Load Name field in Device > Device Settings > Device Defaults. You should then upload the new firmware to UCM by navigating to Software Upgrades > Install/Upgrade.

    After you upload the new firmware, specify the name of the new firmware in the Phone Load Name field for the specific IP phone you want to upgrade by using UCM Administration’s Device > Phone menu. Next, navigate to Device > Device Settings > Device Defaults and replace the new value of the Phone Load Name field with its original value. This will prevent other IP phones from downloading the new firmware after you restart the TFTP service.
    Finally, you should restart the TFTP service in Cisco Unified Serviceability. After the service restarts, the IP phone you edited in UCM Administration should download the new firmware, upgrade the firmware, and restart. Other IP phones might restart as well. However, those IP phones will not be upgraded.

    In contrast to the traditional TFTP server method, the load server download method enables the administrator of the LAN on which the IP phone operates to provide his or her own local TFTP server for firmware upgrades instead of relying on a remotely located default UCM TFTP server. This means that IP phones on remote networks will be able to download firmware updates in approximately the same amount of time it would take for an IP phone that is local to UCM. In addition, the TFTP load can be balanced among multiple TFTP servers at multiple sites. One disadvantage to the load server download method is that the local administrator is responsible for copying the firmware update to the TFTP server. Therefore, the TFTP upload and server configuration is subject to human error.

    When peer firmware sharing is implemented, only one Cisco IP phone at a location is responsible for downloading the new firmware. The firmware is then distributed to the other IP phones on the LAN in a parent-child hierarchy. The downloading phone distributes the firmware to its children. Those children then distribute the firmware to their children, and so on. No one parent in the hierarchy can have more than two children. Some disadvantages to the peer firmware sharing method are that the hierarchies are limited to their own subnets and are specific to phone model. In addition, peer firmware sharing must be enabled on each IP phone.

19. Which of the following commands correctly configures voice VLAN 10 on a trunk port? (Select the best answer.)

    • switchport trunk native vlan 10
    • switchport trunk allowed vlan 10
    • switchport voice vlan 10
    • switchport access vlan 10
    • None of the commands configures a voice VLAN on a trunk port.
    Explanation:
    None of the commands that are provided in this scenario correctly configures voice virtual LAN (VLAN) 10 on a trunk port, because voice VLANs are not supported on trunk ports on Cisco switches. You can configure voice VLANs on only Layer 2 access ports on a Cisco switch. Although you can issue the switchport voice vlan command on a trunk port, the voice VLAN will not be configured on the port unless you also issue the switchport mode access command.

    Creating voice VLANs on a switch enables the separation of voice traffic from data traffic on a network. If data and voice devices are configured to operate on the same VLAN, the voice traffic can experience quality problems, such as jitter or choppiness. To enable the voice VLAN feature on a Cisco switch, you should issue the switchport voice vlan {vlanid | dot1p | none | untagged} command in interface configuration mode. The dot1pkeyword configures voice traffic to be sent with a default 802.1p priority of 5 and to use VLAN 0 as the VLAN ID.
    The switchport voice vlan untagged command configures voice traffic to be untagged and sent over the native VLAN. Traffic that is passed over the native VLAN is sent untagged, which means that the packet is sent without 802.1Q encapsulation. When the switchport voice vlan untagged command is issued, both voice traffic and data traffic are transmitted over the native VLAN. You do not need to specify a voice VLAN when the switchport voice vlan untagged command is used.
    The switchport voice vlan vlanid command configures voice traffic to be tagged and sent over a user – specified voice VLAN. Voice traffic will be carried in 802.1Q frames and will be carried on a different VLAN than data traffic. For example, if you issue the switchport voice vlan 2 command, voice traffic will be tagged with 802.1Q information and sent over VLAN 2.
    Similar to the switchport voice vlan untagged command, the switchport voice vlan none command configures voice traffic to be untagged and sent over the same VLAN as data traffic, which is the native VLAN. When the none keyword is used, voice traffic does not use 802.1p priority tagging or Class of Service (CoS) and voice traffic is transmitted with data traffic.
    You can configure data VLANs and native VLANs on both trunk ports and access ports on a Cisco switch. To configure a data VLAN or the native VLAN on an access port, you should issue the switchport access vlan vlanid command, where vlanid is the ID of the data VLAN or the native VLAN you want to configure.
    To configure a trunk port with the native VLAN, you should issue the switchport trunk native vlan vlanid command, where vlanid is the ID of the native VLAN. In addition, trunk ports are by default configured to allow traffic from all data VLANs that are configured on the switch. You can issue the switchport trunk allowed vlan remove vlanidlist command to specifically remove a list of data VLANs from a trunk port. You can add a specific VLAN to a trunk port by issuing the switchport trunk allowed vlan add vlanidlist command, where vlanidlist is a list of the VLAN IDs you want to add.

20. You are the administrator of a VoIP network. Your supervisor reports that calls to both internal and external users are connecting; however, users are unable to hear any audio. No recent configuration changes have been made to the VoIP network.

    Which of the following is most likely the cause of the problem? (Select the best answer.)

    • CUE network module failure
    • Ethernet cable failure
    • router WIC failure
    • digital signal processor failure
    Explanation:
    A digital signal processor (DSP) failure is most likely the cause of the problem. A DSP failure is likely if you are experiencing any of the following problems:

    -Callers are unable to hear audio in one or both directions.
    -Call setup is failing.
    -Channels become stuck in the PARK state.
    -Error messages report DSP timeouts.

    When analog audio is received by a DSP, the DSP samples and quantizes the analog audio data, encodes the data into binary format, and optionally, compresses it to conserve bandwidth. For example, Cisco Integrated Services Routers (ISRs) with DSP resources are used to process calls between Cisco Unified Communications Manager (UCM) and the public switched telephone network (PSTN).

    A Cisco Unity Express (CUE) network module failure is not the cause of the problem. CUE is a voice mail and automated attendant hardware platform for small to medium-sized offices. It provides up to 250 mailboxes and up to 24 simultaneous voice sessions, depending on the CUE hardware platform. A CUE network module failure could result in a loss of voice mail and automated attendant features.

    An Ethernet cable failure is not the cause of the problem. An Ethernet cable failure could result in a failure to connect to the voice router or voice gateway, thereby resulting in disconnected or intermittently disconnected calls.

    A router WAN interface card (WIC) failure is not the cause of the problem. A router WIC failure could result in a loss of connectivity to an IP WAN, thereby resulting in disconnected calls or an inability to connect calls over the IP WAN.