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The Cisco Unified Real-time Monitoring Tool (RTMT) is a client-side application that enables an administrator to monitor devices on a Cisco Voice over IP (VoIP) network in real time by using Secure Hypertext Transfer Protocol (HTTPS). RTMT uses HTTPS to connect to VoIP devices and gather information, such as device status and performance statistics, in real time. The data that is gathered by RTMT can then be used to pinpoint problems on the VoIP network or to monitor performance thresholds. 
To access RTMT, you should first ensure that the Cisco RTMT Reporter Servlet and Cisco Serviceability Reporter services are running in the Cisco Unified Communications Manager (UCM) environment. Next, you should install the RTMT plugin on a workstation by clicking Application > Plugins in the UCM administrative graphical user interface (GUI). After you have installed the plugin, you should launch the Real-time Monitoring Tool application on the workstation, type the appropriate IP address and credential information for accessing the UCM server or cluster, select the Secure Connection check box, and then click OK. 
Cisco Unified Reporting is not a client-side application that enables an administrator to monitor devices on a Cisco VoIP network in real time by using HTTPS. Unified Reporting is a browser-based troubleshooting tool that uses HTTPS to access information that is provided by other reporting tools, such as RTMT and Cisco Unified Call Detail Records (CDR) Analysis and Reporting Tool (CAR). However, Unified Reporting does not provide access to feature activation tools and network service activation tools. You can access Unified Reporting by clicking Navigation > Cisco Unified Reporting from within the UCM administrative GUI or by using the HTTPS address https://ip-address:8443/cucreports/, where ip-address is the IP address of the UCM server or cluster. For example, after you have navigated to Cisco Unified Reporting, you could navigate to System Reports > Unified CM Data Summary > Generate Report to monitor system activities.
Cisco Unified Serviceability is not a client-side application that enables an administrator to monitor devices on a Cisco VoIP network in real time by using HTTPS. Unified Serviceability is a browser-based troubleshooting tool that uses HTTPS to access information that is provided by other reporting tools, such as RTMT and Cisco Unified CAR. In addition, Unified Serviceability provides access to several feature services that can be activated by using the Service Activation window, including database services, CDR services, and security services. You can access Unified Serviceability by clicking Navigation > Cisco Unified Serviceability from within the UCM administrative GUI, or by using the HTTPS address https://ip-address:8443/ccmservice/, where ip-address is the IP address of the UCM server or cluster.
CAR is not a client-side application that enables an administrator to monitor devices on a Cisco VoIP network in real time by using HTTPS. CAR generates CDR reports, Quality of Service (QoS) reports, traffic reports, and billing reports. In addition, CAR reports are not real-time reports. You can access CAR by clicking Tools > CDR Analysis and Reporting in Unified Serviceability if you are a system administrator or by using the HTTPS address https://ip-address:8443/car/Logon.jsp, where ip-address is the IP address of the UCM server or cluster, if you are a CAR administrator or user.
Reference:
Cisco: Understanding Cisco Unified RTMT: Understanding Cisco Unified Real-time Monitoring Tool

Quality of Service (QoS) and audio signal compression are functions that are often provided by an Internet telephony service provider (ITSP) but are not typically provided by the public switched telephone network (PSTN). ITSPs enable customers to use Voice over IP (VoIP) to make phone calls over the Internet. 
QoS is a VoIP technique that ensures call quality and integrity by mitigating delay and dropped packets, which can interrupt the flow of a VoIP call. Typical QoS techniques include buffer management and the use of multiple transmission queues to separate types of multimedia packets. Because voice traffic is sent in real time, quality is critical. 
Audio signal compression replaces consecutive repeating audio signals with code that instructs an endpoint to play one specific signal a given number of times. The bandwidth consumed by a call is reduced when compression is used. 
Call setup and teardown, call supervision, and call routing are all functions that are provided by ITSPs and the PSTN. Call setup involves the series of events between a phone going off-hook and establishing a connection; these events include dial-tone signals and ring signals. Call supervision involves the change in the state of a line or trunk port, such as line seizure, answer, or disconnect. Call routing involves selecting the path on which a call is transported from a source endpoint to a destination endpoint. On a VoIP network, voice gateways are responsible for call routing. 
Reference:
Packetizer: Understanding VoIP: How Does VoIP Work?
Internet Telephony: The Basics of Internet Telephony
Cisco: Voice Network and Signaling Control: Signaling System 7 U.S. PSTN Features

The problem is most likely that the IP phone at extension 5545 is using a search space that does not include the NY60xxPT partition. A partition is a logical grouping of Voice over IP (VoIP) route patterns and directory numbers (dns). A search space is an ordered list of partitions that a device is allowed to search for patterns that match a dialed number. A device that is not able to match a dialed number in any of the search spaces that are assigned to the device will generate a busy signal. For example, an administrator could segregate local and long distance route patterns into two partitions named LocalPT and LongDistancePT, respectively. The administrator could then assign devices that should not be able to make long distance calls to a search space named Local that includes only the LocalPT partition. Thus devices that have been assigned to only the Local search space could not make long distance calls. 
In the network shown above, the NY60xxPT partition contains an IP phone that has been assigned the dn 6045. When the user at the IP phone that has been assigned the extension 5545 attempts to dial 6045, the user receives a busy signal. Therefore, it is most likely that the IP phone that has been assigned extension 5545 has not been assigned a search space that includes the NY60xxPT partition. 
It is not likely a problem that the IP phone at 5545 is using a device search space. It is also not likely a problem that the IP phone at 5545 is using a line search space. A device search space is a search space that is assigned to a device itself. The information in a device search space will be searched no matter which line on a device is chosen for the outgoing call. A line search space, on the other hand, is a search space that is assigned to a single line on a device, not to the device itself. The information in a line search space will be searched when the user chooses the line to which the search space is assigned as the outgoing line for the call. 
If a device is configured with both a device search space and a line search space, the device will combine the two search spaces and search the information that is contained in the line search space first. For example, if you assign the Local search space to a device and the LongDistance search space to a line on the same device, the LongDistance search space will be searched first, even if the user dials a local extension. If the same route pattern or dn exists in both search spaces, the LongDistance search space will be used to break the tie. 
It is not likely that dn 6045 is listed in the <None> search space. Any VoIP endpoint can match the route pattern or dn that is contained within the <None> search space because every VoIP endpoint is assigned the <None> search space by default. The <None> search space contains the <None> partition. The <None> partition initially contains all the dns that are configured in the VoIP network. Therefore, you should move dns from the <None> partition to a custom partition to limit specific pattern matching to specific endpoints. The IP phone at 5545 would be able to call the IP phone at 6045 if 6045 were listed in the <None> search space. 
It is not likely that the IP phone at 5545 is using a search space that does not include the NY50xxPT partition, does not include the LA55xxPT partition, or does not include the LA65xxPT partition. Nothing in the scenario indicates that the IP phone at 5545 is receiving a busy signal when the user attempts to dial extensions that are included in those partitions. 
Reference:
Cisco: Partitions and Calling Search Spaces: Understanding Partitions and Calling Search Spaces

A Session Initiation Protocol (SIP) interface handles the exchange of availability information between Cisco Unified Communications Manager (UCM) and Cisco Unified Presence Server (CUPS). UCM and a CUPS server together are the primary components of a Cisco Presence deployment. A UCM SIP trunk interface must point to the CUPS server in order for availability information to be exchanged between the two systems. CUPS is also capable of sending SIP subscribe messages to UCM over the SIP trunk if UCM is configured as a Presence gateway. 
The Extensible Messaging and Presence Protocol (XMPP) interface does not handle the exchange of availability information between UCM and CUPS. However, the XMPP interface is used to handle the exchange of availability information between UCM and XMPP clients, such as instant messaging (IM) clients that are developed by third parties. 
The Lightweight Directory Access Protocol (LDAP) interface does not handle the exchange of availability information between UCM and CUPS. However, the LDAP interface is used to synchronize user information between UCM and CUPS in order to create a single sign-on (SSO) user experience or to perform contact searches. For example, a Cisco Unified Personal Communicator user can be authenticated to both the CUPS server and UCM by connecting directly to the CUPS server. LDAP is a directory protocol that is used by other servers, such as CUPS, to perform contact lookups. LDAP listens on Transmission Control Protocol (TCP) port 389 unencrypted or on port 636 over Secure Sockets Layer (SSL). Third-party XMPP clients can also use LDAP to search the database and add users as contacts. 
The Cisco Administrative Extensible Markup Language (AXL)/Simple Object Access Protocol (SOAP) interface does not handle the exchange of availability information between UCM and CUPS. However, the AXL/SOAP interface is used to handle database synchronization tasks from UCM to the CUPS database. For synchronization to start, the Sync Agent service must be started on the CUPS server. 
Reference:
Cisco: Cisco Unified Presence Features and Functions: Cisco Unified Presence Components

Of the available choices, you should instruct the user to press the settings button on the IP phone in order to enable the user to adjust the contrast on a Cisco IP Phone 7961. The Cisco IP Phone 7961 series includes a bank of buttons with iconography designed to represent the button's functions. For example, the settings button is represented by a selected check box. The Contrast function is a user preference because it can be adjusted on a per-user basis. Therefore, the user should press settings > User Preferences > Contrast to change the contrast on the IP phone. A typical bank of buttons for a Cisco IP Phone 7961 series appears in the following exhibit:

   
You should not instruct the user to press the services button. The services button, which is represented by a globe icon, is used to launch IP phone applications. The applications that are available from the services button are dependent on the Cisco Unified Communications deployment and user privilege levels. 
You should not instruct the user to press the directories button. The directories button, which is represented by an open book icon, is used to display lists of missed calls, received calls, placed calls, or local directory contacts. If configured, the directories button can also be used to access a custom Personal Speed Dial directory. 
You should not instruct the user to press the help button. The help button, which is represented by a question mark (?), is used to provide the end user with information about the specific features of the IP phone. You can press the help button twice while on a call on an IP phone to view statistical information about the call, such as the codec that is being used by the IP phone, the codec that is being used by the calling phone, and packet error information. 
Reference:
Cisco: Cisco Unified IP Phone 7961G

You should modify the Voice Mail Password Settings section of the Cisco Unity Connection user template to force new users to change their telephone user interface (TUI) passwords when they log in. Cisco Unity Connection users have two passwords: the voice mail system password that is issued by using the telephone keypad and the web application password that is issued by using Cisco Unity Connection’s web-based graphical user interface (GUI). To force new users to change their TUI passwords when they log in, you should click Templates > User Templates in Cisco Unity Connection and then select Voice Mail from the Choose Password dropdown menu. In the Voice Mail Password Settings section, select the User Must Change at Next Sign-In check box and click the Save button.
You should not modify the Location section of the user template. The Location section contains fields that enable an administrator to configure geographic information about Cisco Unity Connection users, such as address, language, and time zone settings. If your company has multiple locations in varying time zones, you could create individual user templates for each time zone so that you are not required to modify the time zone each time you add users to a unique location. 
You should not modify the Web Application Password Settings section of the user template. The Cisco Unity Connection web-based GUI enables access to the User Web Pages feature of a Cisco Unified Communications environment. The User Web Pages feature can be used to provide users with the ability to perform some customization of their Cisco IP phones or other Unified Communications settings. Because a user enters the web application password by using the GUI, not by using the TUI, you do not need to modify the Web Application Password Settings section in this scenario. To access the Web Application Password Settings section, you should click Templates > User Templates in Cisco Unity Connection and then select Web Application from the Choose Password drop-down menu. 
You should not modify the Phone section of the user template. The Phone section enables a Cisco Unity Connection administrator to apply dial plan settings, such as the partition and search space, as well as Class of Service (CoS) settings to users. A partition is a logical grouping of Voice over IP (VoIP) route patterns and directory numbers (dns). A search space is an ordered list of partitions that a device is allowed to search for patterns that match a dialed number. CoS settings enable an administrator to apply a specific set of privileges to Cisco Unity Connection users. 
You should not modify the Name section of the user template. The Name section contains the Alias field and the Display Name field, both of which are required fields when creating users in Cisco Unity Connection. An administrator can also configure the way names are displayed by choosing either the First Name, Then Last Name radio button or the Last Name, Then First Name radio button in the Name section. 
Reference:
CCNA Voice 640-461 Official Cert Guide, Chapter 13, Password Settings, pp. 359-360

Of the available choices, Voice over IP (VoIP) hopping is least likely to occur as a result of a failure to implement Quality of Service (QoS) on the LAN. VoIP hopping is a form of virtual LAN (VLAN) hopping; it is a data security risk associated with VoIP devices and data devices that are connected to the same physical port on a switch. Many IP phones contain switching technology that enables an administrator to daisy chain a workstation to the phone, causing both the workstation and the IP phone to use the same physical connection to the switch. 
Jitter, packet delays, and packet drops could all result from deploying VoIP on an existing LAN without implementing QoS. Jitter is a variation in delay, which can cause packets to arrive out of sequence or at a different rate than they were sent. As a result, the end user might experience choppiness in the audio connection. Thus shorter packet roundtrip times contribute to better voice quality. The effects of VoIP issues like jitter and latency on a network can be analyzed by using data analysis techniques such as Mean Opinion Score (MOS) or R-Factor. 
Congested networks often cause dropped packets. 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 QoS and congestion avoidance mechanisms, increasing bandwidth, and increasing buffer space. In addition, some codecs can correct small amounts of packet loss. 
Bandwidth is also crucial to the successful deployment of a VoIP network. A lack of bandwidth can lead to issues such as serialization delay. Serialization delay is the time required to place a packet onto a medium, such as a copper or fiber-optic cable. Serialization delay is directly related to the clocking method and the bandwidth of the line. 
Reference:
Cisco: Network Infrastructure: Impairments to IP Communications if QoS is Not Employed
Symantec: VoIP Hopping: A Method of Testing VoIP security or Voice VLANs

You cannot display Route and Line Group Utilization information by using the Call Detail Records (CDR) Analysis and Reporting (CAR) System Reports menu. In addition, you cannot display the top number of users by maximum length of calls by using the CAR System Reports menu. CAR is a reporting system that can be used to examine a variety of statistics about a Cisco Unified Communications system, including system load and performance. 
You can display Route and Line Group Utilization information by using the Device Reports menu. The Route and Line Group Utilization report can be accessed by clicking Device Reports > Route Patterns/Hunt Pilots in the Cisco Unified Communications Manager (UCM) CAR graphical user interface (GUI). This report enables a CAR administrator to view Route and Line Group Utilization as a percentage? the report can also be used to determine whether capacity needs to be added to an existing Voice over IP (VoIP) implementation. 
You can display 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 display 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 display information about call volume for a given period of time by using the System Reports > Traffic > Summary by Phone Number report in the UCM CAR 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 display malicious call details by using the System Reports > Malicious Call Details report in the UCM CAR GUI. This report enables a CAR administrator to view call information that is tracked by the UCM Malicious Call Identification (MCID) service. An administrator can choose to view MCID information over a period of time. 
You can display 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. 
Reference:
Cisco: Understanding CAR Device Reports: Device Reports Summary Descriptions
Cisco: Understanding CAR User Reports: User Reports Summary Description
Cisco: Configuring route pattern and Hunt Pilot Device Reports: Configuring Route Pattern/Hunt Pilot Utilization Reports
Cisco: Configuring Top N User Reports: Configuring Top N by Duration Reports

You should click Tools > Service Activation in Cisco Unified Serviceability if you want to enable a new set of features. 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. 
You should not click Tools > Serviceability Reports Archive > Service Activation, because the Service Activation option is not available under the Serviceability Reports Archive option. However, 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 > Service Activation, because the Service Activation option is not available under the Alarm menu. The Cisco Unified Serviceability Alarm menu helps identify problems that exist with the Cisco Unified Communications system. The Alarm menu can be used as part of the troubleshooting process. 
You should not click Trace > Service Activation, because the Service Activation option is not available under the Trace menu. The Trace menu can be used to access voice application tracing tools that can be used in troubleshooting efforts. 
Reference:
Cisco: Configuring Services: Activating and Deactivating Feature Services

You should modify the user template and manually reconfigure existing users in Bulk Edit mode to change the voice mail message length limit for the voice mail users to 300 seconds by using the least administrative effort. Modifying the user template to limit voice mail messages to 300 seconds will ensure that any new accounts that are created from that template will automatically be configured with the correct message length limit. In addition, Bulk Edit mode enables an administrator to select a specific subset of Unity Connection users and make identical changes to every user in that subset at once. Therefore, you can use Bulk Edit mode to modify the message length limit for all existing users who have voice mail. 
Modifying the user template and using the Bulk Administration Tool (BAT) to update existing users will not complete your task by using the least administrative effort. The BAT enables administrators to import users, update user settings, and delete users by importing comma-separated values (CSV) files. The BAT is also capable of exporting users to CSV files. In order to update the message length limit by using the BAT, you would need to export the subset of voice mail users to a CSV file, change the message length limit value for each user in the CSV file, and then import the CSV file again. Exporting, updating, and importing the CSV files would consume more time than applying a single change to a subset of accounts at once by using the Bulk Edit tool. 
Modifying the user template alone will not complete your task. If you modify an existing user template, only new user accounts that are based off that template will be affected by the changes that you make to the template. Any users who have already been created from the template will retain the old template settings. 
Modifying the user template and manually reconfiguring existing users one by one will not complete your task by using the least administrative effort. If you were to manually reconfigure existing users one by one, you would need to search for the users, open a user, modify the Maximum Message Length field, save the user, and then open the next user in the search results. Having to manually open and modify each user would consume more time than editing the entire subset of users at once. 
Reference:
Cisco: Adding, Modifying, or Deleting a User Template in Cisco Unity Connection 8.x: Modifying a User Template in Cisco Unity Connection 8.x
Cisco: Managing Cisco Unity Connection 8.x User Accounts in Bulk: Editing Cisco Unity Connection 8.x User Account Information in Bulk Edit Mode