210-060 CICD PDF

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Latest Cisco pdf Test exam practice questions and answers, Try Cisco dumps Brain Du When we confirm the payment for Cisco exam products, all the practice materials will be accessible in Pdf and test engine formats. Powering devices through an Ethernet cable offers many advantages over using a local power supply.

First, you have a centralized point of power distribution. Many users expect the phone system to continue to work even if the power is out in the company offices. PoE also enables you to power devices that are not conveniently located next to a power outlet. For exam- ple, it is a common practice to mount wireless access points in the ceiling, where power is not easily accessible.

Third-party switches from vendors other than Cisco will deliver PoE and power Cisco IP phones, but they may not support Cisco proprietary configurations and capabilities that make management, quality of service QoS , and traffic control easier and more powerful.

PoE became an official standard However, the IP telephony industry was rapidly evolving before this. To power the IP phones without an official PoE standard, some proprietary methods were created, one such method being Cisco inline power.

In addition, some proprietary implementations of PoE have reached 51W of power by using all four pairs of wire in the Ethernet cable. To upgrade all switches to support PoE would be a significant expense.

These organizations may choose to install intermediary devices, such as a patch panel, that are able to inject PoE on the line. The physical layout for this design is demonstrated in Figure By using the powered patch panel, you still gain the advantage of centralized power and backup without requiring switch upgrades. Be sure that your switch supports these features before you consider a power patch panel solution.

Inline PoE injectors provide a low-cost PoE solution for single devices one device per injec- tor. This syntax is a newer form of configuration for IP phone connections.

This was less secure because hackers could remove the IP phone from the switch port and attach their own device another managed switch or PC and perform a VLAN-hopping attack. By understanding the IP phone boot process, you can more fully understand how the Cisco IP phone operates which aids significantly in troubleshooting Cisco IP phone issues. Here is the Cisco IP phone boot process, start to finish: The Cisco IP phone connects to an Ethernet switch port.

Included in the configuration file is a list of valid call processing agents such as Cisco Unified Communications Manager or Cisco Unified Communications Manager Express. The Cisco IP phone attempts to contact the first call processing server the primary server listed in its configuration file to register. If this fails, the IP phone moves to the next server in the configuration file.

This process continues until the IP phone reg- isters successfully or the list of call processing agents is exhausted. Using a router as a DHCP server is a common practice in smaller networks. Once you move into larger organizations, DHCP services are typically centralized onto server platforms.

Using a router as a DHCP server is simple and stable and makes sense if there is already a router in place that could perform DHCP in addition to its routine jobs. That said, most larger organizations use a Windows server or some other centralized device for DHCP services. Cisco routers take the opposite approach: You first specify a range of addresses that you do not want to hand out to clients using the ip dhcp excluded-address syntax from global configuration mode.

Configuring the excluded addresses before you configure the DHCP pools ensures that the Cisco router does not accidentally hand out IP addresses before you have a chance to exclude them from the range.

In Example , this is Likewise, all the NTP-enabled devices on your network will have the exact same time. These advantages make NTP the preferred clock-setting method. The accuracy of the clock on your device depends on the stratum number of the NTP server.

A stratum 1 time server is one that has a GPS clock or atomic clock directly attached. The device that receives its time from this server via NTP is considered a stratum 2 device. The device that receives its time from this stratum 2 device via NTP is considered a stratum 3 device, and so on. There are many publicly accessible stratum 2 and 3 and even some stratum 1 devices on the Internet. Note You can obtain a list of publicly accessible NTP servers at http: If this is the only command you enter, your clock on your device will set itself to the universal time coordinated UTC time zone.

The previous syntax example set the time zone for Arizona to —7 hours from UTC. Now that we configured the router to synchronize with an NTP server, we can verify the NTP associations and the current time and date using the commands shown in Example The asterisk indicates that your Cisco device has synchronized with this server.

After you configure the Cisco router to synchronize with an NTP server, you can configure it to provide date and time information to a CUCM server, which can then provide that date and time information to the Cisco IP phones in your network. For example, as soon as a user picks up the handset of the phone, it sends a SCCP or SIP message to the call processing server indicating an off-hook condi- tion. Hopefully, you get the idea: Quality of Service Quality of service QoS is a topic that is referenced in nearly every chapter of this book.

For a VoIP network to operate successfully, the voice traffic must have priority over the data traffic as it traverses its way from one end of the network to the other. The Cisco defi- nition of QoS is as follows: Quality of service is the ability of the network to provide better or special service to a set of users and applications at the expense of other users and applications. That sounds exactly like what the voice traffic needs as it crosses the network: The voice traffic needs this not so much because of bandwidth require- ments VoIP uses very little bandwidth compared to most data applications , but rather delay requirements.

Unlike data, the time it takes a voice packet to get from one end of the network to the other is critical. If a data packet crossing the network experiences delay, a file transfer might take a couple more seconds to complete or a web page might take a half second longer to load. However, if voice traffic crossing the network experiences delay, conversations begin to overlap a person begins speaking at the same time as another person ; the conversation breaks up; and, in some extreme cases, the voice call drops.

The key here is that we hear the network prob- lems in real-time as it messes up our phone conversation; people notice, and it is unaccept- able because we are used to speech sounding like speech.

To combat these issues, you need to ensure not only that there is bandwidth available for VoIP traffic, but that the VoIP traffic gets the first bandwidth available. This means if a bot- tleneck occurs in the network and a router has to queue traffic before it is sent, the router will move the waiting voice traffic ahead of the data traffic and give transmit priority to the voice packets. Accomplishing this is the job of QoS. QoS is not a tool in itself, but rather, a category of many tools aimed at giving you complete control over the traffic crossing your network.

There might be times when you just use a single QoS tool aimed at decreasing the delay of traffic. Other times, you might employ multiple QoS tools to control delay, reserve bandwidth, and compress data that is heading over the WAN. How and when you use each of the QoS tools depends on the network requirements of your traffic and the characteris- tics such as bandwidth, delay, and so on of the network supporting the traffic.

Current configuration: If a user removes this device, the trust boundary is broken and is not restored until the device is reattached. If you enter the command auto qos voip trust, the switch trusts the markings from the attached device regardless of what it is.

You need to use this command if you download non-Cisco IP phones. Keep in mind that if you use this command, the network sus- ceptible to users removing the non-Cisco IP phone and attaching rogue devices. Notice how many commands are generated after entering the auto qos voip command. It is beneficial that the Cisco switch and router shows you all the individual commands so that you can optionally tune the settings to exactly fit your environment. If the configuration generated by the auto qos voip command is not desired, you can remove this configuration simply by entering no auto qos voip.

Example shows the AutoQoS syntax to use on the switch for the interface connecting to the router. Understanding Cisco IP Phones 79 switchport access vlan 10 switchport mode access mls qos trust cos auto qos voip trust wrr-queue bandwidth 10 20 70 1 wrr-queue min-reserve 1 5 wrr-queue min-reserve 2 6 wrr-queue min-reserve 3 7 wrr-queue min-reserve 4 8 3 wrr-queue cos-map 1 0 1 wrr-queue cos-map 2 2 4 wrr-queue cos-map 3 3 6 7 wrr-queue cos-map 4 5 priority-queue out end You can configure the interface between the switch and router with the auto qos voip trust command, because you would consider the QoS markings from the router as trusted.

Note In Example , after entering the auto? This enables a cool version of AutoQoS that allows the router to monitor your network for an extended time to discover known types of data, voice, and video traffic that are considered higher priority based on common high-priority application types.

After the router captures enough traffic, it generates QoS policy recommendations that you can choose to apply or ignore. Chapter 3: This sec- tion details the required steps to get the router ready for Cisco Configuration Professional CCP to connect to and configure the router. CCP has evolved to become the most fully functional all-in-one graphic interface for managing your CME router. This section discusses the CCP interface and its initial configuration tasks.

This decision was not taken lightly; people have strong opinions about this, myself included. The CLI is the most powerful and arguably the fastest method of configuring CME; in the real world, many admins working with CME are using the command line exclusively.

But the fact is that there is already a lot of stuff to learn for the exam, and parts of the CLI were deferred to more advanced study. As the author of the Official Certification guide, my primary goal is to give you the infor- mation you need to pass the exam. My secondary goal is to help you understand the sys- tems discussed in the book at a level of proficiency that makes you able to set them up and administer them at a basic level.

Given that, here is what I have done for this new edition: I have removed large parts of the CLI from the body of the book and moved it to an appen- dix at the end. I did this mostly because removing it lines up with the exam blueprint and the CICD course, but keeping it available makes it a great reference to have handy. And, of course, it is always possible that the people who create the exam might change their minds again and put the CLI back into the exam after we print this book, and if they do you will still have the CLI material to study.

Creating a privilege level 15 user account c. Enabling HTTP services on the router d. Configuring Telnet or SSH for local logins 2. Choose three.

What type of archive can you download from Cisco to reinstall all support files for the integrated GUI back into the router flash? TAR file b. ZIP file c. GZ file d. RAR file 5. Which version of CCP installs into the flash of a Cisco router?

There is only one CCP version, which cannot install into the router flash. CCP Lite. CCP Express. Chapter 4: Which of the following best describes a community in CCP? A group of users managed by a single administrator b. A group of up to 10 devices enabled for CCP configuration c.

A group of up to 5 devices enabled for CCP configuration d. A group of similar devices that share a common configuration 7. How does the communication change by selecting the Connect Securely check box when adding a device to a CCP community?

Only level 15 user accounts are permitted to access the device. First, many small offices employ an all-in-one administrator whose knowledge and time is spread across many different technologies.

Requiring this level of administrator to learn a complete command-line operating system to interact with CME is unrealistic. Similarly, some offices use consultants or contract network administrators to manage their network. Providing an easy-to-use graphical user interface GUI allows one of the more technically inclined users at the office to take care of the day-to-day administra- tion changing directory numbers, adding phones, and so on without the involvement of dedicated IT staff.

Finally, the point-and-click of a graphic interface can be more efficient at times than typing configuration commands. Typically, the CME router ships with these files preloaded by Cisco into the flash; however, you can also download a. TAR package of files from Cisco.

You can download the latest version of the CCP software from the Cisco website free of charge. After you install it, you can manage any supported Cisco platform using the utility. You do not need to install any- thing on the managed devices, but they do require the configurations described in the previous section.

Most brand new devices have a default startup-config file loaded that implements the necessary configs when you boot them, and all you have to do is get the correct IP address on your PC and some models have a DHCP service running by default, so you do not even need to set the IP manually. You can also download the appropriate startup-config file for your device from Cisco if you want, instead of configuring it yourself. It is not able to configure Unified Communications features.

When you initially open CCP, it prompts you to configure a community, as shown in Figure Secure connections are always better. After this process completes, you can configure the device. Although CCP has many configuration options available relating to routing and security, we primarily focus on the Unified Communications features, as shown in Figure None configures the router as a voice gateway only does not support IP phones.

After CCP applies the initial CME configuration to the router, the Unified Communications menu refreshes to display additional configuration options, which will vary depending on the selections made on the Features screen.

The upcoming chapters discuss several of these configuration screens as we move on and continue to focus on using CCP to set up CME. Application development and operating system development get out of sync and can cause compatibility issues, typically when trying to run CCP on Windows 7 and 8 machines. Java has also had some compatibility issues, and we sometimes find ourselves in the situation where an application such as CCP requires an older version of Java, which is not secure and maybe even not available any more.

There are ways to make it all work, and there are sev- eral good threads in the Cisco support forums and elsewhere on the Internet that go into detail on exactly how to do that. It does work well once you get all the pieces in place; you just need to do your research and pay attention to detail. It is not related to the exam, so I do not go into that detail here. One of the best pieces of advice I can give you is this: The virtual machine can also be prevented from accessing anything other than its intended devices to improve security.

You should be learning about building virtual machines anyway, and this tactic will become a very useful one as you move forward. This section reviews the theory of end user configura- tion in CME. If you are in doubt about your answers to these questions or your own assessment of your knowledge of the topics, read the entire chapter.

If you do not know the answer to a question or are only partially sure of the answer, you should mark that question as wrong for purposes of the self-assessment. Giving yourself credit for an answer you correctly guess skews your self-assessment results and might pro- vide you with a false sense of security.

CME provides three user access levels. What are they? Remote user b. System administrator c. Privileged user d. End user e. Customer administrator 2. Which command authenticates GUI users against entries in the running configuration? What does the command web admin system name username password password do? Identifies a web GUI configuration profile tag b.

Defines the system administrator user for CME c. Defines the customer administrator user for CME d. Enables the router web server for the specified user 5. Choose all that apply. Dynamic routing configuration b. VPN configuration c. Unified Communications configuration d. Access list configuration 6. Harry is building a phone in CME. Which of the following would be an accurate statement by Harry? An ephone-dn is assigned to a SIP ephone.

A voice register dn is assigned to a SIP ephone. A voice register dn is assigned to a SIP voice register global. A voice register dn is assigned to a SIP voice register pool. Chapter 5: Which of the following are required steps to create a phone with a functioning line in CME? Create a phone ephone. Create an extension ephone-dn. Associate the extension with the phone. Autoregister the phone. Bob has decided not to use an external TFTP server for all his new SIP phone firm- ware loads, even though that is possible and would work fine.

Which of the following is true for Bob? Bob can either use existing firmware files or upload new ones to the router flash using CCP under the Unified Communications menu. In CME, how would you create a directory entry and an internal caller ID name that is visible on the IP phones registered to the system? Create an extension using CCP b. Create a phone using CCP c. Create a directory entry list using CCP d.

Create an address list service using the CLI Which of the following is true of the relationship between users, phones, and exten- sions in CME? A user must be associated with a phone. Associating a user with a phone allows that user to make changes to their phone, including speed dials and ringer choices.

There is no telephony functionality at all if the user is not associated with the phone. Some place the files in the root of the flash directory; others specify a flash: In my experience, placing the files in the root of flash works well.

The list that follows summarizes the steps to get the built-in GUI running: Step 1. Step 2. Extract the GUI files to flash. Step 3. Step 4. Configure the router with the path to the GUI files. Step 5. Set the method of HTTP authentication. Step 6. Create the CME system administrator account. This may be the same account as the one you used for CCP, or it may be different.

Step 7. Example summarizes the commands to perform those actions. Optional Enables the ability to add directory numbers through the web interface. Router config-telephony time-webedit! Enables the ability to set the phone time for the CME system through the web inter- face. Ephones are identified by the device type , for example and MAC address. Other configurations can be applied to the ephone to control button capabilities assignments such as speed dials and intercom, softkey templates, and many other system and feature configurations—including the associ- ated username.

The associated username is important because it allows that named user to modify the phone configuration—adding a speed dial or setting the ringer, for example— from the CME GUI. The relationship between the ephone-dn and the user differs slightly.

An ordinary user is not allowed to modify the ephone-dn configuration. Associating the user to an ephone-dn serves multiple purposes: The configurations are completely different, executed under a different configuration prompt voice register global instead of telephony-service at the CLI , but there are very clear parallels: Table provides a quick reference for you to keep those terms straight.

We will examine the CCP interface and its general capabilities, and although the focus remains on Unified Communications configurations, you will gain some insight into the other capabilities for GUI-based configuration that CCP offers as well.

From the Configure menu shown in Figure 5. This is not a complete list, just some of the highlights. Flash file management, configuration editor, save config to PC, write to startup config, Telnet, reload device, ping and traceroute, view running config, view IOS show commands, and so on For this book, we focus on the Unified Communications menu.

These settings are related to your licensing limits but have addi- tional important implications because the router will reserve RAM memory for the number of phones you specify, so you will want to be accurate with your selections. Remember that an extension ephone-dn is not just a number you can call; it also provides one of the associations for an end user as well as sever- al other configuration options.

Figure shows the creation of extension Note that we have identified the username of Gord Downie for the extension, but the user association is not complete until we create a phone for Gord and associate the extension to it. From the list of extensions at the lower left, select the appropriate entry in this case and move it to the Associated Extensions column by clicking the right arrow. Figure shows this step completed. This is where we create the end user; note that in doing so on the Phone configuration screens, we automatically associate the user with the phone.

Figure shows us creating the end user, leaving the PIN and password blank. Complete the phone and user cre- ation operation by clicking OK. We have implemented the required configuration to allow us to use the built-in GUI and CCP to create extensions, phones, and users and to associate users to phones and exten- sions. At this point, you should be able to ring phones, look at the directory, and see the internal calling name for IP phone-to-IP phone calls.

Note For those of you using this book as a step-by-step guide to setting up your voice lab to study for the exam, remember that CME is and always will be a CLI environment at heart. It is not unusual. Whether this is due to your operating system settings, browser brand or version, or Java or other software or configuration is a separate issue, the objective for the exam is to learn what CCP can do and how.

According to the OSI model, the physical layer is where it all begins. Similarly, some basic voice-port configuration items can prove essential to setting up your connections to the legacy voice environ- ment. This section breaks down the key commands used in analog and digital voice connections.

If there were one most important topic for all things related to voice in CME, this would be it. Even the simplest of all VoIP environments need to modify dialed digits or caller ID information for incoming and outgoing calls.

This section breaks down the core digit-manipulation commands and explores the flow of a typical CME-handled call. Just like you can use access lists to secure the data plane of your router, COR lists secure access to dialed numbers in your VoIP environment. This section discusses the concepts and configu- ration of COR lists. Plugging in the cable is the easiest part of the configuration. After the physi- cal connections are in place, the router configuration begins. Instead of routing tables, voice gateways build the logical dial plan through a system of dial peers.

This chapter explores the configuration and testing of dial peers in a voice environment. Which of the following interface types would you use to connect an analog fax machine to the VoIP network? FXS b. FXO c. BRI 2. Which of the following commands would you use to configure a T1 line to use chan- nels 1 through 6 to connect to the PSTN using FXO loop start signaling?

What type of dial peer would you create? Analog b. Digital c. POTS d. VoIP 4. You have the following configuration entered on your voice router: What is the default codec used by a VoIP dial peer?

Which of the following destination patterns could you use to match any dialed num- ber up to 32 digits in length? After you create a translation rule, how is it applied? To an interface b. To a translation profile c.

Globally d. Which of the following digit manipulation commands will work under a VoIP dial peer? What is the final method used by a router to match an inbound dial peer for incom- ing calls? Using the answer-address command b. Using dial peer 0 c. Using the port command d. Using the destination-pattern command If an ephone-dn lacks an incoming COR list and attempts to dial a dial peer assigned an outgoing COR list, what behavior occurs? CME denies the call.

CME permits the call. The call is rerouted to the next dial peer without an outgoing COR list. Which of the following is true?

Chapter 6: Understanding the CME Dial Plan When you are ready to configure your FXS voice ports, the best place to start is to find out what voice ports your router is equipped with. You can do this quickly by using the show voice port summary command, as shown in Example FXS ports have three common areas of configuration: The signal type dictates the method used by the attached device to signal that a phone is going off-hook. Table briefly describes the differences between ground start and loop start signaling.

Based on your geographical location, dial tones might be higher or lower and busy signals might be fast or slow. These are all considered call progress tones: If your router is serving another part of the world, use the command shown in Example to adjust the call progress tones. Finally, you can use the syntax shown in Example to configure caller ID information for the device attached to the FXS port. These cards offer you the flexibility to configure them for a data con- nection or a voice connection.

Unlike analog interfaces, you must configure digital interfac- es before they will operate correctly because the router does not know the type of network you will be using. The type of network to which you are connecting dictates the command you use to configure your VWIC card: This command enables you to identify the T1 interfaces on your router.

These interfaces do not appear in the show ip interface brief output because the router does not know whether you will configure the interface as a voice or data connection that is, whether it will end up with an IP address.

This gives you the ability to connect to many different network types. This might differ depending on your location and service provider. After you enter the ds0-group command, the router automatically creates a voice port for each time slot you provision, as you can see from the show voice port summary output in Example Make a note of this port identifier because you need it to configure the dial peers.

Each port listed represents a different chan- nel on the T1 interface. Example sets this to primary- 5ess. After you configure the switch type, the router allows you to enter the pri-group command.

This works identically to the ds0-group command in that it allows you to provi- sion a specific number of time slots for use with the PSTN carrier. If your router does not have enough DSPs, it displays an error message when you try to use the pri-group command.

The error message says exactly how many channels the router can support. After you enter the pri-group command, the router creates 24 ISDN voice ports that it will use for incoming and outgoing voice calls. This is verified with the show voice port summary command. This represents channel 23 time slot 24 of the T1 ISDN PRI connection channels are listed from 0—23, whereas time slots are listed 1—24 , which is the dedicated signaling channel used to bring up the other 23 voice bearer channels.

So, to provide a two-way calling environment that enables x to call x and vice versa, you need to configure a total of eight call legs. It is critical to understand the concept of call legs to properly configure the dial peers on your router. Each call leg identified in Figure represents a dial peer that must exist on your router. These dial peers define not only the reachability information phone numbers for the devices, but also the path the audio must travel.

As you can see from Figure , call legs are matched on both the inbound and outbound directions. In the same way, you must configure dial peers to match voice traffic in both the 6 inbound and outbound directions. In some cases, you can use a single dial peer for bidirec- tional traffic. For example, creating a single POTS dial peer for x will match incoming and outgoing calls to x At other times, you must create more than one dial peer for inbound and outbound traffic.

As you see the multiple examples of dial peers in the upcom- ing sections, these concepts become clearer. These workhorses convert the analog audio to digital VoIP packets , and vice versa. Understanding the CME Dial Plan Note Although you can manually create an international dial plan without using the T symbol, doing so can become tedious. The forward-digits number command allows you to specify the number of right-justified digits you want to forward.

Notice the first dial peer 90 in Example With a destination pattern of 9[]11, the router would automatically strip the 9 and the two 1s from the pattern before sending the call; this is because of the default behavior of POTS dial peers, which is to strip remove all the digits that match the destination pattern before forwarding them on to the target of the dial peer.

By entering the command forward-digits 3, the router forwards the last three right-justified digits , , or and only strips the 9. The prefix number command adds any specified digits to the front of the dialed number before routing the call. This is useful for dial peer 93 in Example Now, Example shows what happens if you add a third dial peer to this configuration. Example Sample Dial Peer Configuration 2 dial-peer voice 1 voip destination-pattern [] Likewise, the router processes only the digits and drops the digits.

This can be useful for emer- gency patterns such as or in North America because the call is immediately routed when a user dials this specific pattern. Tip If you ever have a question of which dial peer will match a specific string, Cisco rout- ers include a handy testing feature. From privileged mode, enter the command show dialplan number number, where number is the number you want to test.

The router dis- plays all the matching dial peers in the order in which the router will use them. The router lists more specific matches first. Tip Because the router immediately routes the call after it makes a specific match, it is best to avoid overlapping dial plans if possible.

Avoiding overlapping dial plans may be impossible at times. In these cases, you need to get creative with your dial peers to accomplish your objectives. For example, if you are required to have a dial peer matching the destination pattern while a second dial peer has the destination pattern Note The router applies the num-exp command the instant it receives a dialed number, even before it attempts to match an inbound dial peer.

Practical Scenario 3: Specific POTS Lines for Emergency Calls As organizations move more to VoIP connections, they are finding cost-saving benefits by eliminating traditional telephony connections at remote offices in favor of centralizing all PSTN calls and toll charges at a central site. Figure illustrates this type of network design.

This type of call routing allows an organization to get higher call volume from a single loca- tion, which typically allows the organization to negotiate cheaper long-distance rates with its PSTN carrier. You should always check with the local government regulations before you do this. Thankfully, the United States is not one of those countries.

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Although the centralization of PSTN calls offers significant cost savings, the remote sites need to keep at least one local PSTN connection for emergency calling. If emergency calls from the remote offices were to traverse the IP WAN and leave the PSTN connection at the central site, the emergency service provider would receive location information for the central site.

Depending on the size of the remote office, you can typically dedicate one or two analog FXO ports for emergency calls. For example, you can add some digits using the prefix command or ensure digits do or do not get stripped with the forward-digits command. The num-exp command allows you to make the biggest changes of all, but these changes are applied glob- ally to the router, which might not give you the flexibility all situations require.

Translation profiles are useful to address these needs. Implementation of translation profiles requires a three-step process: Define the rules that dictate how the router will transform the number.

Associate the rules into a translation profile.

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Assign the translation profile to a dial peer. In a way, this is similar to access-list configuration on a router. To demonstrate the configuration of translation profiles, consider the scenario illustrated in Figure This allows PSTN callers to dial directly into the organization without being redirected by a receptionist. The administrator of this network would like to translate all 6XXX dialed numbers to 5XXX, but only if these dialed numbers come in from the T1 PSTN interface, so as to not interfere with the numbering scheme of the remote office.

To accomplish this, he cannot use the num-exp This situation is ideal for translation profiles. The first step to configure translation profiles is to create the translation rules. These use the general syntax shown in Example You can use the test voice translation-rule command from privi- leged mode to test the rules you create before you apply them, as shown in Example Next, you need to take the voice translation rule and assign it to a translation profile.

Because the scenario requires you to change the DID information, this is the proper assignment. Assigning the translation rule as a calling translation would change the caller ID of a person calling into the organization.

The last step is to assign the translation profile. Understanding the CME Dial Plan Notice that the example applies the translation profile in the incoming direction.

This causes it to affect calls coming in from the PSTN rather than outgoing calls. The translation profile is now in effect, accomplishing the objective of the scenario. Note You can do far more with translation profiles and far more complex patterns that you can match with translation rules.

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With all these various methods of digit manipulation, two questions quickly arise: Which method gets applied first? Will the router remove added prefix digits because of the auto- matic digit-stripping rule?

Figure answers these questions by displaying the order of operations for outgoing POTS dial peers. Cisco neatly organized all dial plan configurations in a Dial Plans folder, shown in Figure These correlate to the two categories of dial peers you most frequently configure in CME. In an attempt to ease the configuration, Cisco provides three wizard-like configuration items: Allows you to configure inbound dial-peers using PSTN trunks. Allows you to configure outbound dial-peers using PSTN trunks.

The wizard also provides you the ability to specify access digits such as dialing 9 for an outside line and caller ID information. Allows you to import a dial plan from a CSV file template. In this configuration pane, you can create manually configured dial peers through a GUI configuration. Figure shows the creation of a ten-digit dialing PSTN dial peer. However, there might be times when you want to prevent some users from calling certain numbers, such as the following examples: Sometimes, it is necessary to place calling restrictions on users of the VoIP network.

Without a doubt, COR lists take some practice before the concept sinks in fully. A corporation wants to implement the following restrictions in CME: It breaks down into the following steps: Define the COR tags we will use for the restrictions.

Create the outbound COR lists. Create the inbound COR lists. Assign the outbound COR lists. Assign the inbound COR lists. This seems like an extensive process, but it goes rather quickly when you get to enter the syntax. To tackle the first step, we must define the tags we will use for the restrictions.

Some documentation calls them COR list members, whereas other documentation calls them keys. Here, we call them tags. The tag names you create are typically based on the restrictions you want to apply. Example shows the process of defining these tags. At this point, these tags are doing absolutely nothing, but because we defined the names, the CME router allows us to use them to create COR lists. Next, we create the incoming COR lists, as shown in Example Both the inbound and outbound COR lists are created the same way, but the effect they have is based on how you apply them.

Example shows how this is done. Example puts it all together. You might need to flip back through the last couple pages to see the building examples to fill the pieces in your own mind. To see this in action, we follow a call originated from the lobby phone ephone-dn 1: Someone picks up the lobby phone and calls Because the lobby phone was only assigned the tag and not LOCAL , the call fails with a reorder tone. Okay, you have now seen how COR lists deny a call. An employee picks up her phone and calls Click Edit, and in the configuration window, select Call Restrictions in the left pane.

Note that there are settings for both incoming permis- sions and outgoing permissions. Make the appropriate changes for the other extensions one at a time. Remember, if you choose not to apply a COR list to an extension, that extension will be able to dial any destination number. This section walks through the creation of a local directory of CME devices, which gives your users an easier method to find and dial local DNs.

This section discusses the concepts and configuration of the call-forwarding features in the CME environment. This section discusses the concepts and configuration of the call-transfer features in the CME environment. This section discusses the concepts and configuration of the call park features in the CME environment.

This section discusses the concepts and configuration of the call pickup features in the CME environment. This section discusses the concepts and configuration of the intercom features in the CME environment. This section discusses the concepts and configuration of the paging features in the CME environment. This section discusses the methods you can use to allow or deny specific dialing patterns in the after-hours time frame for all or specific IP phones.

This section discusses the configuration of CDRs and call-accounting features. This section covers the simple feature of shared ephone-dn shared line. Beyond basic tele- phony, organizations expect modern telephony systems to support a whole host of features, such as call transfer, Music on Hold MoH , conference calling, and so on.

This chapter is dedicated to adding these types of features to the CME voice network. What process must you follow to build the local phone directory for the CME environment? Assign directory entries under each ephone-dn using the directory command. Assign directory entries under each ephone using the directory command. Enter the directory configuration mode and begin associating ephone-dn values with directory entry values.

What three conditions can be configured for call forwarding settings in CCP? Forward all calls b. Forward when busy c. Forward all to voicemail d.

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Forward unattended calls 3. What happens if the user selects Do Not Disturb on his phone but there is no Call Forward setting configured for that line? The call is automatically forwarded to voicemail. The call is automatically forwarded to the operator. The call is dropped. The call information is displayed on the phone screen, but it does not ring and the visual indicator lamp does not blink. Which of the following transfer modes does a Cisco router support by default?

Blind b. Consult c. Full-blind d. Full-consult 5. What is required to provide call transfer capability to CME users? Add the Transfer softkey to the softkey template configuration for the phone. No action is required because the default template includes the Transfer softkey.

Enable H. Night Service is a legacy feature that has been deprecated in current versions of CME. Night Service automatically activates the auto-attendant during closed hours. Up to three IP phones can be designated for Night Service. Night Service rings a designated ephone-dn during a specified schedule, or when manually activated using a code.

Pick up a ringing phone in your group b. Pick up a ringing phone in another group c. Answer your own ringing phone d. You are watching an administrator configure an intercom line using CCP.

You notice that the intercom directory number was autocreated as A What is the signifi- cance of this number? The A in A indicates that this is the first intercom button on the phone.

The A in A prevents the number from being dialed from an IP phone. A is autogenerated to indicate that an intercom button is configured on button 8 to ring the target phone that has extension A is autogenerated to indicate that an intercom button is configured on the phone with extension , on button 8.

What is the maximum number of paging groups to which a Cisco IP phone can belong? No practical limit A user is on an active call at the office using his desk phone. Midway through the call, he presses the Mobility softkey on the screen of his IP phone. What process occurs? The call transfers to his preconfigured single number reach number. CME places the call on hold and allows retrieval from a remote phone.

CME places the call on hold and allows retrieval from a predefined call park number. The user logs out of the phone and then logs in to a new phone where he retrieves the call. You can enter names under ephone-dn configuration mode either as you are configuring new lines for the organization or separately, after you configure the lines. These names are used both for building the internal corporate phone directory often called the local direc- tory and for caller ID information.

Most of the current Cisco IP phone models allow you to browse the corporate directory by pressing the Directory button on the phone itself. Some low-end IP phones may not have a dedicated Directory button, but instead have a menu-driven process to get there. After you press the Directory button, you are able to browse categories, including Missed Calls, Received Calls, and so on. Move down to the option showing the Local Directory, as shown in Figure You can enter as many characters as you like to filter down the number of results, or simply press the Select soft- key to see the entire corporate directory, as shown in Figure Chapter 7: You can change this setting by using the directory command from telephony service con- figuration mode.

In addition, you can also add manual entries to the directory by using the directory entry command. This is useful for devices in the company that do not have an explicit ephone-dn configuration. Example demonstrates these two commands in action. WORD A sequence of digits representing dir.

LINE A string - representing directory name max length: Also, keep in mind that sorting alphabetically by last name flips all the information in the directory to list last name first. You might remember that the CCP utility does not associate extensions ephone-dns directly to phones ephones. Instead, after you create the necessary extensions and phones, they are linked together through the user account. Once you add a first name and last name to the user account, the name is applied to the extension associated to that user account.

Although this is typically a voicemail number which represents in Example , this could also be another IP phone if this DN was a member of a hunt group.

Tip In Canada and the United States, the phone rings for 2 seconds followed by 4 seconds of silence. Knowing this can be useful in calculating a good no answer noan timeout value. Also notice that you can specify a max-length value after the call-forward command. Using this, you can restrict the IP phone from forwarding to external destinations. Tip At this point, you should have a good idea that plenty of configurations under each 7 ephone-dn are similar to all the others.

Make an ephone-dn and ephone template in Notepad or some other text editor in which you list all the common configuration commands you will be applying in your environment.

That way, if you ever need to add new ephone-dns, you will already have a template listing the common commands you need to enter. Using the call-forward pattern Command to Support H. This command enables you to enter a pattern for numbers that will support the H. To understand the benefits of H. When a call enters the network and hits a forwarded device, that device takes responsibility for the call and becomes a tandem hop in the call flow. That means that the voice traffic now forwards through the IP phone that forwarded the call.

This can cause quality problems if the device that forwarded the call is a large geographical distance away from the phone receiving the forwarding call. The H. Figure illustrates this concept. This is the recommended mode for most systems.Time slot 16 c.

This command enables you to enter a pattern for numbers that will support the H. Night Service is a legacy feature that has been deprecated in current versions of CME.

She thought she was too sympathetic to her man. A stratum 1 time server is one that has a GPS clock or atomic clock directly attached. Your assistance made this one happen; I will download the beer when we finally meet in person. Which of the following governs the number of concurrent calls supported by a Cisco Unity Connection server? This application which has multiple caller platform capa- bilities allows callers to click a website link and establish a video call on demand, even if they do not have a video endpoint or soft client installed.

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