Project Manager: Susan S. Bradley Content Architect
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- Bu səhifədəki naviqasiya:
- Configure extensions.conf
- Overview of the extensions.conf file
- Constructing Your extensions.conf File
- Configuring the X-Lite Client
- Troubleshooting Asterisk
Installation of AsteriskThere are a variety of ways to implement Asterisk, and its many flavors or subversions. In this example, we show one of the quickest ways to install and configure Asterisk. To deploy Asterisk on a CentOS virtual machine, you must install the following modules:
On CentOS, you can install these modules by using Linux Yellowdog Updater Modified (Yum) as follows: yum –y install gcc gcc-c++ libtermcap-devel kernel-devel bison openssl-devl ncurses-devel Note. If you run an symmetric multiprocessing (SMP) kernel, you need kernel-smp-devel instead of kernel-devel. The latest release of Asterisk 1.8 supports communication with Google Talk natively. To install Asterisk, log on as root or if logging on as a regular user, use the sudo command. To download Asterisk on the CentOS server, change to the directory /usr/src and then run the wget command to download the package. After you download it, create a directory called /asterisk and untar the package. Run the commands as follows:
Move to the /asterisk directory that you created, and then install Asterisk by running the following commands: ./configure make make install make samples make config Note. The make samples command creates a default set of configuration files for Asterisk. These can be a useful reference if you are unfamiliar with Asterisk config files. Then, restart the server to complete the installation by using the following command: shutdown –r now To access the Asterisk console, run the following command: asterisk -vvvvvvr Note. The letter r is for remote. The number of v characters sets the verbose level of the Asterisk console, which is a verbose level 6 in this case. ConfigurationTo help you setup direct SIP with Asterisk, step-by-step instructions for configuring Asterisk, Lync Server and the X-Lite SIP client for Asterisk are discussed in this section. Configure AsteriskTo configure Asterisk, you must edit a series of configuration files. The following files are in a text format and are normally found in the /etc/asterisk directory:
Configure sip.confThe sip.conf file defines all SIP configurations for Asterisk. The first section in this text file is labeled [general]. In the [general] section, you define general SIP settings for the entire Asterisk server. We recommend that you build a new sip.conf file and use the original for reference. This makes a cleaner configuration file. Create a directory under /etc/asterisk called "samples" and then copy all files into this directory as follows: mkdir /etc/asterisk/samples cp /etc/asterisk/*.* /etc/asterisk/samples The following is an example of a [general] section from a sip.conf file. [general] context = default bindport = 5060 bindaddr = 0.0.0.0 tcpbindaddr = 0.0.0.0 tcpenable = yes Note. This example is only basic. For a detailed explanation of configuring the Asterisk sip.conf, see the Asterisk Wiki website at http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf. Following is an explanation of each field in this file:
Next, we define a section in the sip.config file to instruct Asterisk how to connect to the Lync Server 2010, Mediation Server. In Contoso’s example, the administrator named this section, to_lync; however, you can call it anything you like in your sip.conf file, except general because that is a special section in the sip.conf file. [to_lync] type = peer host = qualify = yes transport = tcp Let’s examine each of these lines that define this SIP object:
Finally, we need to add a section to our sip.conf file to enable our X-Lite desktop client to connect to Asterisk. We add the following lines to the end of the sip.conf file. [ type = friend callerid = caller_name secret =
dtmfmode = rfc2833 disallow = all allow = ulaw transport = udp Let’s examine each of these lines:
Let’s put all of this together in one complete sip.conf file, using Contoso’s example. [general] context = default bindport = 5060 bindaddr = 0.0.0.0 tcpbindaddr = 0.0.0.0 tcpenable = yes [to_lync] type = peer host = 192.168.100.20 qualify = yes transport = tcp [1000] type = friend callerid = Test User 1000 secret = 1000 dtmfmode = rfc2833 disallow = all allow = ulaw transport = udp In the previous example, only one context called default is defined in the [general] section. All calls that Asterisk receives, regardless of where they come from, are assigned a context. This context dictates which dial plan (that is, extensions.conf) Asterisk should use to process incoming calls.
After you’ve modified and saved the sip.conf configuration file, you must reload it using the following commands: asterisk -vvvvvvr asterisk*CLI> sip reload To view the configured SIP clients and devices, run the following command: asterisk*CLI> sip show peers Configure extensions.confThis file defines the dial plan configuration in Asterisk. The dial plan dictates how calls flow in Asterisk. Every incoming call that Asterisk receives is processed depending on the instructions defined in the dial plan. Note. A full and detailed explanation on dial plans in Asterisk would fill an entire book. The following information is enough to get started with a basic Asterisk dial plan. For details, see www.voip-info.org/wiki/view/Asterisk. In the previous sip.conf section, we talked about using a context in Asterisk and how this context dictates where an incoming calls is processed in the dial plan. We are going to see how this context is used in the dial plan. We are also going to examine two other concepts within an Asterisk dial plan, extensions and macros. First, let’s look at how an extensions.conf file is structured. Overview of the extensions.conf fileAs with the previous section regarding the sip.conf file, the extensions.conf file has two special sections that must be present and must be named in a specific way. These are the [general] and [globals] section. They must appear at the beginning of the extensions.conf file. You may choose to leave them empty, but the sections must be present as follows: [general] [globals] Following these, you can define your own sections. Asterisk calls these sections contexts. This is the context (or contexts) where your incoming calls will be processed. Therefore, it is important that you name these contexts with the same name that you have referenced in your sip.conf file.
ContextsContexts are nothing more than a convenient container for grouping extensions. We'll look at extensions in the next section. You name the context by placing the name between square brackets. In our sip.conf file we put together in the previous section, we used only one context and called it default, so in our extensions.conf file, we'll define a context called [default] as follows: [general] [globals] [default] Inside this new [default] context, we'll add some instructions for dealing with calls. Asterisk refers to these instructions as extensions.
ExtensionsExtension lines in an Asterisk dial plan are defined as: Exten => For example: exten => 1000,1,command exten => 1000,2,command exten => 1000,3,command Extension lines are composed of the following components:
While the exact number being called can be matched (for example, 1000), number patterns can be used to match a wider range of numbers (for example, _1XXX matches all numbers starting with 1 and containing three more numbers). For details about how to configure extension lines in Asterisk, see the Asterisk Wiki at http://www.voip-info.org/wiki/view/Asterisk. Table 2 defines the special characters that you can use to define number patterns. Table 2. Number Pattern Table
Let’s make things a little clearer with some examples. Examples: exten => 333 Matches calls to 333 only. exten => _3XX Matches all calls between 300 and 399. exten => _1XXX Matches all calls between 1000 and 1999. This number pattern does not match calls in the 100 to 199 range. Let’s look at a simple example using three extension lines in a dial plan: exten => 333,1,Answer exten => 333,n,Playback(demo-moreinfo) exten => 333,n,Hangup Let’s examine each line:
Note. If you are new to Asterisk, all of this may seem difficult to understand. We are covering only the basics to get your Lync Server working with Asterisk. For details, see www.voip-info.org/wiki/view/Asterisk Constructing Your extensions.conf FileLet’s take a look at Bob’s example extensions.conf file and use this to help you construct your own. There are some new commands that we'll use in this file, but we'll examine each line to show you what is happening. Here’s Bob’s extensions.conf file: [general] [globals] [default] exten => _+XX.,1,Set(CALLERID(num)=${CALLERID(num):1:11}) exten => _+XX.,n,Goto(${EXTEN:1},1) exten => 1000,1,Dial(SIP/1000) exten => 1000,n,Hangup exten => 14255550150,1,Dial(sip/to_ocs/+${EXTEN}) exten => 14255550150,n,Hangup Let’s examine each of these lines:
The following sections describe the exten=> lines in more detail. exten=>_+XX.,1,Set(CALLERID(num)=${CALLERID(num):1:11})Note the underscore character (that is, "_") indicates that this is a number matching pattern being used. The +XX. matches any number starting with a + and having at least two more digits (that is, XX), followed by any other amount of digits (using the period character (that is, ".") as a wildcard). This matches numbers coming from Lync, starting with + and that are at least three digits long or longer. Next, the number 1 indicates this is the first instruction for numbers that match this pattern. Asterisk has a number of special variables that can be referenced from within a dial plan. On this line, we are referencing the number associated with caller ID. The command is removing the + character from the caller ID of outgoing Lync calls, and replacing that caller ID with the last 11 digits from the original caller ID. For example, a caller ID coming from Lync of +14255550150 would be changed to 14255550150. Note the placement of square brackets (that is, "[ ]") and the curly brackets (that is, "{ }"). It is important where you use these in an Asterisk command. exten=>_+XX.,n,Goto(${EXTEN:1},1)Again we see the same number matching pattern as the last command. In this case, it is followed by the letter n, which indicates that this is the next instruction for numbers that match this pattern. The Goto command is telling Asterisk to move this call to another point in the dial plan. It is also referencing another special Asterisk variables, called EXTEN. This is the extension being called (that is, the number being called). In this case, the outcome is to remove the first digit of the number being called. For example, if the number being called is +1000, this command removes the + sign and moves the call to the 1000 section of the dial plan. Note the 1 at the end of this command, which tells Asterisk to move to the first instruction for 1000. This ends the instructions for numbers that match the number pattern of +XX., but we do not need to do anything to tell Asterisk this, we can just start a different set of extensions. exten=>1000,1,Dial(SIP/1000)This matches the number 1000 only. We can tell from the 1, after the number matching section, that this is the first instruction in this series. Here we can see the dial command being used in an Asterisk dial plan. This particular example tells Asterisk to place a call out, using SIP, to the object called 1000. In the previous sip.conf example, we defined a SIP object in that file called 1000. That object was to be used by our X-Lite desktop client to connect to Asterisk. That object is being referenced here in the dial plan to place a call out to X-Lite. exten=1000,n,HangupThis matches the number 1000 only, and we can tell from the character n, that this is the next instruction for 1000. It’s useful to understand that Asterisk will not move to this line until the call created by the Dial command is finished. The Hangup command instructs Asterisk to end the call. It’s always a good idea to add an extra, final line using the Hangup command just to make sure all calls are terminated when they should be.
exten=>14255550150,1,Dial(sip/to_lync/+${EXTEN})This line of instruction may seem a little daunting, but it's the same as the others. Let’s break it down. We see that this instruction matches the number 14255550150 only. Then, we see a single number 1, showing this is the first instruction for this number matching pattern. Finally we see a Dial command. In this example, it tells Asterisk to dial, using SIP, to the SIP object called to_lync. Remember the sip.conf file? We created a SIP object to link to the Mediation Server called to_lync. It also adds the end section of +${EXTEN}. This instructs Asterisk to add a + character to the beginning of the number being called (that is, 14255550150), and pass that to the to_lync object (that is, the Mediation Server). To explain this more simply, Asterisk pattern matches calls placed to the number 14255550150, adds a plus + to the front of that number, and dials +14255550150 to the Mediation Server (that is, defined in the sip.conf file as the to_lync object).
exten=>14255550150,n,HangupThe final line on our dial plan again matches the number 14255550150 only. We can see from the n character that this is the next instruction for calls to this number. The Hangup command instructs Asterisk to end the call. Remember that Asterisk will only reach this second line after the call created in the first line (that is, the Dial command) is finished. Save changes to the extensions.conf file, and then reload the dial plan using the following commands:
To view the dial plan in use, run the following command: asterisk*CLI> dialplan show Configuring the X-Lite ClientYou can use X-Lite as a SIP soft client to connect to Asterisk. To install X-Lite, download the free version of X-Lite from http://counterpath.com/x-lite.html. After you install it, run X-Lite and configure it by right-clicking the main panel to reach the menu. Click SIP Account Settings and fill in the relevant details of the user account with extension (for example, 1000) defined in the Asterisk sip.conf file (that is, the sip.conf file defined earlier in this chapter). The only other piece of information you need to add in the X-Lite client configuration is the IP address of your Asterisk server. In Bob’s example, his Asterisk server’s address is “192.168.100.10”. 
Figure 8. Bob’s configuration of the X-Lite client After you configure it, X-Lite automatically attempts to log on the user to Asterisk. Troubleshooting AsteriskDuring his research, Bob learned some helpful tips for trying to diagnose and troubleshoot issues. To connect to the console of the Asterisk server, run the following command from the Linux prompt, where the number of v characters sets the verbose level and the r characters indicate a remote Asterisk console: asterisk –vvvvvvr To get help, run the following command: asterisk*CLI> core show help After you are connected to the Asterisk console, calls being processed will scroll up in the console in real time (that is, because the verbose level is set to 6). Individual instructions from the dial plan are shown in the console as they are processed in sequence and calls are routed. This is can help you to determine whether calls are actually reaching the Asterisk server and whether they are being processed as expected. The following command shows the SIP users and devices configured in Asterisk: asterisk*CLI> sip show peers You see results similar to the following: Name/username Host Dyn Nat ACL Port Status 1000/1000 192.168.100.40 D 40108 Unmonitored to_ocs 192.168.100.20 5060 OK (1 ms)
To turn on the SIP debug mode, use the following command: asterisk*CLI> sip set debug on To debug only a single SIP user or device (called a peer), use the following command: asterisk*CLI> sip set debug peer [peer-name] To debug the Mediation Server only, use the following command to identify the Mediation Server as to_lync in the sip.conf file: asterisk*CLI> sip set debug peer to_lync To turn off SIP debugging, use the following command: asterisk*CLI> sip set debug off Yüklə 207,34 Kb. Dostları ilə paylaş:
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