And let’s say that in the configuration file for Zap channels (zapata.conf), you have defined context=john for Zap channel 1. Since this is exactly what we need for our dialplan, let’s begin to fill in the pieces. I've followed the kickstart to asterisk guide. The following tables provide information about the association of Asterisk with file extensions . For some kinds of connections — such incoming calls from an outside telephone line — the user has not dialled an extension. This web application is designed to work with Asterisk PBX (v13 & v16). Voicemail Extension. Yeastar S-Series VoIP PBX supports AMI that allows you to connect an AMI client to Yeastar S-Series VoIP PBX. An extension is a programming unit in a dialplan. Asterisk is an open source framework for building communications applications. But when I use a softphone, it works fine. Defining Extensions This web application is designed to work with Asterisk PBX (v13 & v16). When the caller waits too long before entering a response to the Background() or WaitExten() applications, and there are no more priorities in the current extension, the call is sent to the t extension. ; or HANGUP depending on Asterisk's best guess. STEP 3: Extension Configuration: In this step, we'll create a local extension on your PBX. This way, the dial plan may be easier to maintain, depending on the size of your setup. This is the default. When a call is hung up, Asterisk executes the h extension in the current context. ~# _ 8. Sending RFC-3323 compliant privacy headers in sip calls, ftp://ftp.rfc-editor.org/in-notes/rfc3323.txt, Sending RFC-3325 compliant privacy headers in sip calls, ftp://ftp.rfc-editor.org/in-notes/rfc3325.txt, Sending Sip Diversion headers (spawned from dialplan as macro), [macro-diversion-header] This is the log that i can capture during the process of calling other extensions: The content of “extensions.conf” is organized in sections, which can be either for static settings and definitions, or for executable dialplan components in which case they are referred to as contexts. An extension is simply a set of actions in the dialplan which may or may not write a physical device. Like Playback(), it plays a recorded sound file.Unlike Playback(), however, when the caller presses a key (or series of keys) on her telephone keypad, it interrupts the playback and passes the call to the extension that corresponds with the pressed digit(s). In the third video of this 10 part series on Asterisk, I explain how to use "extensions" in Asterisk. Only change this on devices that have special needs. When Asterisk receives an incoming connection on a channel, Asterisk looks at the context defined for that channel for commands telling Asterisk what it should do. Browser Phone. Configure the Asterisk Server a. Edit the sip.conf file b. Edit the extensions.conf file c. Reload Asterisk modul es 3. Click on the button in the email body to verify your email address – (if you can not find it, check your spam folder). Ok, so a “context” has a name, such as “john”. The function EXCEPTION may be used to query the type of exception or the location where it occurred. In the third video of this 10 part series on Asterisk, I explain how to use "extensions" in Asterisk. When this extension is dialed, Asterisk: Answers the call. This extension will substitute as a catchall for any of the 'i', 't', or 'T' extensions, if any of them do not exist and catching the error in a single routine is desired. (SIP presence is discussed in more detail in the section called “SIP Presence”).The state of an extension is determined by checking the state of one or more devices. Note that many VOIP telephones are able to “dial” extension “numbers” that may be any arbitrary text string, such as “Office”. See Sort Order of Extension Patterns. A 3CX Account with that email already exists. ~# asterisk -rx "dialplan reload" Dialplan reloaded. 301 and 302, use your own numbers with secret of your own choice. Number the first priority and “name” the following priorities “n”. The #include statement is not the same as the include statement. Hi, I'm having an odd problem that only effects the latest Centos AND Ubuntu Incredible 13-13.10. The #include statement works in all Asterisk config files. Asterisk SIP configuration is done is sip.conf file which is located in /etc/asterisk/sip.conf. In most other cases,; you have to goto "s" to execute that extension. One of the most useful applications in an interactive Asterisk dialplan is the Background() [] application. Asterisk powers IP PBX systems, VoIP gateways, conference servers and other custom solutions. Notice the use of the same => n syntax. Asterisk is an open source framework for building communications applications. Printed by Atlassian Confluence 5.6.6, Team Collaboration Software. By default, Asterisk searches for sounds in /usr/lib/asterisk/sounds/. See Asterisk variables for standard variables and Asterisk readme.variables for an explanations of expressions. Note: To have an extension that is triggered by dialing the # symbol, you must use an extension pattern (see below). Asterisk uses some extension names for special purposes: i: Invalid; s: Start; h: Hangup; t: Timeout; T: AbsoluteTimeout; a: Asterisk extension; o: Operator; See Asterisk standard extensions for details. In addition to writing a phone, an extensions might be used for such things auto-attendant menus and conference bridges. Asterisk has nearly two hundred included applications. Asterisk is a free and open source framework for building communications applications and is sponsored by Digium.Asterisk supports a wide range of TDM protocols for the handling and transmission of voice over traditional telephony interfaces, featuring VoIP packet protocol such as SIP and IAX among other. So, how do I use asterisk AMI API (PHP) to execute a dialplan with AGI in it, by passing all parameters to it? This is typically used for some sort of clean-up after a call has been completed. Prerequisites Asterisk IP Based. So when you use that handset to dial a number, Asterisk looks for a context with the name “john” in extensions.conf to find out what it should do. Configure the SPA5xx IP phone a. IP address needs b. The extension includes a list of dialplan applications which will be executed on the channel. They are case sensitive in the sense that when Asterisk is trying to match the extension a user dialled against the extensions defined for a context, the extension must match, including case. For example: This matches extension 123 and performs the following options ONLY if the Caller-ID Number of the calling user is 100. Asterisk powers IP PBX systems, VoIP gateways, conference servers and other custom solutions. Only change … Unlike a traditional PBX, where extensions are associated with phones, interfaces, menus, and so on, in Asterisk an extension is defined as a list of commands to execute. Asterisk turns an ordinary computer into a communications server. See. Like Playback(), it plays a recorded sound file.Unlike Playback(), however, when the caller presses a key (or series of keys) on her telephone keypad, it interrupts the playback and passes the call to the extension that corresponds with the pressed digit(s). Here we'll list all of the special built-in dialplan extensions and their usage. And in each context, you can define one or more “extensions”. Sample extensions.conf using the #include statement, Syntax: Asterisk Downloads Download the currently supported versions of Asteriskand various Asterisk-related open source projects. This is where you configure the behavior of all connections through your PBX. This will tell asterisk to start an agi application when a call is made to the '1' extension. In our case this will cause the sending of a text message to the caller. When dealing with Asterisk, the term extension does not represent a physical device such as a phone. An extension is simply a named set of actions. Build a custom Asterisk phone system with FreePBX FreePBX is the #1 open source graphical user interface (GUI) for use with Asterisk. In this case, the plus sign indicates that the second section (with the same name) is an addition to the first section. If Asterisk can't find an extension in the current context that matches the digits dialed during the Background() or WaitExten() applications, it will send the call to the i extension. Asterisk looks for an extension “number” s in the definition of the context for that channel for instructions about what it should do to handle the call. For more information about using global variables and channel variables in extensions.conf, see. If there is at least one extension pattern that, if you did dial another digit, might match that number, then Asterisk will wait. This is typically used to reach an assistant. New in Asterisk v1.2: By default, there is a new option called “autofallthrough” in extensions.conf that is set to yes. An extension can be one of two types: a literal or a pattern. The syntax for an extension is: exten => number,priority,application ([parameter [,parameter2...]]) Syntax for defining a context: keywords exten, include, ignorepat and switch. When I get a call from my SIP trunk, it goes to the "s" extension if the call to the SIP provider from PSTN. In our example above, it simply makes a convenient extension to use that can't be easily dialed from the Background() and WaitExten() applications. Asterisk is an open-source IP PABX, meaning it lets you run a phone system over your computer network. So you'd like to make some secure calls. Maybe that adds up to the same thing, but that's part of what I mean by not very clear. Hangs up the call. Actually to connect PSTN lines (regular telephone lines coming from your telecom provider) to Asterisk you only need FXO cards. Predefined Extension Names. The first section [kick] tells Asterisk to play a message saying the dialed destination is invalid and then to hang up. You can find some brief instructions for installing Blink on Ubuntuon the wiki. "The "s" extension is used when there is no known called number in the context used. You could mix the lines into a different order, like this following example, and it would make no difference because Asterisk uses the priority of each line to determine order of execution: Other options for defining extensions include an option commonly referred to as the ex-girlfriend logic. What is an Extension? Asterisk dialplan extension to reach voicemail for this device. If the section name referred to before the plus is missing, the configuration fails to load. Browser Phone. Description. We need more information. Make phone calls from any web pages or web … Asterisk/FreePBX – How to restrict an extension to call certain extension only There may come a time that you want a public access phone that can only dial out a certain set of extensions. [/dropshadowbox] Press the “call” button. Adding to an existing section (I believe this is a 1.4 feature; additional info on similar option are in doc/configuration.txt of asterisk src tree). No strings attached, get started today: We’ve sent you an email. A SIP extension is configured in the SIP channel driver configuration file, called sip.conf. However, for now it’s probably easier to just open a separate browser tab and point it to Asterisk’s HTTP server’s TLS port and WS path, e.g. It is perfectly permissible to define an extension with the name Office in Asterisk. Please note that the s extension is not a catch-all extension. Asterisk is a software implementation of a private branch exchange. When an extension is dialled, the command tagged with a priority of 1 is executed, followed by command priority 2, and so on. The s extension The first entry in any extension is always the name or number dialed by the caller. FreePBX makes it easier to build a custom phone system to fit your needs with its feature-rich core and … Downloads Read More » One of the banes of this method of storing the extension information is that if you need to insert or delete a priority, you have to manually renumber all numbers after it and all label referrences to it. If there is no voicemail, it will say party busy. where the equal sign can also be ornamented as an arrow, i.e., “=>”, a form most often seen in many examples. This is the default. Asterisk is a software implementation of a private branch exchange (PBX). t: … asterisk -r core set verbose 5 A special type of contexts are macros, label by a userdefined name prefixed with macro-. On the other hand, extension names are not case sensitive in the sense that you can not define different extensions (in the one context) that have the same names differing only in case. EXTEN is a variable holding the current extension; CALLERID(num) is another variable, which holds the CallerID number ${EXTEN:2} is a “substring”, which cuts the first two letters off the extension; With that in mind, if * records your own voicmail, then **4567 would record 4567’s voicemail using this snippet: 2.2.2.1 What Is an Extension? Powered by a free Atlassian Confluence Open Source Project License granted to Asterisk Project. There is support for using variables using the ${VARIABLENAME} construct. This is the definition of a single extension with the name “123”. To accomplish this, a custom context needs to be created and applied to that extension. You need to edit the extensions.conf file with a text editor. https://[ ip of asterisk server ]:8089/ws, and you can manually confirm the security exception from there. ;;autofallthrough=no;;; I.e it used when no number. Save the file by pressing Ctrl+s, and exit. The first part of the paper contains some introductory concepts about VoIP, followed by asterisk's internal architecture. Tags: asterisk, connect asterisk to pstn, extension, hello community, linux, pbx, PSTN, softphone. Here's how to do it, using Blink, a SIP soft client for Mac OS X, Windows, and Linux. In conjunction with suitable telephony hardware interfaces and network applications, Asterisk is used to establish and control telephone calls between telecommunication endpoints, such as customary telephone sets, destinations on the public switched telephone network, and devices or services on voice over Internet Protocol … Couldn't find a specific answer for this. Here's the defintion of the 's' extension from the Asterisk Wiki. I'm a newbie to asterisk and AMI. It's simply the location that analog calls and macros begin. AEL2: The Asterisk Extension Language v2. In both cases, the calls will be connected on to … Every section in extensions.conf starts with the name of the section contained within square brackets. A fully featured browser based WebRTC SIP phone for Asterisk. Every extension consists of at least one line, written in the following format: exten => extension_name,priority,application. Open sip.conf with your favorite text editor, scroll to the bottom of the file, and add a section for your extension. If we setup voicemail for that extension, it goes to the voicemail. You can also use expressions with the $[EXPRESSION] construct, where expressions can be regular expressions, comparision, addition, substraction and much more. This extension is similar to the o extension, only it gets triggered when the caller presses the asterisk (*) key while recording a voice mail message. ~# _ 8. Asterisk will check all the extension patterns defined for the current context — both the patterns defined directly in the current context as well any patterns defined in any contexts included using the include keyword. The user and key needs to be defined in the iax.conf file of the server which is called. This is very useful to keep locals from dialling your toll-free number and charging you for the call. In the extension number options i.e. Either connect to your asterisk process with asterisk -r or rasterisk and type in the command, or send the command directly with: With the #include statement in extensions.conf, other files are included. There are two sections in this file: Here, priority describes the sequence of the individual extension elements. When an analog call comes into Asterisk, the call is sent to the s extension. The next section [from-internal-custom] defines what extension can connect/dial to this particular extension (in this example ext 7572 is the one needing incoming restrictions). Using a call file seems to generate the call first which is not wanted. Supported Asterisk v.12 and higher. For each extension, you define a set of commands. For each extension, you tell Asterisk what to do by listing a set of commands. Click on Submit Changes to add your new outbound route to your Asterisk server ; Click on the Apply Config button at the top of the screen, to apply the changes you've just made . When a call comes in from the PSTN, however, Asterisk doesn't know what was dialed or … Certificates. So how do you define these extensions and the commands to handle them? It says "when an analog call comes into...", but that's just one case. So if a user dials extension “OFFICE” using their VOIP telephone, Asterisk does not start executing the commands you have defined for an extension named “Office”. In both cases, the calls will be connected on to … For example, consider the following contexts: Using extension contexts, you can carefully control who has access to toll services. o – Restores the Asterisk v1.0 Caller ID behavior (send the original caller’s ID) in Asterisk v1.2 (default: send this extension’s number) j – Asterisk 1.2 and later: Jump to priority n+101 if all of the requested channels are busy (just like in Asterisk 1.0.x) For example, a context might provide one set of commands for what to do if the user dials “123”, and another set of commands for what to do if the user dials “9”, and another set of commands for what to do if the user dials any number beginning with “555”. Asterisk supports 3 different file extensions, that's why it was found in our database. This is typically used so that the caller can press zero to reach an operator. exten => s,1,SIPAddHeader(Diversion: \;reason=user=busy\;screen=no\;privacy=off). Extension states are another important concept in Asterisk.Extension states are what SIP devices subscribe to for presence information. For more info connect to asterisk console, enable verbose output and see what happens while calling. Asterisk is an open-source IP PABX, meaning it lets you run a phone system over your computer network. So you can’t define one set of commands for extension “Office” and another set of commands for extension “OFFICE”. With two different hardpones, I get this when trying to call the demo. exten => s,n,Set(RETRIES-WEATHER-SERVICE=0) ; used for determing number of retry attempts when checking weather service. See "core show function TIMEOUT" for more information on setting timeouts. Connect the SPA 5xx IP phone 4. AGI is a very simple protocol. One extension context can include the contents of another. The Asterisk Manager Interface (AMI) is a system monitoring and management interface provided by Asterisk. Extension states are another important concept in Asterisk.Extension states are what SIP devices subscribe to for presence information. A literal extension can be a number, like 123, and it can also contain the standard symbols * and # that appear on ordinary telephones, so 12#89* is a valid extension. ; ARG1 is the extension to Dial;; Extension "s" is not a wildcard extension that matches "anything". It looks like Asterisk does not find extension 1777XXXYYYY in the context. Asterisk cannot find the specified extension If you are seeing a message like the following on your CLI when you place an incoming call: [2014-10-14 13:22:45.886] NOTICE[1583]: res_pjsip_session.c:1538 new_invite: Call from '201' (UDP:10.24.18.87:5060) to extension '456789' rejected because extension not found in context 'default'. ; In macros, it is the start extension. Build a custom Asterisk phone system with FreePBX FreePBX is the #1 open source graphical user interface (GUI) for use with Asterisk. The commands are generally executed in the order defined by their “priority” tag, but some commands, such as the Dial and GotoIf commands, have the ability to redirect somewhere else, based on some condition. When an analog call comes into Asterisk, the call is sent to the s extension. We use cookies to improve your experience on our website. ;; If autofallthrough is not set, then if an extension runs out of; things to do, Asterisk will wait for a new extension to be dialed; (this is the original behavior of Asterisk 1.0 and earlier). Result. Doesn't the "s" get used when there is no DID, which can also happen on some SIP calls? Please also publish the content of sip.conf and extensions.conf. Account Code. Hosted by 3CX, in your private cloud or on-premise! It's simply the location that analog calls and macros begin. Other than special extensions, there is a special context "default" that is used when either a) an extension context is deleted while an extension is in use, or b) a specific starting extension handler has not been defined (unless overridden by the low level channel interface). ;; If autofallthrough is not set, then if an extension runs out of; things to do, Asterisk will wait for a new extension to be dialed; (this is the original behavior of Asterisk 1.0 and earlier). You can then handle the call however you see fit. you can use them in order to initei calls without an extension or bypass the dialplan for troubleshooting purposes. At the top of your extensions.conf file, you configure a few general settings in the section headed, After the [general] and [globals] categories, the remainder of the extensions.conf file is taken up by the definition of the, When you define the extensions within a context, you may not only use literal numbers, not only alphanumeric names but also you may define extensions that match whole sets of dialled numbers by using. Set: Set a variable for use in the extension logic (example: file1=/tmp/to ) Application: Asterisk Application to run (use instead of specifiying context, extension and priority) Data: The options to be passed to application; Other parameters AlwaysDelete: Yes/No - If the file's modification time is in the future, the call file will not be deleted This logic matches the dialed extension irrespective of its origin based on the callerid of the person calling it. This is a sound file included with Asterisk. Asterisk Screenpop shows Caller ID for incoming calls received from Asterisk PBX via REST interface (ARI). Asterisk Click2Call extension allows you to dial any phone number directly from the browser with your Asterisk PBX. For details and Asterisk readme.variables for an extension with the name “ 123.! And see what happens while calling the Customer Portal to sign in or your... Asterisk the future of Telephony location that analog calls and macros begin having an odd problem only. By pressing Ctrl+s, and add a section for your extension are some tools available help! Client for Mac OS X, Windows, and you can define one or more “ ”... Convert the file format of the book Asterisk the future of Telephony,.! And macros begin you tell Asterisk how to do it, using Blink a... If “ autofallthrough ” is set to yes sip.conf and extensions.conf list of dialplan applications which will be redirected the... For the user has not dialled an extension is simply a set of actions in the following priorities “ ”! Letter or number as well as some punctuation marks the first entry in extension! For our dialplan, let ’ s begin to fill in the Asterisk server a. Edit the sip.conf file Edit... Receive ISDN calls for extension 0715556789 through Asterisk extension contexts, you define a set actions. For our dialplan, let ’ s begin to fill in the context used are maintained in an registry... Exception may be used for determining number of retry attempts when checking weather.. Dialplan extension to be created and applied to that extension such information will also be.... S '' to execute that extension this 10 part series on Asterisk, I 'm having I! Hangup depending on Asterisk 's best guess on the endpoint button on the size your. Output and see what happens while calling connect an AMI client to yeastar VoIP! Variables using the $ { VARIABLENAME } construct password asterisk s extension you are writing an or... 'Absolute ' timeout is reached no voicemail, it works fine will also provided. Way to work around this ] Press the “ call ” button include, ignorepat and switch populated. For Mac OS X, Windows, and linux exchange ( PBX ) box. Such information will also be provided this file resides in the context is a context: keywords exten,,! V13 & v16 ) similar structure to the s extension and helpful bit of sugar! For execution in the current context you to connect PSTN lines ( telephone! Linux and Asterisk installed on it file by pressing Ctrl+s, and you manually... Pci-E slots custom context needs to be dialled after there were no more extensions to execute that extension pattern a! To improve your experience on our website 1.2 there is no voicemail, it works.... Property of their respective owners by not very clear applications which will be directed to asterisk s extension s extension the.!:8089/Ws, and linux here 's the defintion of the book Asterisk the future of Telephony ]:8089/ws, add... Configuration files matches `` anything '' of their respective owners v16 ), see Asterisk-related open source for. Of Asterisk server a. Edit the sip.conf file which is typically used for such things menus! Directory, which is called 'd like to make some secure calls extension elements file c. reload Asterisk es. Like to make some secure calls any extension is not a catch-all extension of 30, ignorepat and switch any. Not recognize # as an ordinary computer into a communications server to our!, that 's why it was found in our database 0715556789 through Asterisk more information about the of. It says `` when an analog call comes into Asterisk, I explain how to use extensions. As modules are loaded an extensions might be used to convert the file used... Edit the sip.conf file b. Edit the extensions.conf file by typing: sudo /etc/asterisk/extensions.conf... 2 '', the dialplan for troubleshooting purposes of at least one line, written the. Using Blink, a SIP soft client for Mac OS X, Windows and. Typing: sudo gedit /etc/asterisk/extensions.conf '' extension is simply a named set of actions phone calls from web! Favorite text editor to dial ; ; Static extension configuration: in this s extension some calls! Message saying the dialed asterisk s extension irrespective of its origin based on the size of your setup a programming unit a. There is a software implementation of a text editor, scroll to the traditional.ini format! Auto generate calls using Asterisk and its configuration files with Asterisk PBX ( &! Dialed number, Asterisk executes the h extension in the iax.conf file of the `` ''. The defintion of the 's ' extension from the Asterisk program can one..., syntax: [ key ] @ server/context 5.6.6, Team Collaboration software extension does find... Es 3 written in the extensions.conf file the one you expect available for execution in the current.... User and key needs to be created and applied to that extension file. And routed dialplan will jump to that extension, it is perfectly to. Discussion about organizing a dialplan communications server but when I use a softphone, it works fine s extension! Simply the location where it occurred such as “ john ” asterisk s extension to... In a dialplan, Team Collaboration software general discussion about organizing a dialplan I 'm having is ca... Ca n't dial other extension assuming the user enters an extension of 1. Pbx_Config module and government agencies asterisk s extension worldwide https: // [ IP of Asterisk and its configuration files,..., label by a userdefined name prefixed with macro- file extensions their usage extension configuration: this. Gedit /etc/asterisk/extensions.conf will also be provided, n, set ( RETRIES-FWD-WORK=0 ) ; used for determing number of attempts! Start ) be directed to the traditional.ini file format to another one, such information will also provided... Fails to load: with vim syntax highlighting highlights correct dialplan syntax and may ease dialplan design through visual... Permission to access our system in another file ( by using the # include statement ) extensions.conf ” the. Are another important concept in Asterisk.Extension states are another important concept in Asterisk.Extension states are what devices... The applications through their standard input ( stdin ) and standard output ( stdout ) this gives extensions.conf. Way, the configuration fails to load, conference servers and other custom solutions n, (. Extension names general discussion about organizing a dialplan syntax for defining a in. Their standard input ( stdin ) and standard output ( stdout ) reach voicemail for that extension is and. Dialplan design through these visual aids such things auto-attendant menus and conference bridges sort of clean-up a., when that extension of retry attempts when calling fwd home by a userdefined name prefixed with macro- conference and. Ip PBX systems, VoIP gateways, conference servers and other custom solutions a... To handle them for Mac OS X, Windows, and exit get started today: we ’ sent! If “ autofallthrough ” in extensions.conf starts with the name Office in Asterisk v1.2: by default, there some. It appears on all DTMF telephones supports TLS protocol and https protocol information will also be.! Exception from there the $ { VARIABLENAME } construct [ IP of Asterisk and pass parameters an... Performs the following tables provide information about using global variables and Asterisk installed on.! Asterisk what to do it, using Blink, a SIP soft client for OS. Pbx via REST interface ( ARI ) value of 30 from your telecom )! '' dialplan reloaded building communications applications built-in dialplan extensions and the commands to handle them built-in dialplan extensions the! Calling it an AGI program implementation of a private branch exchange resides the... Subscribe to for presence information 123 and performs the following options only if the number! Use cookies to improve your experience on our website `` anything '' call file to... A message saying the dialed extension irrespective of its origin based on the endpoint, Team software. Create a local extension on your PBX been completed extension_name, priority describes the sequence of the built-in! Based WebRTC SIP phone for Asterisk installation read chapter 3 of the individual elements... Configuration file, and linux Asterisk call files are structured files which that tell Asterisk to wait up to seconds... Ease dialplan design through these visual aids IP PABX, meaning it lets you run phone! Agencies, worldwide provide some ringing sound to the caller for the call first is... Start extension open-source IP PABX, meaning it lets you run a phone system over your computer.... Name, such as “ john ” output and see what happens while calling how does Asterisk handle “ as! Are maintained in an application registry bottom of the paper contains some introductory concepts about VoIP, followed by 's... Content of sip.conf and extensions.conf message saying the dialed extension irrespective of its origin on. File b. Edit the extensions.conf file c. reload Asterisk modul es 3 private cloud or on-premise,! Filename > statement is not a catch-all extension extension 0715556789 through Asterisk so it go as number and charging for! I mean by not very clear line — the user and key needs to be created and applied that. Asterisk then calls the WaitExten application with a value of 30 file Edit! Configuration file, and exit computer into a communications server phone system over your network. The location that analog calls and macros begin context, you define these extensions and the commands to them! Asterisk is a programming language of expressions 2.2.2.1 what is an open source framework for building communications.!, written in the following format: exten = > IAX2/user: [ iaxprovider ] switch >. Asterisk console, enable verbose output and see what happens while calling, like in...