Asterisk 16 documentation

Asterisk 16 documentation. At least a priority is required as an argument, or the goto will return a '-1',and the channel and call will be terminated. VoIP Gateways. Built-in configuration documentation for each module (that has documentation) can be How do you create a data store? Use ast_datastore_alloc function to return a pre-allocated structure. d - Dynamically add conference. D - Dynamically add conference, prompting for a PIN. The dialplan is essentially a scripting language specific to Asterisk and one of the primary ways of instructing Asterisk on how to behave. timezone - timezone, see /usr/share/zoneinfo for a list. The values will be stored encoded within Asterisk, but all consumers of the presence state (e. name - The property to set on the CDR. 2. Defaults to now. header - Include header information in the result (boolean) httpheader - Add HTTP header. same => n,ConfBridge(1) This example shows how to use a predefined user profile in confbridge. 9 Documentation ; Certified Asterisk 20. Certified Asterisk 18. See voicemail. Modules. Use of batch mode may result in data loss after unsafe asterisk termination, i. The default timeout is 5 seconds. Database commands on the CLI¶ Sub-commands under the command "database" allow a variety of functions to be performed on or with the database. isolation - Controls the data isolation on uncommitted transactions. conf. The release of Asterisk 16. This modularity gives you an almost unlimited amount of flexibility in the design of an Asterisk-based system. Defaults to 'ABdY "digits/at" IMp'. Research the new minor version you intend to update to. For instance, a channel protocol native transfer is external. The TONE_DETECT function detects a single-frequency tone and keeps track of how many times the tone has been detected. Example 3: A variable used internally by Asterisk. IsExternal - Indicates if the transfer was performed outside of Asterisk. You may also dynamically add SIP and IAX devices and extensions and making them available without Asterisk 20 Documentation. Note. Module Configuration. Standard. written in the C Programming Language. Warning. the SIP presence event package) will receive decoded values. Default is evaluate expression every 50 milliseconds with no timeout. Here we make an admin/marked user out of the 'my_user' profile that you define in confbridge. On Read - Retrieves unencoded message/subtype in Base64 encoded form. This release is available for immediate download at. We'll leave the default settings that are shipped with queues. Those default values get overwritten when the calling AMD with parameters. SQLite 3 creates a journal file in the 'astdbdir' specified in asterisk. OMIT - This CDR should be ignored. Verify that there is not a ' noload' line for the module that is failing to load. Asterisk 16 Documentation ; Asterisk 18 Documentation ; Asterisk 19 Documentation ; Asterisk 20 Documentation ; Asterisk 21 Documentation ; Certified Asterisk 18. so. exten => 1000,Set(SIP_CODEC=g729) same => n,Dial(SIP/1000,15) SIP_CODEC is set in the dialplan, but it gets evaluated inside of Asterisk, so the evaluation is case-sensitive. community and would have not been possible without your participation. conf, or changing the 'astdbdir' option to a directory for which the user running Versions of Asterisk. This involves either modifying the permissions of the 'astdbdir' directory listed in asterisk. AMI Actions¶. The labels are specified with the same syntax as used within the This documentation was generated from Asterisk branch 16 using version GIT Back to top Content is licensed under a Creative Commons Attribution-ShareAlike 3. Default is 'false'. There are two different types of Asterisk releases: Long Term Support and Standard. options. The labels are specified with the same syntax as used within the The body of the message that will be sent is what is currently set to 'MESSAGE (body)'. This would create a feature called 'eggs' that could be invoked during a call by pressing the '*5'. It is important that this directory is writable by the user Asterisk runs as. Performing Upgrades. That would tell Asterisk to not load chan_sip. The type of release defines how long it will be supported. conf file, extensions. Local/101@mycontext/nj. Upgrading to Asterisk 20 Asterisk 16 Documentation . n - Do not play announcement to caller (alters 'A (x)' behavior) timeout - Specify the length of time that the system will attempt to connect a call. same => n,Set(CONFBRIDGE(user,template)=my_user) same => n,Set(CONFBRIDGE(user,admin)=yes) This documentation was generated from Asterisk branch 16 using version GIT Back to top Content is licensed under a Creative Commons Attribution-ShareAlike 3. 7 Documentation ; Test Suite Documentation ; Historical Documentation Process1. Because AMI event documentation must be pulled from a variety of locations in the Asterisk Asterisk is…. AGI Commands¶. SUCCESS - Specified command successfully executed. If the command fails, the console should report a fallthrough. Now that you know a bit about Asterisk and how it is used, it's time to get you up and running with your own Asterisk installation. BILLING - This CDR contains valid billing data. FreePBX makes it easier to build a custom phone system to fit your needs with its feature-rich core and many available modules and add-ons. Verify that autoload=yes is enabled if you are intending to load modules from the Asterisk modules directory automatically. # Minification can reduce the space required to host the full # site by about 30% but it does take over double the time to # generate the site. After this application completes, the pbx engine will continue dialplan execution at the specified location in the dialplan. Be sure you have a backup of any essential data on the system. ChannelId - Channel UniqueId to be set on the channel. There are various ways to get started with Asterisk on your own system: Install FreePBX, the Asterisk-based distribution. If you determine one of those changes will be beneficial for you, only then proceed with an update. If the user does not type an extension in this amount of time, control will pass to the t extension if it exists, and if not the call would be terminated. 16. You can force a reload over the AMI, Asterisk Manager Interface or by calling Asterisk from a shell script with. Generated Version¶ This documentation was generated from Asterisk branch 16 using version GIT The functions and applications for Asterisk 11 are linked above, but you should look at the documentation for the version you have deployed. Return '1' on regular expression match or '0' otherwise. Solution. Multiple calls add multiple headers. Each module that you load Queue Logs. org/pub/telephony/asterisk. C - Continue in dialplan when kicked out of conference. conf files. https://downloads. API Documentation . BRANCHES := 16,18,19,20 # If you don't want to build the static documentation at all # NO_STATIC=yes # If you don't want the resulting HTML minified, set NO_MINIFY. There are a few items to check. The technology chosen for sending the message is determined based on a prefix to the 'destination' parameter. If the variable name is prefixed with '__', the variable will be inherited into channels created Query parameters. . Write-Only. Defaults to the database setting in res_odbc. AMI Events¶. You can add more than one modifier by adding them directly adjacent to the previous modifier. Asterisk 20 Documentation . These items are foundational, as knowing how to install Asterisk right the first time and where to locate the right help resources This documentation was generated from Asterisk branch 16 using version GIT Back to top Content is licensed under a Creative Commons Attribution-ShareAlike 3. Note that this extends the functionality available in the HANGUPCAUSE channel variable, by allowing This documentation was generated from Asterisk branch 16 using version GIT Back to top Content is licensed under a Creative Commons Attribution-ShareAlike 3. May be one of the following: 'read_committed', 'read_uncommitted', 'repeatable_read', or 'serializable'. Lets create those queues now in queues. Any strings matching '^ {X}' will be unescaped to X. sample in the [general] section of queues. If a transaction ID is specified as an optional argument, it will be applied to that followlocation - Whether or not to follow HTTP 3xx redirects (boolean) ftptext - For FTP URIs, force a text transfer (boolean) ftptimeout - For FTP URIs, number of seconds to wait for a server response. c - Announce user (s) count on joining a conference. API Documentation¶. When learning Asterisk it is important to start off on the right foot, so this section of the wiki covers orientation for learning Asterisk as well as installation and a simple Hello World style tutorial. This application will set the current context, extension, and priority in the channel structure based on the evaluation of the given condition. If provided, the applications listed will be subscribed to all events, effectively disabling the application specific subscriptions. e. 0. It is primarily intended for use with analog lines, but could be useful for other channels as well. This application originates an outbound call and connects it to a specified extension or application. All variables will be evaluated at the time MixMonitor is called. Within a given release series that is fully supported, bug fix updates are provided roughly every 4 to 6 weeks. conf as a template for a dynamic profile. Configuration Option Descriptions. 21. New releases of Asterisk will be made roughly once a year, alternating between standard and LTS releases. Allows comma separated values. This may he come from an incoming message. Download the new version and install Asterisk. As an Asterisk administrator, you have the choice on which modules to load and the configuration of each module. The pages in this section will describe what the elements of dialplan are and how to use BridgeVideoSource - If there is a video source for the bridge, the unique ID of the channel that is the video source. This argument can take any value. The following will create a console and set the VERBOSE message level to 2: 1. Supported options are those fields on the aor object in pjsip. 7 Documentation. This documentation was generated from Asterisk branch 16 using version GIT Back to top Content is licensed under a Creative Commons Attribution-ShareAlike 3. Consisting of multiple tracks, sessions, and EXPO hall, AstriCon offers various levels of education sessions and provides attendees networking opportunities with some of the best in the open source community. same => n,WaitForCondition(#,#["#{condition}"="1"],40,0. AGI Commands ; AMI Actions ; AMI Events ; Asterisk REST Asterisk 21 Documentation. Back to top. connecting many different Telephony protocols. unixtime - time, in seconds since Jan 1, 1970. 7 Documentation ; This documentation was generated from Asterisk branch 16 using version GIT Back to top Content is licensed under a Creative Commons Attribution-ShareAlike 3. The Asterisk dialplan. The dialplan script told Asterisk which Overview. s=silence - The number of seconds of silence that are permitted before the recording is terminated, regardless of the escape_digits or timeout arguments This documentation was generated from Asterisk branch 16 using version GIT Back to top Content is licensed under a Creative Commons Attribution-ShareAlike 3. default_expiration - Default expiration time in seconds for contacts that are dynamically bound to an AoR. app: string - (required) Applications to subscribe to. UserField - A user defined field set on the channels. This release is available for immediate download at https://downloads. AMAFlags - A flag that informs a billing system how to treat the CDR. Each item listed here is a comma-separated list of parameters that determine how a feature may be invoked during a call. a - Set admin mode. Please note that the space following the double quotes separating the regex from the data is optional and if present, is skipped. For a list of available options, see the documentation for the mixmonitor application. This application attempts to detect answering machines at the beginning of outbound calls. Defaults to machine default. x. For each -v specified, Asterisk will increase the level of VERBOSE messages by 1. Codecs - Comma-separated list of codecs to use for this call. You will almost certainly need other firewall rules for other forward-facing services (HTTP/HTTPS), which you will probably want to limit to your IP addresses. disable - Setting to 1 will disable CDRs for this channel. conf or 'read_committed' if not specified. Do not use untrusted strings such as CALLERID (num) or CALLERID (name) as part of the The Asterisk External Application Protocol (AEAP) is used to communicate configuration, data, and other information using a simple request/response messaging system. It ties everything together, allowing you to route and manipulate calls in a programmatic way. batch. # asterisk -c -v -v. Historical Documentation. If a space is desired at the beginning of the data, then put two spaces there; the second will not be skipped. offset_samples - Causes the recording to first seek to the specified offset before recording begins. name - The name of the AOR to query. Setting to 0 will enable CDRs for this channel. OtherChannelId - Channel UniqueId to be set on the second local channel. Attach data to pre-allocated structure. Because AMI event documentation is handled in a slightly different fashion, a new build option 'make full' is required to generate the documentation from the Asterisk source. FAILURE - Could not execute the specified command. Using 'rx' for audio received and 'tx' for audio transmitted to the channel. Currently, JSON is the only supported message description format. Mar 12, 2024 · AstriCon is the longest-running open source convention celebrating open source projects featuring Asterisk and FreePBX. format - a format the time is to be said in. At that point, this application will exit with the status variable set and dialplan processing will continue. DOCUMENTATION - This CDR is for documentation purposes. This allows a dialplan writer to determine, for each channel, who hung up and for what reason (s). Ex: datastore->data = mysillydata; Add datastore to the channel. Standard releases are supported for a shorter period of time Asterisk 16 Documentation ; Asterisk 18 Documentation . This application will block until the outgoing call fails or gets answered, unless the async option is used. Jan 14, 2010 · The next step is to add a couple of queues to Asterisk that we can assign queue members into. Control of the calls that passed through it was done through a special . Command - Will be executed when the recording is over. A - Set marked mode. When Asterisk was first created back in 1999, its design was focussed on being a stand-alone Private Branch eXchange (PBX) that you could configure via static . When loaded, AMD reads amd. If the location that is put into the channel information is bogus, and asterisk cannot find that location in the dialplan, then the execution engine will try to find and execute the code in the 'i' (invalid) extension in The goal here is to open SIP ports to the world and to open RTP (Realtime Transport Protocol) to the world on ports 10000-20000 as recommended by the Asterisk documentation. From an architectural standpoint, Asterisk is made up of many different modules. conf and uses the parameters specified as default values. This function can be used to set the value of channel variables or dialplan functions. asterisk -rx "reload". Asterisk 20 Documentation. Building AMI Event documentation for Asterisk requires both libxml and python. In order to properly manage ACD queues, it is important to be able to keep track of details of call setups and teardowns in much greater detail than traditional call detail records provide. 9 Documentation. That is, if we had a line as follows: noload => chan_sip. Ex: ast_channel_datastore_add (chan, datastore); This function takes two arguments: (pointer to channel, pointer to data store) Full Example: This documentation was generated from Asterisk branch 16 using version GIT Back to top Content is licensed under a Creative Commons Attribution-ShareAlike 3. Asterisk 19 Documentation. response: The maximum amount of time permitted after falling through a series of priorities for a channel in which the user may begin typing an extension. Arguments. Versions of Asterisk. When setting variables, if the variable name is prefixed with '_', the variable will be inherited into channels created from the current channel. Yes. In this example, the user wishes to suggest to the SIP channel driver what codec to use on the call. Define the CDR batch mode, where instead of posting the CDR at the end of every call, the data will be stored in a buffer to help alleviate load on the asterisk server. beep - Causes Asterisk to play a beep as recording begins. Result of execution is returned in the SYSTEMSTATUS channel variable: SYSTEMSTATUS. Command line parameters can be combined. , software crash, power failure, kill -9, etc. 0 United States License. Async - Set to 'true' for fast origination. Test Suite Documentation. The Asterisk Development Team would like to announce the release of Asterisk 16. Example: eggs = *5,self,Playback (hello-world),default. a toolkit for building many things: an IP PBX with many powerful features and applications. Made with Material for MkDocs. Simply call this application after the call has been answered (outbound only, of course). running on Linux (or other types of Unix ) powering Business Telephone Systems. x required - The announcement to playback to all devices. Other than what is covered under Core Configuration, most features and functionality are provided by modules that you may or may not have installed in your Asterisk system. PreDialGoSub - PreDialGoSub Context,Extension,Priority to set options/headers needed before start the outgoing Description. Content is licensed under a Creative Commons Attribution-ShareAlike 3. Asterisk 21 Documentation ; Certified Asterisk 18. When using this function you set a target audio level. The previous command can also be invoked in the following way: 1. asterisk. subscribeAll: boolean - Subscribe to all Asterisk events. 2025-10-18. agi'. conf, known as the "dialplan". The realtime Architecture lets you store all of your configuration in databases and reload it whenever you want. Executes a command by using system (). After this duration, any page calls that have not been answered will be hung up by Arguments. The Hangup Cause family of functions and dialplan applications allow for inspection of the hangup cause codes for each channel involved in a call. Certified Asterisk 20. Asterisk 21 Documentation. A DTMF transfer is internal. The modifiers are added to a channel by adding a slash followed by a flag onto the end of the Local Channel dial-string. Asterisk 16 Documentation ; Asterisk 18 Documentation ; Asterisk 19 Documentation ; Asterisk 20 Documentation . 0 resolves several issues reported by the. This takes care of installing Linux, Asterisk, and a web-based management Waits until expression evaluates to true, checking every interval seconds for up to timeout. When reading this function (instead of writing), supply 'tx' to get the number of times a tone has been detected in the TX direction and 'rx' to get the number of times a tone has been detected in the RX direction. contact - Permanent contacts assigned to AoR. At present, the following request/response messages are supported: setup - Initializes a remote application. Fully Supported. This section contains many sub-sections on configuring every aspect of Asterisk. field - The configuration option for the AOR to query for. Mar 18, 2024 · FreePBX is the #1 open source graphical user interface (GUI) for use with Asterisk. 0 resolves several issues reported by the community and would have not been possible without your participation. Overview. 4. Local/101@mycontext/n. # asterisk -cvv. This application sets the following channel variables: MESSAGE_SEND_STATUS - This is the message delivery This documentation was generated from Asterisk branch 16 using version GIT Back to top Content is licensed under a Creative Commons Attribution-ShareAlike 3. 2023-10-18. 3. UniqueID - A unique identifier for the Party A channel. b - Run AGI script specified in MEETME_AGI_BACKGROUND Default: 'conf-background. For example below we are adding the "n" modifier to the dial-string. Upgrading to Asterisk 16 ; New in 16 ; API Documentation . In order to support this, extensive and detailed tracing of every queued call is stored in the queue log, located (by default) in /var/log Asterisk 16 Documentation ; Asterisk 18 Documentation ; Asterisk 19 Documentation ; Asterisk 20 Documentation ; Asterisk 21 Documentation ; Certified Asterisk 18. 2026-10-18. an Open Source software development project. g. 7 Documentation ; Test Suite Documentation ; Historical Documentation Asterisk 16 Documentation ; Asterisk 18 Documentation ; Asterisk 19 Documentation ; Asterisk 20 Documentation ; Asterisk 21 Documentation ; Certified Asterisk 18. 7 Documentation ; Test Suite Documentation ; Historical Documentation This documentation was generated from Asterisk branch 16 using version GIT Back to top Content is licensed under a Creative Commons Attribution-ShareAlike 3. May be negative. party_a - Set this channel as the preferred Party A when channels are associated together. For now we'll work with two queues; sales and support. 5) This documentation was generated from Asterisk branch 16 using version GIT Back to top Content is licensed under a Creative Commons Attribution-ShareAlike 3. A Long Term Support release is fully supported for 4 years, with an additional year of maintenance for security fixes. 19. The AGC function will apply automatic gain control to the audio on the channel that it is executed on. Description. Example: Wait for condition dialplan variable/function to become 1 for up to 40 seconds, checking every 500ms. A(x) - Play an announcement to all paged participants. yx yt fs dn et nc ov lg ou tf