Asterisk 16 documentation

Asterisk 16 documentation. FAILURE - Could not execute the specified command. offset_samples - Causes the recording to first seek to the specified offset before recording begins. conf files. 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. options. This documentation was generated from Asterisk branch 16 using version GIT Back to top Content is licensed under a Creative Commons Attribution-ShareAlike 3. For each -v specified, Asterisk will increase the level of VERBOSE messages by 1. After this application completes, the pbx engine will continue dialplan execution at the specified location in the dialplan. This function can be used to set the value of channel variables or dialplan functions. The previous command can also be invoked in the following way: 1. timezone - timezone, see /usr/share/zoneinfo for a list. That is, if we had a line as follows: noload => chan_sip. PreDialGoSub - PreDialGoSub Context,Extension,Priority to set options/headers needed before start the outgoing Description. The following will create a console and set the VERBOSE message level to 2: 1. When setting variables, if the variable name is prefixed with '_', the variable will be inherited into channels created from the current channel. 0 United States License. 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. There are two different types of Asterisk releases: Long Term Support and Standard. . Write-Only. This release is available for immediate download at. This involves either modifying the permissions of the 'astdbdir' directory listed in asterisk. From an architectural standpoint, Asterisk is made up of many different modules. 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. Return '1' on regular expression match or '0' otherwise. 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. In this example, the user wishes to suggest to the SIP channel driver what codec to use on the call. D - Dynamically add conference, prompting for a PIN. community and would have not been possible without your participation. Asterisk 16 Documentation ; Asterisk 18 Documentation ; Asterisk 19 Documentation ; Asterisk 20 Documentation . API Documentation¶. Yes. This argument can take any value. May be negative. Example: eggs = *5,self,Playback (hello-world),default. Asterisk 20 Documentation. Within a given release series that is fully supported, bug fix updates are provided roughly every 4 to 6 weeks. Attach data to pre-allocated structure. 2. Default is 'false'. For a list of available options, see the documentation for the mixmonitor application. d - Dynamically add conference. Executes a command by using system (). 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. 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. Performing Upgrades. Allows comma separated values. conf, known as the "dialplan". The release of Asterisk 16. At present, the following request/response messages are supported: setup - Initializes a remote application. Research the new minor version you intend to update to. Verify that autoload=yes is enabled if you are intending to load modules from the Asterisk modules directory automatically. This may he come from an incoming message. 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. 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. Asterisk 19 Documentation. The values will be stored encoded within Asterisk, but all consumers of the presence state (e. Example: Wait for condition dialplan variable/function to become 1 for up to 40 seconds, checking every 500ms. 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. Because AMI event documentation must be pulled from a variety of locations in the Asterisk Asterisk is…. 21. Database commands on the CLI¶ Sub-commands under the command "database" allow a variety of functions to be performed on or with the database. Versions of Asterisk. default_expiration - Default expiration time in seconds for contacts that are dynamically bound to an AoR. contact - Permanent contacts assigned to AoR. Made with Material for MkDocs. Asterisk 21 Documentation ; Certified Asterisk 18. e. isolation - Controls the data isolation on uncommitted transactions. 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. header - Include header information in the result (boolean) httpheader - Add HTTP header. subscribeAll: boolean - Subscribe to all Asterisk events. 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. AMI Events¶. conf or 'read_committed' if not specified. A DTMF transfer is internal. Using 'rx' for audio received and 'tx' for audio transmitted to the channel. ChannelId - Channel UniqueId to be set on the channel. Mar 18, 2024 · FreePBX is the #1 open source graphical user interface (GUI) for use with Asterisk. app: string - (required) Applications to subscribe to. Historical Documentation. 0. You can force a reload over the AMI, Asterisk Manager Interface or by calling Asterisk from a shell script with. 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. Defaults to 'ABdY "digits/at" IMp'. https://downloads. Defaults to machine default. Configuration Option Descriptions. Content is licensed under a Creative Commons Attribution-ShareAlike 3. For example below we are adding the "n" modifier to the dial-string. If you determine one of those changes will be beneficial for you, only then proceed with an update. conf. Test Suite Documentation. Example 3: A variable used internally by Asterisk. 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. 0 resolves several issues reported by the community and would have not been possible without your participation. When using this function you set a target audio level. Codecs - Comma-separated list of codecs to use for this call. Command line parameters can be combined. # 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. 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. g. See voicemail. A(x) - Play an announcement to all paged participants. 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. 3. a toolkit for building many things: an IP PBX with many powerful features and applications. That would tell Asterisk to not load chan_sip. Default is evaluate expression every 50 milliseconds with no timeout. Download the new version and install Asterisk. 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. conf as a template for a dynamic profile. same => n,ConfBridge(1) This example shows how to use a predefined user profile in confbridge. same => n,WaitForCondition(#,#["#{condition}"="1"],40,0. party_a - Set this channel as the preferred Party A when channels are associated together. Lets create those queues now in queues. Warning. 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 . Local/101@mycontext/n. There are various ways to get started with Asterisk on your own system: Install FreePBX, the Asterisk-based distribution. # asterisk -cvv. AGI Commands¶. Result of execution is returned in the SYSTEMSTATUS channel variable: SYSTEMSTATUS. connecting many different Telephony protocols. org/pub/telephony/asterisk. Note. 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. The TONE_DETECT function detects a single-frequency tone and keeps track of how many times the tone has been detected. Each item listed here is a comma-separated list of parameters that determine how a feature may be invoked during a call. May be one of the following: 'read_committed', 'read_uncommitted', 'repeatable_read', or 'serializable'. The modifiers are added to a channel by adding a slash followed by a flag onto the end of the Local Channel dial-string. The dialplan is essentially a scripting language specific to Asterisk and one of the primary ways of instructing Asterisk on how to behave. This application attempts to detect answering machines at the beginning of outbound calls. The Hangup Cause family of functions and dialplan applications allow for inspection of the hangup cause codes for each channel involved in a call. a - Set admin mode. This application will set the current context, extension, and priority in the channel structure based on the evaluation of the given condition. 7 Documentation ; Test Suite Documentation ; Historical Documentation Process1. 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. Supported options are those fields on the aor object in pjsip. an Open Source software development project. Upgrading to Asterisk 16 ; New in 16 ; API Documentation . Modules. SUCCESS - Specified command successfully executed. There are a few items to check. C - Continue in dialplan when kicked out of conference. conf, or changing the 'astdbdir' option to a directory for which the user running Versions of Asterisk. OMIT - This CDR should be ignored. Use of batch mode may result in data loss after unsafe asterisk termination, i. 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. The Asterisk dialplan. Async - Set to 'true' for fast origination. When loaded, AMD reads amd. x. Those default values get overwritten when the calling AMD with parameters. As an Asterisk administrator, you have the choice on which modules to load and the configuration of each module. Currently, JSON is the only supported message description format. b - Run AGI script specified in MEETME_AGI_BACKGROUND Default: 'conf-background. 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. Local/101@mycontext/nj. 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. 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. disable - Setting to 1 will disable CDRs for this channel. It is primarily intended for use with analog lines, but could be useful for other channels as well. AMAFlags - A flag that informs a billing system how to treat the CDR. IsExternal - Indicates if the transfer was performed outside of Asterisk. At that point, this application will exit with the status variable set and dialplan processing will continue. The default timeout is 5 seconds. # asterisk -c -v -v. The AGC function will apply automatic gain control to the audio on the channel that it is executed on. Defaults to the database setting in res_odbc. It is important that this directory is writable by the user Asterisk runs as. conf file, extensions. SQLite 3 creates a journal file in the 'astdbdir' specified in asterisk. Standard releases are supported for a shorter period of time Asterisk 16 Documentation ; Asterisk 18 Documentation . If provided, the applications listed will be subscribed to all events, effectively disabling the application specific subscriptions. Description. The dialplan script told Asterisk which Overview. If the command fails, the console should report a fallthrough. 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. The technology chosen for sending the message is determined based on a prefix to the 'destination' parameter. asterisk. Control of the calls that passed through it was done through a special . c - Announce user (s) count on joining a conference. name - The name of the AOR to query. Setting to 0 will enable CDRs for this channel. x required - The announcement to playback to all devices. running on Linux (or other types of Unix ) powering Business Telephone Systems. This release is available for immediate download at https://downloads. 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. You can add more than one modifier by adding them directly adjacent to the previous modifier. If a space is desired at the beginning of the data, then put two spaces there; the second will not be skipped. 2026-10-18. batch. At least a priority is required as an argument, or the goto will return a '-1',and the channel and call will be terminated. Fully Supported. Mar 12, 2024 · AstriCon is the longest-running open source convention celebrating open source projects featuring Asterisk and FreePBX. The type of release defines how long it will be supported. Certified Asterisk 20. VoIP Gateways. Here we make an admin/marked user out of the 'my_user' profile that you define in confbridge. , software crash, power failure, kill -9, etc. Solution. 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. UserField - A user defined field set on the channels. 16. beep - Causes Asterisk to play a beep as recording begins. This application will block until the outgoing call fails or gets answered, unless the async option is used. Simply call this application after the call has been answered (outbound only, of course). This application originates an outbound call and connects it to a specified extension or application. 2023-10-18. name - The property to set on the CDR. 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. Standard. 0 resolves several issues reported by the. On Read - Retrieves unencoded message/subtype in Base64 encoded form. Multiple calls add multiple headers. sample in the [general] section of queues. Each module that you load Queue Logs. Certified Asterisk 18. Ex: datastore->data = mysillydata; Add datastore to the channel. The realtime Architecture lets you store all of your configuration in databases and reload it whenever you want. Jan 14, 2010 · The next step is to add a couple of queues to Asterisk that we can assign queue members into. field - The configuration option for the AOR to query for. written in the C Programming Language. All variables will be evaluated at the time MixMonitor is called. Command - Will be executed when the recording is over. Verify that there is not a ' noload' line for the module that is failing to load. 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. New releases of Asterisk will be made roughly once a year, alternating between standard and LTS releases. asterisk -rx "reload". It ties everything together, allowing you to route and manipulate calls in a programmatic way. 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)'. AMI Actions¶. Be sure you have a backup of any essential data on the system. Back to top. 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. agi'. 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. AGI Commands ; AMI Actions ; AMI Events ; Asterisk REST Asterisk 21 Documentation. Upgrading to Asterisk 20 Asterisk 16 Documentation . This section contains many sub-sections on configuring every aspect of Asterisk. This would create a feature called 'eggs' that could be invoked during a call by pressing the '*5'. You may also dynamically add SIP and IAX devices and extensions and making them available without Asterisk 20 Documentation. After this duration, any page calls that have not been answered will be hung up by Arguments. unixtime - time, in seconds since Jan 1, 1970. Building AMI Event documentation for Asterisk requires both libxml and python. UniqueID - A unique identifier for the Party A channel. Any strings matching '^ {X}' will be unescaped to X. Asterisk 21 Documentation. Please note that the space following the double quotes separating the regex from the data is optional and if present, is skipped. 7 Documentation. 2025-10-18. API Documentation . BILLING - This CDR contains valid billing data. 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. DOCUMENTATION - This CDR is for documentation purposes. Asterisk 20 Documentation . This modularity gives you an almost unlimited amount of flexibility in the design of an Asterisk-based system. 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. OtherChannelId - Channel UniqueId to be set on the second local channel. 9 Documentation ; Certified Asterisk 20. 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. Module Configuration. If the variable name is prefixed with '__', the variable will be inherited into channels created Query parameters. For instance, a channel protocol native transfer is external. Defaults to now. 9 Documentation. The Asterisk Development Team would like to announce the release of Asterisk 16. 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. so. 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 SIP presence event package) will receive decoded values. Arguments. 4. This allows a dialplan writer to determine, for each channel, who hung up and for what reason (s). A Long Term Support release is fully supported for 4 years, with an additional year of maintenance for security fixes. For now we'll work with two queues; sales and support. conf and uses the parameters specified as default values. Asterisk 16 Documentation ; Asterisk 18 Documentation ; Asterisk 19 Documentation ; Asterisk 20 Documentation ; Asterisk 21 Documentation ; Certified Asterisk 18. We'll leave the default settings that are shipped with queues. A - Set marked mode. 19. format - a format the time is to be said in. Overview. ey qm um tv rk yr vv co ol rc