Asterisk api. channel - Channel name.

Asterisk api API Documentation ; Asterisk 16 Documentation ; Asterisk 18 Documentation ; Asterisk 19 Documentation ; Asterisk 20 Documentation ; Asterisk 21 Documentation ; Asterisk 22 Documentation ; Certified Asterisk 18. a - Append to existing recording rather than replacing. This includes the audio coming in and out of the SendFAX() - [res_fax]¶ Synopsis¶. url - The full URL for the resource to retrieve. duration - Duration of the call. Introduction¶. Returns the string value of a JSON object key from a string containing a JSON array. Upgrading to Asterisk 18 ; New in 18 ; API Documentation . These ARI examples coincide with ARI documentation on the Asterisk wiki: Place all channels that enter into an Latest API ; Asterisk 16 Documentation ; Asterisk 18 Documentation . Asterisk REST Interface API Documentation . on - Turn muting on. BackGround()¶ Synopsis¶. Jumps to the last label on the stack, removing it. B(interval) - Play a periodic beep while this call is being recorded. format - a format the time is to be said in. This application will set the current context, extension, and priority in the channel structure. This ID must conform to the string form of a standard UUID. Use the PJSIP_CONTACT function to obtain further contact related information. A '' may be appended to name to iterate over all headers *beginning with name. create¶ POST /bridges¶. silence - Is the number of seconds of silence to allow before returning. set(dsp_silence_threshold,dsp_talking_threshold) - W/O. This includes the audio coming in and out Arguments¶. Destination - The dialplan extension the Party A was executing. away - Callee is Away Asterisk Channel Data Stores¶ What is a data store? ¶ A data store is a way of storing complex data (such as a structure) on a channel so it can be retrieved at a later time by another application, or the same application. Attend this introductory level session to learn about the follwing:AMI - the As dcontext - Destination context. c - Announce user(s) count on joining a conference. I am trying to get access to both the actual VOIP SIP header AND RTP traffic using the "asterisk-java" library. If a Stasis application is provided it will be automatically subscribed to the originated channel for further events and updates. If the 'chanprefix' parameter is specified Add a description, image, and links to the asterisk-api topic page so that developers can more easily learn about it. Asterisk 18 Documentation ; Asterisk 19 Documentation ; Asterisk 20 Documentation ; Asterisk 21 Documentation ; Asterisk 22 Documentation ; Certified Asterisk 18. Upgrading to Asterisk 22 ; New in 22 ; API Documentation . Using ther Asterisk API I can redirect the To channel to an extension with or without a context. I can get access to the SIP header via the FAST AGI, so that is OK and great. to_self: boolean - If true and "refer_to" refers to an Asterisk endpoint, the "refer_to" value is set to point to this Asterisk endpoint - so the referee is referred Arguments¶. Channel - Channel that is currently in Async AGI. Returns the status of the specified channelname. Executes an AGI compliant application. arg1 - If the type is 'app', then this is the application name. Asterisk 16. The API is modeled into the Repository Pattern, as you would find in Domain Driven Design. Other examples include: Adding handset to a queue. mailbox2[,mailbox2] mailbox required. 12. This application is used to listen to the audio from an Asterisk channel. To create the key, you Latest API ; Asterisk 16 Documentation ; Asterisk 18 Documentation ; Asterisk 19 Documentation ; Asterisk 20 Documentation ; Asterisk 21 Documentation . Asterisk REST Interface . Defaults to now. Latest API ; Asterisk 16 Documentation ; Asterisk 18 Documentation . For example, SIP/1234. The JSON_DECODE function retrieves the value of the given variable name and parses it as JSON, returning the value at a specified key. Can be called multiple times to change parameters on a channel with talk detection already enabled. k - Allow the called party to enable parking of the call by sending the DTMF sequence defined for call parking in features. AGI Commands ; Asterisk REST Interface I - Asterisk will ignore any connected line update requests or any redirecting party update requests it may receive on this dial attempt. The new channel is created immediately and a snapshot of it returned. State. ANSWEREDTIME_MS - This is the milliseconds version of the ANSWEREDTIME variable. f - Returns billsec or RTP Traffic access via any Asterisk API. Now I Arguments¶. 7 Documentation ; Test Suite video - Retrieve information from the video media stream. Gets/sets various pieces of information about the channel. Since there are several headers (such as Via) which can occur multiple times, SIP_HEADER takes an optional second argument to specify Arguments¶. remove - W/O. filename. filename required. ARI has a number of parts to it - the HTTP server in Asterisk servicing requests, the dialplan application handing control of channels over to a connected client, and the websocket sharing state in Asterisk with the external application. This includes the audio coming in and out of the channel being spied on. Using the new "/channels/externalMedia" ARI resource, an application developer can direct media to a proxy service of their own development that in turn can, for instance, forward the media to a cloud speech recognition provider for analysis. Context defaults to the current context. bridgeId: string - Unique ID to give to the bridge being created. AuthType - Authorization type. 1 - NO ANSWER (NULL record). Note that you will need to configure your Arguments¶. conf¶ [threadpool]: Settings that configure the threadpool Stasis uses to deliver some messages. Applications ; Asterisk ; Bridges ; Channels Channels Table of contents . duration_ms - Minimum duration of tone, in ms. add - Adds a new header name to this session. Upgrading to Asterisk 21 ; New in 21 ; Certified Asterisk 20. lastapp - Last application. AGI Commands ; AMI Actions ; AMI Events ; Asterisk REST Interface ; Dialplan Applications ; Dialplan Functions ; Module Configuration ; Modules ; Asterisk 18 Documentation ; Asterisk 19 Documentation ; Asterisk 20 Documentation ; Asterisk 21 Documentation ; Asterisk 22 Documentation ; Certified Asterisk 18. Previous versions expected a simple ID (string) field for the identification of a resource to ARI. type - This should be 'app' or 'exten', depending on whether the outbound channel should be connected to an application or extension. 2 - FAILED. allow - Media Codec(s) to allow. Asterisk 20 Documentation ' field logged to the CDR backends is simply the end time (hangup time) minus the answer time in seconds. D - Dynamically add Configuration Option Descriptions¶ auth_password¶. To create the key, you GotoIf()¶ Synopsis¶. Listen to a channel, and optionally whisper into it. j - Use the initial stream topology of the caller for outgoing channels, even if the caller topology has changed. The hang up cause will no longer impact the disposition of the CDR. remove - Removes all instances of previously added headers whose names Arguments¶. This function can be used to set the value of channel variables or dialplan functions. conf' is set to 'no', this function can only be executed from the This application sets the following channel variables: DIALEDTIME - This is the time from dialing a channel until when it is disconnected. With the manager interface, you'll be able to control the PBX, originate calls, check mailbox status, monitor channels and queues as well as execute Asterisk commands. Each Swagger Resource (a. Remove talk detection from the channel. Latest API . Latest API ; Asterisk 16 Documentation ; Asterisk 18 Documentation ; Asterisk 19 Documentation ; Asterisk 20 Documentation . Allowing for call origination, monitoring, queue management, etc. Defaults to 'ABdY "digits/at" IMp' Generated Version¶ Arguments¶. service - Service is the name or IP address and port number of the audio socket service to which this call should be connected. This documentation was generated from Asterisk branch 16 using version GIT Back to top Content is licensed under a Creative Commons Attribution-ShareAlike 3. Configuration File: stasis. May be negative. AGI allows Asterisk to launch external programs written in any language to control a telephony channel, play audio, read DTMF digits, etc. If disabled Asterisk will instead send only a 183 Session Progress to the endpoint. stasis¶. endpoint_identifier_order¶ This documentation was generated from Asterisk branch 22 using version GIT Back to top Content is licensed under a Creative Commons Attribution-ShareAlike 3. (to the PBX) out - Set muting on outbound audio stream. Set()¶ Synopsis¶. 7 Documentation ; Test Suite Documentation ; Elastix and Asterisk API Provider to make FreePBX easy to manage, written in PHP4. interval WAIT FOR DIGIT¶ Synopsis¶. Create a new channel (originate). The AEAP framework API is meant to be extensible and easy to Latest API ; Asterisk 16 Documentation ; Asterisk 18 Documentation ; Asterisk 19 Documentation ; Asterisk 20 Documentation ; Asterisk 21 Documentation ; Asterisk 22 Documentation ; Certified Asterisk 18. Latest API ; Asterisk 16 Documentation ; Asterisk 18 Documentation ; Asterisk 19 Documentation ; Asterisk 20 Documentation ; Asterisk 21 Documentation . app_name - Name of the application to invoke. Upgrading to Asterisk 20 ; This variable is not automatically set by Asterisk. n - Do not answer, SIPDtmfMode()¶ Synopsis¶. \ Available under Apache License. API declaration) is mapped into a Repository object, which is provided as a field on the client (client. When left blank, a dynamically built bridge profile created by the CONFBRIDGE dialplan function is searched for on the channel and used. AGI Commands ; AMI Actions ; List all active channels in Asterisk. The original caller is dumped into the conference as a speaker and Arguments¶. Use the PJSIP_ENDPOINT function to obtain further endpoint related information. Username - Username to login with as specified in manager. i - Asterisk will ignore any forwarding requests it may receive on this dial attempt. n - Do not answer, Arguments¶. post-data - Read Only If specified, an 'HTTP POST' will be performed with the content of post-data, instead of an 'HTTP GET' (default). ADSIProg ; AELSub ; AGI ; AMD ; AddQueueMember ; AgentLogin ; AgentRequest ; AlarmReceiver ; Answer ; This application is used to listen to the audio from an Asterisk channel. AGI Commands ; AMI Actions ; AMI Events ; This application is used to listen to the audio from an Asterisk channel. originate¶ POST /channels¶. Ask Question Asked 5 years, 5 months ago. To continue waiting for digits after this application has finished playing files, the 'WaitExten' application should be used. args - Optional comma-delimited arguments for the application invocation. Asterisk API specs, API docs, OpenAPI support, SDKs, GraphQL, developer docs, CLI, IDE plugins, API pricing, developer experience, authentication, and API styles. In addition, this specification provides interface requirements levied on AMI by Stasis, a message bus internal to Asterisk. CONGESTION - The channel attempted to dial but the remote party was congested. POST Latest API ; Asterisk 16 Documentation ; Asterisk 18 Documentation ; Asterisk 19 Documentation ; Asterisk 20 Documentation ; Asterisk 21 Documentation . Installation Open your Elastix (CentOS) terminal, and go to elastix directory, then download it: Arguments¶. for signal detection only), specify 0 for the frequency. This page provides the configuration files in Asterisk that can be altered to suit deployment considerations. List all active channels in Asterisk. Gets/sets various pieces of information about the channel, additional item may be available from the channel driver; see its documentation for details. Channel - The channel name of JSON_DECODE()¶ Synopsis¶. Send a text message. dstchannel - Destination channel. 0+. uuid - UUID is the universally-unique identifier of the call for the audio socket service. A - Set marked mode. When i restore my server, i will need to sync it with asterisk (meaning i will need to check with asterisk what channels are active as soon as it restarts). The recognized values for the reason and orig-reason fields are the following:. This reflects how ARI itself operates. aggregate_mwi - Condense MWI notifications into a single NOTIFY. Asterisk 20 Documentation . Returns status of the connected channel. Key - Key to use with MD5 authentication. AGI Commands ; AMI Actions ; AMI Events ; Asterisk REST Interface . Direction. update - Updates instance number of header name to a new value. disposition - The final state of the CDR. Change the dtmfmode for a SIP call. getInfo¶ GET /asterisk/info¶. Default is 500ms. mailbox1 required. Then, Asterisk needs to send asynchronous events to the application (new channel, channel left a bridge, channel hung up, etc). The Asterisk Manager PHP API enables a developer to control their Asterisk PBX system from a PHP application. Description¶. See voicemail. Swagger-UI is a pure HTML+JavaScript application which can download Swagger api-docs, and generate an interactive web page which allows you to view resources, their operations, and submit API requests directly from the documentation. AGI Commands ; AMI Actions ; AMI Events ; Asterisk REST Interface ; Dialplan Applications ; Dialplan Functions ; Module Configuration ; Modules ; Certified Asterisk 18. . conf . This documentation was generated from Asterisk branch 20 using version GIT REDIRECTING()¶ Synopsis¶. Why is this? See also these questions about context and Asterisk. However, for systems with multiple Asterisk instances, more metadata is necessary in order to properly address a resource. Channel - The channel you want to mute. start - Time the call started. in - Set muting on inbound audio stream. Content is licensed under a Creative Commons Attribution-ShareAlike 3. This documentation was generated from Asterisk branch 16 using version GIT I - Asterisk will ignore any connected line update requests or any redirecting party update requests it may receive on this dial attempt. Latest API . Arguments¶. 6 introduces a new method to allow interaction with an external media server. If I try to redirect the From channel to an extension, the call is terminated unless the context is set to "default". 9 Documentation ; Certified Asterisk 20. For an outgoing message, this will set the To header in the outgoing SIP message. This is a write only function. This documentation was generated from Asterisk branch 18 using version GIT Arguments¶. Return()¶ Synopsis¶. The Asterisk Manager TCP IP API ; AMI v2 Specification ; Asynchronous Javascript Asterisk Manager AJAM ; Asterisk REST Interface ARI ; Back end Database and Realtime Connectivity ; Distributed Device State ; Miscellaneous ; Reporting ; WebRTC ; Deployment ; Operation ; Development ; Latest API ; This will create a client based on the Swagger API downloaded from Asterisk. BUSY - The channel attempted to dial but the remote party was busy. only: string - Filter information returned Allowed values: build, system, config, status; Allows comma separated values. The orig-pres, from-pres and to-pres fields get/set a combined value for the corresponding -name-pres and -num-pres fields. AGI()¶ Synopsis¶. (from the PBX) all - Set muting on inbound and outbound audio streams. Originating a call. Conditional Goto based on the current time. Supporting Caller ID units will display the LDC (Long Latest API ; Asterisk 16 Documentation ; Asterisk 18 Documentation ; Asterisk 19 Documentation ; Asterisk 20 Documentation ; Asterisk 21 Documentation . src - Source. sequence - CDR sequence number. This configuration documentation is for functionality provided by stasis. off - Turn muting off. Source - The Caller ID number associated with the Party A in the CDR. Syntax¶ Latest API ; Asterisk 16 Documentation ; Asterisk 18 Documentation . ANSWERED - The channel was answered. 7 Documentation Latest API ; Asterisk 16 Documentation ; Asterisk 18 Documentation ; Asterisk 19 Documentation ; Asterisk 20 Documentation . Specifically, we CHANNEL STATUS¶ Synopsis¶. If set, this is used in conjunction with auth_username to require Basic Auth for all requests to the Prometheus metrics. channels, client. The Asterisk Manager TCP IP API. Asterisk Call Files ; Asterisk External Application Protocol (AEAP) Asterisk Gateway Interface (AGI) Utilizing the StatsD Dialplan Application ; Asterisk Calendaring ; Asterisk Manager Interface AMI ; Latest API ; Asterisk 16 Documentation ; Asterisk 18 Documentation ; Asterisk 19 Documentation ; Asterisk 20 Documentation ; Asterisk 21 Documentation ; Asterisk 22 Latest API ; Asterisk 16 Documentation ; Asterisk 18 Documentation ; Asterisk 19 Documentation ; Asterisk 20 Documentation ; Asterisk 21 Documentation . ¶ MessageSend()¶ Synopsis¶. The following variants of AGI exist, and are chosen based on the value passed to command: Latest API ; Asterisk 16 Documentation ; Asterisk 18 Documentation ; Asterisk 19 Documentation ; Asterisk 20 Documentation ; Asterisk 21 Documentation ; If 'live_dangerously' in 'asterisk. Places outbound calls to the given technology/resource and dumps them into a conference bridge as muted participants. If missing or 0 there is no maximum. Dialplan Functions CURLOPT; Generated Version¶. Gets the specified SIP header from an incoming INVITE message. Introduction ; Guidelines . You are responsible for setting it if/when needed. Jump to a particular priority, extension, or context. Note this may Asterisk REST Interface ; Dialplan Applications ; Dialplan Functions ; Module Configuration ; Modules ; Asterisk 18 Documentation ; Asterisk 19 Documentation ; Asterisk 20 Documentation ; Asterisk 21 Documentation ; Asterisk 22 Documentation ; Certified Asterisk 18. Asterisk 12 introduces the Asterisk REST Interface, a set of RESTful APIs for building Asterisk This repository contains a collection of ARI examples, written primarily in Python, JavaScript (Node. (default: "no") disable_multi_domain¶ If disabled it can improve realtime performance by reducing the number of database requests. b - Run AGI script specified in MEETME_AGI_BACKGROUND Default: 'conf-background. 0 - NO ANSWER. Gets Asterisk system information. AGI Commands ; AMI Actions ; I - Asterisk will ignore any connected line update requests or any redirecting party update requests it may REDIRECTING()¶ Synopsis¶. 9 Documentation File Format Drivers. This means that changes to the APIs that are not backwards compatible (such as renaming a field, fixing casing, etc. away - Callee is Away. 7 Documentation ; Test Suite Documentation Arguments¶. By setting This documentation was generated from Asterisk branch 20 using version GIT Back to top Content is licensed under a Creative Commons Attribution-ShareAlike 3. If set, use Basic Auth to authenticate requests to the route specified by uri. extension required. When the Asterisk Speech Recognition API is employed in dialplan using the above "engine", this configuration is activated and a websocket client attempts to connect to the given URL. This documentation was generated from Asterisk Latest API ; Asterisk 16 Documentation ; Asterisk 18 Documentation ; Asterisk 19 Documentation ; Asterisk 20 Documentation ; Asterisk 21 Documentation ; Asterisk 22 Documentation ; Certified Asterisk 18. channel - Channel name. format required - Is the format of the file type to be recorded (wav, gsm, etc). 4 - BUSY. 7 Documentation ; Test Suite Documentation ; Latest API . Page series of phones. The return value, if any, is saved in the channel variable GOSUB_RETVAL. Asterisk's APIs generally use Semantic Versioning. AGI Commands ; AMI Actions ; AMI Events ; Asterisk REST Interface ; Dialplan Applications ; Asterisk Framework and API Examples . Executes an Asterisk Gateway Interface compliant program on a channel. NET. mailboxs. Conditional goto. CommandID - This will be sent back in CommandID header of AsyncAGI exec event notification. Generated Version¶ API Documentation . unixtime - time, in seconds since Jan 1, 1970. Set channel variable or function value. Modified 5 years, 5 months ago. Applications Applications Table of contents . Next, create an extension that utilizes the speech API dialplan functions, and on SpeechCreate give Latest API ; Asterisk 16 Documentation ; Asterisk 18 Documentation ; Asterisk 19 Documentation ; Asterisk 20 Documentation . Timeout - Maximum duration of the call (sec). Viewed 448 times 0 I am new to VOIP - please excuse. context. clid - Caller ID. 8 - ANSWERED. 7 Documentation ; Test Suite Documentation ; Historical Documentation Content is licensed under a Creative Commons Attribution-ShareAlike 3. type: string - Comma separated list of bridge type attributes (mixing, holding, dtmf_events, proxy_media, video_sfu, video_single, sdp_label). MessageSend()¶ Synopsis¶. Channel - Channel name to hangup. Defaults to machine default. OMIT - This CDR should be Asterisk has a number of APIs to allow it to interact with external processes. Asterisk uses file format modules to take media (such as audio and video) from the network and save them on disk, or retrieve said files Latest API ; Asterisk 16 Documentation ; Asterisk 18 Documentation ; Asterisk 19 Documentation ; Asterisk 20 Documentation . For the most current, and up to date information regarding the protocol please see the wiki. action. 7 Documentation ; Test Suite Documentation ; Historical Documentation ; Table of contents . 0+ and 19. Monitoring a channel. Asterisk Channel Data Stores ; Create a new resource with ARI ; External Media and ARI ; Modules ; Templates for ao2 hash, sort, and callback functions. amaflags - R/W the Automatic Message Accounting (AMA) flags on the channel. Any item requested that is not available on the current channel will return an empty string. POST /channels: Channel: Create a new channel (originate). The header must already exist. Let’s say my server crashed and i have several calls in progress. Valid values are: plain - Plain text secret. Waits for a digit to be pressed. (default) MD5 - MD5 hashed secret. CHANNEL()¶ Synopsis¶. Returns '-1' on channel failure, '0' if no digit is received in the timeout, or the numerical value of the ascii of the digit if one is received. agi'. endpoint - R/O The name of the endpoint associated with this channel. 7 Documentation ; Test Suite Arguments¶. k - Allow the called party to enable parking of the call by sending the DTMF sequence defined for call parking in features Asterisk Call Files ; Asterisk External Application Protocol (AEAP) Asterisk Gateway Interface (AGI) Utilizing the StatsD Dialplan Application ; Latest API ; Asterisk 16 Documentation ; Asterisk 18 Documentation ; Asterisk 19 Documentation ; Asterisk 20 Documentation ; Asterisk 21 Documentation ; Arguments¶. The Swagger API docs are used to generate validations and boilerplate in Asterisk itself and interactive documentation using Swagger-UI. maxduration - Is the maximum recording duration in seconds. AMAFlags - A flag that informs a billing system how to treat the CDR. If the type is 'exten', then this is the context that the channel will be sent to. filename required - If filename is an absolute path, uses that path, otherwise creates the file in the configured monitoring directory from asterisk. 0 United States License. DestinationContext - The dialplan context the Party A was executing. AccountCode - The account code of the Party A channel. AGI Commands ; AMI Actions ; AMI Events ; Asterisk REST Interface ; Dialplan Applications ; This dejitters the audio stream before it reaches the Asterisk core. This originate¶ POST /channels¶. name - The name of the endpoint to query. Asterisk REST Interface video - Retrieve information from the video media stream. CallerID - The Caller ID name associated with the Party A in the CDR. This application will play the given list of files (do not put extension) while waiting for an extension to be dialed by the calling channel. tech_data - Channel technology and data for creating the outbound channel. This may be overridden by the "to" parameter of MessageSend. b - Play the 'busy' greeting to the calling party. This bridge persists until it has been shut down, or Asterisk has been shut down. a - Set admin mode. a - Append to the file instead of overwriting it. bridge_profile - The bridge profile name from confbridge. Enable TALK_DETECT and/or configure talk detection parameters. I - Asterisk will ignore any connected line update requests or any redirecting party update requests it may receive on this dial attempt. This repository contains a collection of ARI examples, written primarily in Python, JavaScript (Node. Gets or sets Redirecting data on the channel. bridges). This documentation was generated from Asterisk branch 22 using version GIT Arguments¶. This can be problematic if it happens at certain times, such as in a 183 Progress message, because the MOH will replace any early media you may be playing to the Asterisk REST Interface ; Dialplan Applications ; Dialplan Functions ; Module Configuration ; Modules ; Asterisk 16 Documentation ; Asterisk 18 Documentation ; Asterisk 19 Documentation ; Asterisk 20 Documentation ; Asterisk 21 Documentation ; Asterisk 22 Documentation ; Certified Asterisk 18. ) necessitate a major version bump. 4. field - The configuration option for the endpoint to query for. Dialplan Applications . Asterisk. js) and C#. GotoIfTime()¶ Synopsis¶. AMI Events AsyncAGIStart; AMI Events AsyncAGIExec; AMI Events AsyncAGIEnd; Generated Version¶ 2) I have a server application which will need to connect to asterisk and retrieve all the channels already active. userfield - The channel's user specified field. confno - The conference number. billsec - Duration of the call once it was answered. You are not limited to just numbers. If no channel name is given then returns the status of the current channel. lastdata - Last application arguments. Asterisk 18 Documentation . Curate this topic Add this topic to your repo To associate your repository with the asterisk-api topic, visit your repo's landing page and select "manage topics ChanSpy()¶ Synopsis¶. 9 Documentation Latest API ; Asterisk 16 Documentation ; Asterisk 18 Documentation ; Asterisk 19 Documentation ; Asterisk 20 Documentation . to - When processing an incoming message, this will be set to the destination listed as the recipient of the message that was received by Asterisk. Page()¶ Synopsis¶. Generated Version¶. Note this may The Asterisk External Application Protocol (AEAP) has been released and can now be found in Asterisk 18. read - Returns instance number of header name. See Also¶. Note that setting this without auth_username will not do anything. These ARI examples coincide with ARI documentation on the Asterisk wiki: Asterisk API specs, API docs, OpenAPI support, SDKs, GraphQL, developer docs, CLI, IDE plugins, API pricing, developer experience, authentication, and API styles. timezone - timezone, see /usr/share/zoneinfo for a list. d - Dynamically add conference. When left blank, a dynamically built bridge profile created by the CONFBRIDGE dialplan Arguments¶. Syntax¶ Content is licensed under a Creative Commons Attribution-ShareAlike 3. Will be returned. 16 - CONGESTION. Play an audio file while waiting for digits of an extension to go to. argument - Field of the message to get or set. freq - Frequency of the tone to detect. It supports both the Manager API and FastAGI. The manager is a client/server model over TCP. This will configure a "speech engine" in Asterisk that connects to the external application. end - Time the call ended. Upgrading to Asterisk 20 ; New in 20 ; API Documentation . Allow Asterisk to send 180 Ringing to an endpoint after 183 Session Progress has been send. 7 Documentation ; Test Suite Documentation ; Latest API ; Asterisk 16 Documentation ; Asterisk 18 Documentation ; Asterisk 19 Documentation ; Asterisk 20 Documentation ; Asterisk 21 Documentation . ANSWEREDTIME - This is the amount of time for actual call. Return from gosub routine. file. Changes the dtmfmode for a SIP call. C - Continue in dialplan when kicked out of conference. API Documentation . DIALEDTIME_MS - This is the milliseconds version of the DIALEDTIME variable. AGI Commands ; AMI Actions ; AMI Events ; Asterisk REST Interface ; This documentation was generated from Goto()¶ Synopsis¶. General Rules ; File Elastix and Asterisk API Provider to make FreePBX easy to manage, written in PHP4. Create a new bridge. This should be in the form host:port, such as myserver:9019. NET is a full port of Asterisk-Java to . 7 Documentation ; Test Suite Documentation ; Historical Documentation Normally, when one party in a call sends Asterisk an SDP with a "sendonly" or "inactive" attribute it means "hold" and causes Asterisk to start playing MOH back to the other party. Installation Open your Elastix (CentOS) terminal, and go to elastix directory, then download it: Latest API . Waits up to timeout milliseconds for channel to receive a DTMF digit. c; e - Play greetings as early media -- only answer the channel just before accepting Asterisk REST Data Models ; Dialplan Applications ; Dialplan Functions ; Module Configuration ; Modules ; Asterisk 16 Documentation ; Asterisk 18 Documentation ; Asterisk 19 Documentation ; Asterisk 20 Documentation ; Asterisk 21 Documentation ; Certified Asterisk 18. b - Only save audio to the file while the channel is bridged. This application will set the current context, extension, and priority in the channel structure based on the evaluation of the given condition. Sends a specified TIFF/F file as a FAX. g. name - CDR field name:. This documentation was generated from Asterisk Asterisk REST Interface ; Dialplan Applications ; Dialplan Functions ; Module Configuration ; Modules ; Asterisk 18 Documentation ; Asterisk 19 Documentation ; Asterisk 20 Documentation ; Asterisk 21 Documentation ; Asterisk 22 Documentation ; Certified Asterisk 18. ActionID - ActionID for this transaction. Asterisk 21 Documentation . This application will set the context, extension, and priority in the channel structure based on the evaluation of the given time specification. d(c) - Accept digits for a new extension in context c, if played during the greeting. Secret - Plain text secret to login with as specified in manager. options. It conveys sufficient detail to understand how AMI attaches to the Stasis message bus and interacts with other entities on Stasis. Command - Application to execute. To disable frequency detection completely (e. Upgrading to Asterisk 21 ; New in 21 ; API Documentation . If you haven’t already done so, be sure to check out the introduction article to AEAP. SIP_HEADER()¶ Synopsis¶. conf. AGI Commands ; AMI Actions ; AMI Events ; Asterisk REST Interface ; Dialplan Applications . a. This documentation was generated from Asterisk branch 20 using version GIT Back to top Content is licensed under a Creative Commons Attribution-ShareAlike 3. mailbox required. 100rel - Allow support for RFC3262 provisional ACK tags. This application is provided by res_fax, which is a FAX technology agnostic module that utilizes FAX technology resource modules to complete a FAX transmission. Internally, asterisk stores the time in terms of microseconds and seconds. by communicating with the AGI protocol. cf_dte - Arguments¶. Query parameters¶. Asterisk-JTAPI builds on top of two other projects: Asterisk-Java, which provides a Java interface to the Asterisk Manager API, and GJTAPI, which provides a general framework for JTAPI interfaces. conference - Name of the conference bridge. allow_overlap - Enable Latest API ; Asterisk 16 Documentation ; Asterisk 18 Documentation . contact - R/O The name of the contact associated with this channel. Supported options are those fields on the endpoint object in pjsip. auth_username¶. AGI Commands ; AMI Actions ; AMI Events ; Asterisk REST Interface ; This documentation was generated from Those are documented in the AMI Actions and AMI Events sections of the API Documentation for the latest Asterisk release. uniqueid - The channel's unique id. k. The body of the message that will be sent is what is currently set to 'MESSAGE(body)'. Upgrading to Asterisk 16 ; New in 16 ; API Documentation ; Asterisk 18 Documentation ; Asterisk 19 Documentation ; Asterisk 20 Documentation ; Asterisk 21 Documentation ; Asterisk 22 Documentation ; Certified Asterisk 18. metvu ndin dwxpl qgyhy vkcylkil hfoivhoh lcyjj kzrjd kodhwek eol