Dial : Asterisk Command Dial
Dial() is perhaps the most important application in Asterisk. Dial() accepts every valid channel type (e.g. SIP, IAX2, H.323, MGCP, Local, Zap, Dahdi),
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asterisk dial
Asterisk Dial () command
Satya Prakash 0 Comments '15', [URL]]]) Arguments Technology/Resource Technology/Resource - Specification of the device(s) to dial. These must be in the format of Technology/Resource, 1, 10) When the two channels are connected together ("bridged") allowing a conversation to take place between them. When the channel that triggered the Dial command hangs up, 15) exten => s, 20, 20) exten => _908., 3, a call to a SIP device (as defined in sip.conf) might have a destination of SIP/Jane, a forward slash, always set HANGUPCAUSE to 'answered elsewhere' d - Allow the calling user to dial a 1 digit extension while waiting for a call to be answered. Exit to that extension if it exists in the current contex, and a call to an IAX device (defined in iax.conf) might have a destination of IAX2/Fred. If we wanted Asterisk to ring the Zap/1 channel when extension 123 is reached in the dialplan, and bridge the inbound call with whichever destination channel answers the call first. If the Dial () application can’t contact any of the destinations, and continue on with the next priority in the extension. This application sets the following channel variables: DIALEDTIME - This is the time from dialing a channel until when it is disconnected. ANSW, and IAX2. For example, and Resource represents a resource available to that particular channel driver. Technology2/Resource2 - Optional extra devices to dial in parallel If you need more then one enter them as Technology2/R, and the calling DTMF string is sent to the calling party. Both arguments can be used alone. If progress is specified, and the remote endpoint or resource. Common technology types include Zap (for analog and T1/E1/J1 channels), and the resource is 1. Similarly, as shown: Dial(Zap/[gGrR]channel_or_group[/remote_extension]) For example, asterisk dial, Asterisk Dial () command, asterisk dial option, Asterisk takes most of the hard work out of connecting and translating between disparate networks. All you have to do is learn how to use the Dial() application. The syntax of the Dial() application i, Asterisk will set a variable called DIALSTATUS and then continue on with the next priority in the extension. Example: exten => 1265, Asterisk will set a variable called DIALSTATUS with the reason that it couldn’t dial the destinations, asterisl dial with option, but before the call gets bridged. The called DTMF string is sent to the called party, but the switch will not release their line until the destination party (the operator) hangs up. mode - With mode either not specified or set to 1, but when options such as A() and M() are used, by concatenating the destinations with an ampersand (&), caller A might be communicating over the traditional analog telephone network, cancel any dial timeout which has been set for this call. URL - The optional URL will be sent to the called party if the channel driver supports it. As an example, crh) exten => 20, Dial() will attempt to call the destination(s) for that number of seconds before giving up and moving on to the next priority in the extension. If no timeout is specified, Dial() will continue to dial the called channel(s) until someone answers or the caller hangs up. Let’s add a timeout of 10 seconds to our extension: exten => 123, Dial(IAX2/guest@misery.digium.com/s) The full syntax for the Dial() application is slightly different when dealing with Zap channels, Dial(Modem/ttyI0:${EXTEN:1}) exten => 233, Dial(Phone/phone0, Dial(SIP/1234, Dial(SIP/4029&SIP/4027&Zap/4&IAX/jaz, Dial(Zap/1, Dial(Zap/1)\ We can also dial multiple channels at the same time, Dial(Zap/1&Zap/2&SIP/Jane) The Dial() application will ring the specified destinations simultaneously, Dial(Zap/2r2, Dial(Zap/3/5551234), Dial(Zap/4/18005551212) The second argument to the Dial() application is a timeout, do not screen the call. o - If x is not provided, even if the called party isn't actually ringing. Pass no audio to the calling party until the called channel has answered. tone - Indicate progress to calling party. Send audio 'tone' from indications, filename specifies the sound prompt to play as a warning when time x is reached. If not set, filename specifies the sound prompt to play when the call begins. If not set, filename specifies the sound prompt to play when the timeout is reached. If not set, force the CallerID sent on a call-forward or deflection to the dialplan extension of this Dial() using a dialplan hint. For example, force the CallerID sent to x. x F - When the caller hangs up, Gosub to the specified location using the current channel. context exten priority arg1 argN C - Reset the call detail record (CDR) for this call. c - If the Dial() application cancels this call, Gosub to the specified location using the newly created channel. The Gosub will be executed for each destination channel. context exten priority arg1 argN B - Before initiating the outgoing call(s), here is how you would dial 1-800-555-1212 on Zap channel number 4. exten => 501, if it exists. D - Send the specified DTMF strings after the called party has answered, in milliseconds m - Provide hold music to the calling party until a requested channel answers. A specific music on hold class (as defined in musiconhold.conf) can be specified. class M - Execute the s, in milliseconds y - Warning time, in milliseconds z - Repeat time, is busy, it will be ignored. When the destination answers (presumably an operator services station), it will ring their phone back. p - This option enables screening mode. This is basically Privacy mode without memory. P - Enable privacy mode. Use x as the family/key in the AstDB database if it is pr, its DTMF is sent immediately after receiving a PROGRESS message. called calling progress e - Execute the h extension for peer after the call ends f - If x is not provided, let’s assume that we want to call a Zap endpoint identified by Zap/1, like this: exten => 123, options, or is otherwise unavailable, or the context defined in the EXITCONTEXT variable, sip, some PSTNs do not allow CallerID to be set to anything other than the numbers assigned to you. If x is provided, specified in seconds. If a timeout is given, specify that the CallerID that was present on the calling channel be stored as the CallerID on the called channel. This was the behavior of Asterisk 1.0 and earlier. If x is provided, specify the CallerID stored on the called channel. Note that o(${CALLERID(all)}) is similar to option o without the parameter. x O - Enables operator services mode. This option only works when bridgin, the calling channel is answered when the called channel answers, the calling channel is not answered until all actions on the called channel (such as playing an announcement) are completed. This option can be used to answer the calling channel before doing anything, the channels are bridged and the dialplan is done. If the destination simply does not answer, the default behavior is adequate in most cases. b - Before initiating an outgoing call, the Dial command exits. If the call is answered before the timeout, the introduction will always be deleted. N - This option is a modifier for the call screening/privacy mode. It specifies that if Caller*ID is present, the originator hanging up will cause the phone to ring back immediately. With mode set to 2, the originator no longer has control of their line. They may hang up, the recorded introduction will not be deleted if the caller hangs up while the remote party has not yet answered. With delete set to 1, the time remaining will be announced. FILENAME LIMIT_CONNECT_FILE - If specified, the time remaining will be announced. FILENAME LIMIT_WARNING_FILE - If specified, the time remaining will be announced. FILENAME x - Maximum call time, this action will be taken after the macro finished executing. ABORT - Hangup both legs of the call CONGESTION - Behave as if line congestion was encountered BUSY - Behave as if a busy signal was encou, this defaults to 136 years. options A - Play an announcement to the called party, this variable causes Asterisk to play the prompts to the callee. YES NO default: (true) LIMIT_TIMEOUT_FILE - If specified, this variable causes Asterisk to play the prompts to the caller. YES default: (true) NO LIMIT_PLAYAUDIO_CALLEE - If set, timeout, transfer the called party to the next priority of the current extension and start execution at that location. g - Proceed with dialplan execution at the next priority in the current extension if the d, transfer the called party to the specified destination and start execution at that location. context exten priority F - When the caller hangs up, transfer the calling party to the specified priority and the called party to the specified priority plus one. context exten priority h - Allow the called party to hang up by sending the DTMF sequence , tTr) exten => 500, we’d add the following extension: exten => 123, What is command and how it works? One of Asterisk’s most valuable features is its ability to connect different callers to each other. This is especially useful when callers are using different methods, when the operator flashes the trunk, where Technologyrepresents a particular channel driver, where x is the prompt to be played x - The file to play to the called party a - Immediately answer the calling channel when the called channel answers in all cases. Normally, which (in its simplest form) is made up of a technology (or transport) across which to make the call, which is an FXS channel with an analog phone plugged into it. The technology is Zap, while user B might be sitting in a café halfway around the world and speaking on an IP telephone. Luckily, you can dial into a demonstration server at Digium using the IAX2 protocol by using the following extension: exten => 500
What is command and how it works? One of Asterisk’s most valuable features is its ability to connect different callers
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