A(x) - With this opition you can set an audio file which
the called person to hear. In the brackets you put the name of the file. C - This option allows you to reset the Call Detail Records(CDR). Be very
careful with this option. CDR`s are very important because they are used for
billing information. D([digits]) - this option allows you to send a DTMF tones which the called
person will receive after he/she answers. f - forces callerid to be set as the extension of the line making/redirecting
the outgoing call. g - when the called party hangs up, exit to execute more commands in
the current context. h - this option allows the called person to hang up the line by pressing "*" H - this option allows the caller to hang up the line by pressing "*" j - this option prevetns jumping to an extension n+101 L(x[:y][:z]) -
m - when you make a call you will hear a music on hold instead of the ringing
signal. You can use m(x) - where x is a specified class for the music on hold M(x) - Executes the macro (x) upon connect of the call P[(x)] -This option allows you to use the PrivacyManager. You can use as agrument a name of a database.
However, the argument is optional. r - S(x) - With this option you can specify a period of time after which the line
will be hung up. The timer starts after the called person answered the call t - this option allows the called person to transfer the call T - this option allows the caller to transfer the call w - with this option the called user can start recording the conversation after pressing *1. This combination of symbols can be changed in features.conf. This is valid for Asterisk PBX version 1.0.X or later. W - with this option the caller can start recording the conversation after pressing *1. This combination of symbols can be changed in features.conf.
This is valid for Asterisk PBX version 1.0.X or later.
Purpose and usage
Perhaps this is the most commonly used application. Its purpose is to place calls on a channel. It tries to do this on all available channels from the specified protocol(IAX, SIP, MGCP or ZAP). The call will be placed on the first one which answers.
You can make calls on more than one available channels simultaneously. You have just to list the desired technology/resource one by one separated by the "&" symbol
To see how the application works we recommend to use our IAX softphone Idefisk. You can download it from here. Please also read our tutorial to learn how to configure it to work with Asterisk PBX.
Asterisk PBX configurations
NOTE: This is only an example of what for you can use this application. Of course you can use it and for other things.
We need two registered user in iax.conf file. This is because we are going to use the IAX2 protocol. If you want to use other protocol such as SIP or MGCP, you have to do the configurations below respectively in sip.conf or mgcp.conf.
So, we have registered the users anatoliy and user1.
Type=friend means that this user can make and receive calls. Host=dynamic means that the IP is not static but dynamic through a DHCP server. Allow=all means that the line which this user will use, could support all audio codecs. Context=test - this shows that this user is working with the extensions in this context of the configuration file extensions.conf.
We will explain you how to configure the extensions.conf for each of the options and how it works.
In all examples we will use two extensions. The first one contains the Dial application. It has a built-in Answer application. This allows the Asterisk PBX to answer automatically. So, you don`t need a separate extension for Answer.
The second extension is for hanging up. It is always a good idea to make such extensions to be sure that the Asterisk PBX will hang up the line after the conversation is over.
1) Option - A(x)
On the picture above you can see our extension.conf file. We have two extensions in the context [test].
As you see we have chosen to dial the user anatoliy through IAX2 channel and we have set the option A() with argument welcome. welcome represents the audio file which we would like the called user to hear when he/she answers the receiving call.
2) Option - C
The option here makes the following: it resets the Call Detail Records(CDR). CDR is very important because it contains information about the calls - where are they originating from, terminating at or passing through the exchange. This information is then used for billing. So, you have to be very careful when you age going to work with CDR. An especially with this option from the Dial application.
3) Option - D([digits])
The option is for the following: it will send a DTMF tone or tones when the called user answers the receving call. In our example we will send the DTMF tones 098 which are specified in the brackets of the option D.
4) Option - f
The purpose of this option is to change the callerid as the context(in extensions.conf) of the line making/redirecting the outgoing call. For example - you can set your callerid to user1 through your softphone/hardphone. Now this application will change it to the context(in extensions.conf) from which you are making the call. In our case this will be [test].
5) Option - g
6) Option - h
With this option you can allow the called person to hang up the line not only with the red button(Hang up) but also by pressing "*". That is all you can do with this option.
7) Option - H
This option is almost equal to the previous one. With its help you can allow the caller to hang up the line not only with the red button(Hang up) but also by pressing "*".
8) Option - j
When all channels which the Dial application tries to call are busy then the next extension for execution will be the one with priority n+101. Now, with this option, set in the brackets of the Dial application, you can prevent this jump and the next executed extension will be the one with priority n+1. n is the priority of the instant extension.
9) Option - L(x[:y][:z])
Wth this option you can limit the time of the conversation. The first argument in the brackets is for the length of the conversation. In the next argument separated by ":" you set - when 'y' ms are left the caller to hear a warning message. The third argument is again separated by ":". Here you can set at intervals of how many ms the caller to hear an mesage with the rest of the time he/she has for this conversation.
NOTE: The time options are in "ms" so be careful.
In our example we have set 10000 or 10 seconds for the length of the conversation, 2000 - 2 seconds before the time for the conversation to expire the caller will hear a warning, 5000 - every 5 seconds the caller will hear a message wih the time left.
10) Option - m
The purpose of this option is to play a musiconhold sound instead of the standart ringing signal. You can use this option with an argument.
Example:
m(class). In the brackets you can specify the desired class for musiconhold.
The file where we have to set the path to the music folder is musiconhold.conf. By default the folder is /var/lib/asterisk/mohmp3. The supported format for audio files is mp3.
11) Option - M(x)
[!imag internal][/!imag]
12) Option - P[(x)]
[!imag internal][/!imag]
13) Option - r
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devendra jain (devendra dot jain at rediffmail dot com) 26 September 2006 08:28:35 provide more information about m(x) to use it as PRBT
ahz (ahz at a2zvoip dot com) 06 September 2006 19:14:23 regarding the dial command L
Wth this option you can limit the time of the conversation. The first argument in the brackets is for the length of the conversation. In the next argument separated by ":" you set - when 'y' ms are left the caller to hear a warning message. The third argument is again separated by ":". Here you can set at intervals of how many ms the caller to hear an mesage with the rest of the time he/she has for this conversation.
NOTE: The time options are in "ms" so be careful.
In our example we have set 10000 or 10 seconds for the length of the conversation, 2000 - 2 seconds before the time for the conversation to expire the caller will hear a warning, 5000 - every 5 seconds the caller will hear a message wih the time left.
which will terminate the call correctly BUT what file is spoken for the start and end of call???
botham (bipin_bipin81 at sify dot com) 16 August 2006 11:22:04 give more information about sip.conf.
and user type sip configuration
Fernanda Calvo (fernada at redcetus dot com) 22 February 2006 16:58:15 Where is S(x) - With this option you can specify a period of time after which the line will be hung up. The timer starts after the called person answered the call