[svn-commits] mvanbaak: branch mvanbaak/appdocsxml r119476 - /team/mvanbaak/appdocsxml/apps/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Sun Jun 1 15:04:28 CDT 2008
Author: mvanbaak
Date: Sun Jun 1 15:04:27 2008
New Revision: 119476
URL: http://svn.digium.com/view/asterisk?view=rev&rev=119476
Log:
first shot at a format to document applications.
I could get all of Dial in it so it must at least a step in the correct direction
Modified:
team/mvanbaak/appdocsxml/apps/app_dial.c
Modified: team/mvanbaak/appdocsxml/apps/app_dial.c
URL: http://svn.digium.com/view/asterisk/team/mvanbaak/appdocsxml/apps/app_dial.c?view=diff&rev=119476&r1=119475&r2=119476
==============================================================================
--- team/mvanbaak/appdocsxml/apps/app_dial.c (original)
+++ team/mvanbaak/appdocsxml/apps/app_dial.c Sun Jun 1 15:04:27 2008
@@ -65,6 +65,282 @@
static char *app = "Dial";
static char *synopsis = "Place a call and connect to the current channel";
+
+/*** APP
+ <application name="Dial">
+ <synopsis>
+ Dial(Technology/resourse[&Tech2/resource2...][,timout][,options][,URL])
+ </synopsis>
+ <description>
+ This application will place calls to one or more specified channels. As soon
+ as one of the requested channels answers, the originating channel will be
+ answered, if it has not already been answered. These two channels will then
+ be active in a bridged call. All other channels that were requested will then
+ be hung up.
+ Unless there is a timeout specified, the Dial application will wait
+ indefinitely until one of the called channels answers, the user hangs up, or
+ if all of the called channels are busy or unavailable. Dialplan executing will
+ continue if no requested channels can be called, or if the timeout expires.
+ </description>
+ <option name="A">
+ <argument name="x" required="true">
+ The file to play to the called party
+ </argument>
+ Play an announcement to the called party, using 'x' as the file
+ </option>
+ <option name="C">
+ Reset the CDR for this call.
+ </option>
+ <option name="c">
+ If DIAL cancels this call, always set the flag to tell the channel
+ driver that the call is answered elsewhere.
+ <option name="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 context, or the context defined in the EXITCONTEXT variable,
+ if it exists.
+ </option>
+ <option name="D" argsep=":">
+ <argument name="called" />
+ <argument name="calling" />
+ Send the specified DTMF strings *after* the called\n
+ party has answered, but before the call gets bridged. The 'called'
+ DTMF string is sent to the called party, and the 'calling' DTMF
+ string is sent to the calling party. Both parameters can be used
+ alone.
+ </option>
+ <option name="e">
+ execute the 'h' extension for peer after the call ends
+ </option>
+ <option name="f">
+ Force the callerid of the *calling* channel to be set as the
+ extension associated with the channel using a dialplan 'hint'.
+ For example, some PSTNs do not allow CallerID to be set to anything
+ other than the number assigned to the caller.
+ </option>
+ <option name="F" argsep="^">
+ <argument name="context" />
+ <argument name="exten" />
+ <argument name="pri" required="true" />
+ When the caller hangs up, transfer the called party
+ to the specified context and extension and continue execution.
+ </option>
+ <option name="g">
+ Proceed with dialplan execution at the current extension if the
+ destination channel hangs up.
+ </option>
+ <option name="G" argsep="^">
+ <argument name="context" />
+ <argument name="exten" />
+ <argument name="pri" required="true" />
+ If the call is answered, transfer the calling party to
+ the specified priority and the called party to the specified priority+1.
+ Optionally, an extension, or extension and context may be specified.
+ Otherwise, the current extension is used. You cannot use any additional
+ action post answer options in conjunction with this option.
+ </option>
+ <option name="h">
+ Allow the called party to hang up by sending the '*' DTMF digit.
+ </option>
+ <option name="H">
+ Allow the calling party to hang up by hitting the '*' DTMF digit.
+ </option>
+ <option name="i">
+ Asterisk will ignore any forwarding requests it may receive on this
+ dial attempt.
+ </option>
+ <option name="k">
+ Allow the called party to enable parking of the call by sending
+ the DTMF sequence defined for call parking in features.conf.
+ </option>
+ <option name="K">
+ Allow the calling party to enable parking of the call by sending
+ the DTMF sequence defined for call parking in features.conf.
+ </option>
+ <option name="L" args="x,y,z" argsep=":">
+ <argument name="x" required="true">
+ Maximum calltime in miliseconds
+ </argument>
+ <argument name="y" />
+ <argument name="z" />
+ Limit the call to 'x' ms. Play a warning when 'y' ms are
+ left. Repeat the warning every 'z' ms.
+ <variable name="LIMIT_PLAYAUDIO_CALLER">
+ <value name="yes" default="true" />
+ <value name="no" />
+ Play sounds to the caller.
+ </variable>
+ <variable name="LIMIT_PLAYAUDIO_CALLEE">
+ <value name="yes" />
+ <value name="no" />
+ Play sounds to the callee.
+ </variable>
+ <variable name="LIMIT_TIMEOUT_FILE">
+ <value name="filename">
+ If not set, the time remaining will be said.
+ </value>
+ File to play when time is up.
+ </variable>
+ <variable name="LIMIT_CONNECT_FILE">
+ <value name="filename">
+ If not set, the time remaining will be said.
+ </value>
+ File to play when call begins.
+ </variable>
+ <variable name="LIMIT_WARNING_FILE">
+ <value name="filename">
+ If not set, the time remaining will be said.
+ </value>
+ File to play as warning if 'y' is defined.
+ <variable>
+ </option>
+ <option name="m">
+ <argument name="class" />
+ Provide hold music to the calling party until a requested
+ channel answers. A specific MusicOnHold class can be
+ specified.
+ </option>
+ <option name="M" args="x,arg" argsep="^">
+ <argument name="x" required="true">
+ Macro name that should be executed.
+ </argument>
+ <argument name="arg">
+ Macro arguments seperated by ^
+ </argument>
+ Execute the Macro for the *called* channel before connecting
+ to the calling channel. Arguments can be specified to the Macro
+ using '^' as a delimiter. The Macro can set the variable
+ MACRO_RESULT to specify the following actions after the Macro is
+ finished executing.
+ <variable name="MACRO_RESULT">
+ If set, this action will be taken after the macro finished executing.
+ <value name="ABORT">
+ Hangup both legs of the call.
+ </value>
+ <value name="CONGESTION">
+ Behave as if line congestion was encountered.
+ </value>
+ <value name="BUSY">
+ Behave as if a busy signal was encountered.
+ </value>
+ <value name="CONTINUE">
+ Hangup the called party and allow the calling party to continue dialplan execution at the next priority.
+ </value>
+ <value name="GOTO:<context>^<exten>^<priority>">
+ Transfer the call to the specified priority. Optionally, an extension, or extension and priority can be specified.
+ </value>
+ </variable>
+ You cannot use any additional action post answer options in conjunction
+ with this option. Also, pbx services are not run on the peer (called) channel,
+ so you will not be able to set timeouts via the TIMEOUT() function in this macro.
+ </option>
+ <option name="n">
+ This option is a modifier for the screen/privacy mode. It specifies
+ that no introductions are to be saved in the priv-callerintros
+ directory.
+ </option>
+ <option name="N">
+ This option is a modifier for the screen/privacy mode. It specifies
+ that if callerID is present, do not screen the call.
+ </option>
+ <option name="o">
+ Specify that the CallerID that was present on the *calling* channel
+ be set as the CallerID on the *called* channel. This was the
+ behavior of Asterisk 1.0 and earlier.
+ </option>
+ <option name="O">
+ <argument name="x" />
+ Operator Services\" mode (Zaptel channel to Zaptel channel
+ only, if specified on non-Zaptel interface, it will be ignored).
+ When the destination answers (presumably an operator services
+ station), the originator no longer has control of their line.
+ They may hang up, but the switch will not release their line
+ until the destination party hangs up (the operator). Specified
+ without an arg, or with 1 as an arg, the originator hanging up
+ will cause the phone to ring back immediately. With a 2 specified,
+ when the \"operator\" flashes the trunk, it will ring their phone
+ back.
+ </option>
+ <option name="p">
+ This option enables screening mode. This is basically Privacy mode
+ without memory.
+ </option>
+ <option name="P">
+ <argument name="x" />
+ Enable privacy mode. Use 'x' as the family/key in the database if
+ it is provided. The current extension is used if a database
+ family/key is not specified.
+ </option>
+ <option name="r">
+ Indicate ringing to the calling party. Pass no audio to the calling
+ party until the called channel has answered.
+ </option>
+ <option name="S">
+ <argument name="x" required="true" />
+ Hang up the call after 'x' seconds *after* the called party has
+ answered the call.
+ </option>
+ <option name="t">
+ Allow the called party to transfer the calling party by sending the
+ DTMF sequence defined in features.conf.
+ </option>
+ <option name="T">
+ Allow the calling party to transfer the called party by sending the
+ DTMF sequence defined in features.conf.
+ </option>
+ <option name="U" argsep="^">
+ <argument name="x" required="true">
+ routine to execute via Gosub
+ </argument>
+ <argument name="arg">
+ Arguments for the Gosub routine
+ </argument>
+ Execute via Gosub the routine 'x' for the *called* channel before connecting
+ to the calling channel. Arguments can be specified to the Gosub
+ using '^' as a delimiter. The Gosub routine can set the variable
+ GOSUB_RESULT to specify the following actions after the Gosub returns.
+ <variable name="GOSUB_RESULT">
+ <value name="ABORT">
+ Hangup both legs of the call.
+ </value>
+ <value name="CONGESTION">
+ Behave as if line congestion was encountered.
+ </value>
+ <value name="BUSY">
+ Behave as if a busy signal was encountered.
+ </value>
+ <value name="CONTINUE">
+ Hangup the called party and allow the calling party
+ to continue dialplan execution at the next priority.
+ </value>
+ <value name="GOTO:<context>^<exten>^<priority>">
+ Transfer the call to the
+ specified priority. Optionally, an extension, or
+ extension and priority can be specified.
+ </value>
+ </variable>
+ You cannot use any additional action post answer options in conjunction
+ with this option. Also, pbx services are not run on the peer (called) channel,
+ so you will not be able to set timeouts via the TIMEOUT() function in this routine.
+ </option>
+ <option name="w">
+ Allow the called party to enable recording of the call by sending
+ the DTMF sequence defined for one-touch recording in features.conf.
+ </option>
+ <option name="W">
+ Allow the calling party to enable recording of the call by sending
+ the DTMF sequence defined for one-touch recording in features.conf.
+ </option>
+ <option name="x">
+ Allow the called party to enable recording of the call by sending
+ the DTMF sequence defined for one-touch automixmonitor in features.conf
+ </option>
+ <option name="X">
+ Allow the calling party to enable recording of the call by sending
+ the DTMF sequence defined for one-touch automixmonitor in features.conf
+ </option>
+ </application>
+ ***/
static char *descrip =
" Dial(Technology/resource[&Tech2/resource2...][,timeout][,options][,URL]):\n"
More information about the svn-commits
mailing list