[asterisk-commits] mvanbaak: branch mvanbaak/appdocsxml r119477 - /team/mvanbaak/appdocsxml/apps/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Sun Jun 1 15:30:24 CDT 2008


Author: mvanbaak
Date: Sun Jun  1 15:30:23 2008
New Revision: 119477

URL: http://svn.digium.com/view/asterisk?view=rev&rev=119477
Log:
more changes to come up with a 'perfect' format

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=119477&r1=119476&r2=119477
==============================================================================
--- team/mvanbaak/appdocsxml/apps/app_dial.c (original)
+++ team/mvanbaak/appdocsxml/apps/app_dial.c Sun Jun  1 15:30:23 2008
@@ -62,14 +62,13 @@
 #include "asterisk/global_datastores.h"
 #include "asterisk/dsp.h"
 
-static char *app = "Dial";
-
-static char *synopsis = "Place a call and connect to the current channel";
-
 /*** APP
 	<application name="Dial">
+		<syntax>
+			Dial(Technology/resourse[&Tech2/resource2...][,timout][,options][,URL])
+		</syntax>
 		<synopsis>
-			Dial(Technology/resourse[&Tech2/resource2...][,timout][,options][,URL])
+			Place a call and connect to the current channel.
 		</synopsis>
 		<description>
 			This application will place calls to one or more specified channels. As soon
@@ -81,7 +80,39 @@
 			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.
+			This application will report normal termination if the originating channel
+			hangs up, or if the call is bridged and either of the parties in the bridge
+			ends the call.
+			If the OUTBOUND_GROUP variable is set, all peer channels created by this
+			application will be put into that group (as in Set(GROUP()=...).
+			If the OUTBOUND_GROUP_ONCE variable is set, all peer channels created by this
+			application will be put into that group (as in Set(GROUP()=...). Unlike OUTBOUND_GROUP,
+			however, the variable will be unset after use.
 		</description>
+		<variable name="DIALEDTIME">
+			This is the time from dialing a channel until when it is disconnected.
+		</variable>
+		<variable name="ANSWEREDTIME">
+			This is the amount of time for actual call.
+		</variable>
+		<variable name="DIALSTATUS">
+			This is the status of the call
+			<value name="CHANUNAVAIL" />
+			<value name="CONGESTION" />
+			<value name="NOANSWER" />
+			<value name="BUSY" />
+			<value name="ANSWER" />
+			<value name="CANCEL" />
+			<value name="DONTCALL">
+				For the Privacy and Screening Modes.
+				Will be set if the called party chooses to send the calling partey to the 'Go Away' script.
+			</value>
+			<value name="TORTURE">
+				For the Privacy and Screening Modes.
+				Will be set if the called party chooses to send the calling partey to the 'torture' script.
+			</value>
+			<value name="INVALIDARGS" />
+		</variable>
 		<option name="A">
 			<argument name="x" required="true">
 				The file to play to the called party
@@ -250,7 +281,7 @@
 		</option>
 		<option name="O">
 			<argument name="x" />
-			Operator Services\" mode (Zaptel channel to Zaptel channel
+			"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.
@@ -258,7 +289,7 @@
 			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
+			when the "operator" flashes the trunk, it will ring their phone
 			back.
 		</option>
 		<option name="p">
@@ -339,8 +370,15 @@
 			Allow the calling party to enable recording of the call by sending
 			the DTMF sequence defined for one-touch automixmonitor in features.conf
 		</option>
+		<option name="URL">
+			The optional URL will be sent to the called party if the channel supports it.
+		</option>
 	</application>
  ***/
+
+static char *app = "Dial";
+
+static char *synopsis = "Place a call and connect to the current channel";
 
 static char *descrip =
 "  Dial(Technology/resource[&Tech2/resource2...][,timeout][,options][,URL]):\n"




More information about the asterisk-commits mailing list