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update application documentation dump

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@72920 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Russell Bryant
2007-07-02 16:42:17 +00:00
parent 73d33590ba
commit ee45b352e7
1 changed files with 129 additions and 15 deletions
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144
doc/ast_appdocs.tex
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@@ -1,4 +1,4 @@
% This file is automatically generated. Any manual edits will be lost.
% This file is automatically generated by the "core dump appdocs" CLI command. Any manual edits will be lost.
\section{AddQueueMember}
\subsection{Synopsis}
\begin{verbatim}
@@ -99,7 +99,7 @@ on file descriptor 3
Use the CLI command 'agi show' to list available agi commands
This application sets the following channel variable upon completion:
AGISTATUS The status of the attempt to the run the AGI script
text string, one of SUCCESS | FAILED | HANGUP
text string, one of SUCCESS | FAILED | NOTFOUND | HANGUP
\end{verbatim}
@@ -530,7 +530,7 @@ on file descriptor 3
Use the CLI command 'agi show' to list available agi commands
This application sets the following channel variable upon completion:
AGISTATUS The status of the attempt to the run the AGI script
text string, one of SUCCESS | FAILED | HANGUP
text string, one of SUCCESS | FAILED | NOTFOUND | HANGUP
\end{verbatim}
@@ -601,6 +601,10 @@ however, the variable will be unset after use.
H - Allow the calling party to hang up by hitting the '*' DTMF digit.
i - Asterisk will ignore any forwarding requests it may receive on this
dial attempt.
k - Allow the called party to enable parking of the call by sending
the DTMF sequence defined for call parking in features.conf.
K - Allow the calling party to enable parking of the call by sending
the DTMF sequence defined for call parking in features.conf.
L(x[:y][:z]) - Limit the call to 'x' ms. Play a warning when 'y' ms are
left. Repeat the warning every 'z' ms. The following special
variables can be used with this option:
@@ -662,14 +666,25 @@ however, the variable will be unset after use.
DTMF sequence defined in features.conf.
T - Allow the calling party to transfer the called party by sending the
DTMF sequence defined in features.conf.
U(x[^arg]) - 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 delimeter. The Gosub routine can set the variable
GOSUB_RESULT to specify the following actions after the Gosub returns.
* ABORT Hangup both legs of the call.
* CONGESTION Behave as if line congestion was encountered.
* BUSY Behave as if a busy signal was encountered.
* CONTINUE Hangup the called party and allow the calling party
to continue dialplan execution at the next priority.
* GOTO:<context>^<exten>^<priority> - Transfer the call to the
specified priority. Optionally, an extension, or
extension and priority can be specified.
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.
w - Allow the called party to enable recording of the call by sending
the DTMF sequence defined for one-touch recording in features.conf.
W - Allow the calling party to enable recording of the call by sending
the DTMF sequence defined for one-touch recording in features.conf.
k - Allow the called party to enable parking of the call by sending
the DTMF sequence defined for call parking in features.conf.
K - Allow the calling party to enable parking of the call by sending
the DTMF sequence defined for call parking in features.conf.
\end{verbatim}
@@ -809,7 +824,7 @@ on file descriptor 3
Use the CLI command 'agi show' to list available agi commands
This application sets the following channel variable upon completion:
AGISTATUS The status of the attempt to the run the AGI script
text string, one of SUCCESS | FAILED | HANGUP
text string, one of SUCCESS | FAILED | NOTFOUND | HANGUP
\end{verbatim}
@@ -1200,6 +1215,19 @@ Set for more information.
\end{verbatim}
\section{IVRDemo}
\subsection{Synopsis}
\begin{verbatim}
IVR Demo Application
\end{verbatim}
\subsection{Description}
\begin{verbatim}
This is a skeleton application that shows you the basic structure to create your
own asterisk applications and demonstrates the IVR demo.
\end{verbatim}
\section{JabberSend}
\subsection{Synopsis}
\begin{verbatim}
@@ -1232,6 +1260,19 @@ JabberStatus(Jabber,ScreenName,Variable)
\end{verbatim}
\section{KeepAlive}
\subsection{Synopsis}
\begin{verbatim}
returns AST_PBX_KEEPALIVE value
\end{verbatim}
\subsection{Description}
\begin{verbatim}
KeepAlive(): This application is chiefly meant for internal use with Gosubs.
Please do not run it alone from the dialplan!
\end{verbatim}
\section{Log}
\subsection{Synopsis}
\begin{verbatim}
@@ -1639,8 +1680,8 @@ Valid options:
(range -4 to 4)
<command> will be executed when the recording is over
Any strings matching ^{X} will be unescaped to ${X} and
all variables will be evaluated at that time.
Any strings matching ^{X} will be unescaped to ${X}.
All variables will be evaluated at the time MixMonitor is called.
The variable MIXMONITOR_FILENAME will contain the filename used to record.
\end{verbatim}
@@ -1998,7 +2039,7 @@ Queue a call for a call queue
\end{verbatim}
\subsection{Description}
\begin{verbatim}
Queue(queuename[|options[|URL][|announceoverride][|timeout][|AGI][|macro]):
Queue(queuename[|options[|URL][|announceoverride][|timeout][|AGI][|macro][|gosub]):
Queues an incoming call in a particular call queue as defined in queues.conf.
This application will return to the dialplan if the queue does not exist, or
any of the join options cause the caller to not enter the queue.
@@ -2023,6 +2064,8 @@ it.
The optional AGI parameter will setup an AGI script to be executed on the
calling party's channel once they are connected to a queue member.
The optional macro parameter will run a macro on the
calling party's channel once they are connected to a queue member.
The optional gosub parameter will run a gosub on the
calling party's channel once they are connected to a queue member.
The timeout will cause the queue to fail out after a specified number of
seconds, checked between each queues.conf 'timeout' and 'retry' cycle.
@@ -2215,6 +2258,50 @@ tone to the user.
\end{verbatim}
\section{Rpt}
\subsection{Synopsis}
\begin{verbatim}
Radio Repeater/Remote Base Control System
\end{verbatim}
\subsection{Description}
\begin{verbatim}
Rpt(nodename[|options]): Radio Remote Link or Remote Base Link Endpoint Process.
Not specifying an option puts it in normal endpoint mode (where source
IP and nodename are verified).
Options are as follows:
X - Normal endpoint mode WITHOUT security check. Only specify
this if you have checked security already (like with an IAX2
user/password or something).
Rannounce-string[|timeout[|timeout-destination]] - Amateur Radio
Reverse Autopatch. Caller is put on hold, and announcement (as
specified by the 'announce-string') is played on radio system.
Users of radio system can access autopatch, dial specified
code, and pick up call. Announce-string is list of names of
recordings, or "PARKED" to substitute code for un-parking,
or "NODE" to substitute node number.
P - Phone Control mode. This allows a regular phone user to have
full control and audio access to the radio system. For the
user to have DTMF control, the 'phone_functions' parameter
must be specified for the node in 'rpt.conf'. An additional
function (cop,6) must be listed so that PTT control is available.
D - Dumb Phone Control mode. This allows a regular phone user to
have full control and audio access to the radio system. In this
mode, the PTT is activated for the entire length of the call.
For the user to have DTMF control (not generally recomended in
this mode), the 'dphone_functions' parameter must be specified
for the node in 'rpt.conf'. Otherwise no DTMF control will be
available to the phone user.
\end{verbatim}
\section{SayAlpha}
\subsection{Synopsis}
\begin{verbatim}
@@ -2466,6 +2553,19 @@ SIPDtmfMode(inband|info|rfc2833): Changes the dtmfmode for a SIP call
\end{verbatim}
\section{Skel}
\subsection{Synopsis}
\begin{verbatim}
Skeleton application.
\end{verbatim}
\subsection{Description}
\begin{verbatim}
This application is a template to build other applications from.
It shows you the basic structure to create your own Asterisk applications.
\end{verbatim}
\section{SLAStation}
\subsection{Synopsis}
\begin{verbatim}
@@ -2473,7 +2573,15 @@ Shared Line Appearance Station
\end{verbatim}
\subsection{Description}
\begin{verbatim}
SLAStation():
SLAStation(station):
This application should be executed by an SLA station. The argument depends
on how the call was initiated. If the phone was just taken off hook, then
the argument "station" should be just the station name. If the call was
initiated by pressing a line key, then the station name should be preceded
by an underscore and the trunk name associated with that line button.
For example: "station1_line1". On exit, this application will set the variable SLASTATION_STATUS to
one of the following values:
FAILURE | CONGESTION | SUCCESS
\end{verbatim}
@@ -2485,7 +2593,13 @@ Shared Line Appearance Trunk
\end{verbatim}
\subsection{Description}
\begin{verbatim}
SLATrunk():
SLATrunk(trunk):
This application should be executed by an SLA trunk on an inbound call.
The channel calling this application should correspond to the SLA trunk
with the name "trunk" that is being passed as an argument.
On exit, this application will set the variable SLATRUNK_STATUS to
one of the following values:
FAILURE | SUCCESS | UNANSWERED | RINGTIMEOUT
\end{verbatim}
@@ -2500,9 +2614,9 @@ Communicates with SMS service centres and SMS capable analogue phones
SMS(name|[a][s][t][p(d)][r][o]|addr|body):
SMS handles exchange of SMS data with a call to/from SMS capable
phone or SMS PSTN service center. Can send and/or receive SMS messages.
Works to ETSI ES 201 912 compatible with BT SMS PSTN service in UK
Works to ETSI ES 201 912; compatible with BT SMS PSTN service in UK
and Telecom Italia in Italy.
Typical usage is to use to handle called from the SMS service centre CLI,
Typical usage is to use to handle calls from the SMS service centre CLI,
or to set up a call using 'outgoing' or manager interface to connect
service centre to SMS()
name is the name of the queue used in /var/spool/asterisk/sms