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asterisk/apps/app_read.c

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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 1999 - 2005, Digium, Inc.
*
* Mark Spencer <markster@digium.com>
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/*! \file
*
* \brief Trivial application to read a variable
*
* \author Mark Spencer <markster@digium.com>
*
* \ingroup applications
*/
/*** MODULEINFO
<support_level>core</support_level>
***/
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/file.h"
#include "asterisk/pbx.h"
#include "asterisk/channel.h"
#include "asterisk/app.h"
#include "asterisk/module.h"
#include "asterisk/indications.h"
/*** DOCUMENTATION
<application name="Read" language="en_US">
<synopsis>
Read a variable.
</synopsis>
<syntax>
<parameter name="variable" required="true">
<para>The input digits will be stored in the given <replaceable>variable</replaceable>
name.</para>
</parameter>
<parameter name="filenames" argsep="&amp;">
<argument name="filename" required="true">
<para>file(s) to play before reading digits or tone with option i</para>
</argument>
<argument name="filename2" multiple="true" />
</parameter>
<parameter name="maxdigits">
<para>Maximum acceptable number of digits. Stops reading after
<replaceable>maxdigits</replaceable> have been entered (without
requiring the user to press the <literal>#</literal> key).</para>
<para>Defaults to <literal>0</literal> - no limit - wait for the
user press the <literal>#</literal> key. Any value below
<literal>0</literal> means the same. Max accepted value is
<literal>255</literal>.</para>
</parameter>
<parameter name="options">
<optionlist>
<option name="s">
<para>to return immediately if the line is not up.</para>
</option>
<option name="i">
<para>to play filename as an indication tone from your
<filename>indications.conf</filename>.</para>
</option>
<option name="n">
<para>to read digits even if the line is not up.</para>
</option>
</optionlist>
</parameter>
<parameter name="attempts">
<para>If greater than <literal>1</literal>, that many
<replaceable>attempts</replaceable> will be made in the
event no data is entered.</para>
</parameter>
<parameter name="timeout">
<para>The number of seconds to wait for a digit response. If greater
than <literal>0</literal>, that value will override the default timeout.
Can be floating point.</para>
</parameter>
</syntax>
<description>
<para>Reads a #-terminated string of digits a certain number of times from the
user in to the given <replaceable>variable</replaceable>.</para>
<para>This application sets the following channel variable upon completion:</para>
<variablelist>
<variable name="READSTATUS">
<para>This is the status of the read operation.</para>
<value name="OK" />
<value name="ERROR" />
<value name="HANGUP" />
<value name="INTERRUPTED" />
<value name="SKIPPED" />
<value name="TIMEOUT" />
</variable>
</variablelist>
</description>
<see-also>
<ref type="application">SendDTMF</ref>
</see-also>
</application>
***/
enum read_option_flags {
OPT_SKIP = (1 << 0),
OPT_INDICATION = (1 << 1),
OPT_NOANSWER = (1 << 2),
};
AST_APP_OPTIONS(read_app_options, {
AST_APP_OPTION('s', OPT_SKIP),
AST_APP_OPTION('i', OPT_INDICATION),
AST_APP_OPTION('n', OPT_NOANSWER),
});
static char *app = "Read";
static int read_exec(struct ast_channel *chan, const char *data)
{
int res = 0;
char tmp[256] = "";
int maxdigits = 255;
int tries = 1, to = 0, x = 0;
double tosec;
char *argcopy = NULL;
Merge a large set of updates to the Asterisk indications API. This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@176627 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17 20:41:24 +00:00
struct ast_tone_zone_sound *ts = NULL;
struct ast_flags flags = {0};
const char *status = "ERROR";
AST_DECLARE_APP_ARGS(arglist,
AST_APP_ARG(variable);
AST_APP_ARG(filename);
AST_APP_ARG(maxdigits);
AST_APP_ARG(options);
AST_APP_ARG(attempts);
AST_APP_ARG(timeout);
);
pbx_builtin_setvar_helper(chan, "READSTATUS", status);
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "Read requires an argument (variable)\n");
return 0;
}
argcopy = ast_strdupa(data);
AST_STANDARD_APP_ARGS(arglist, argcopy);
if (!ast_strlen_zero(arglist.options)) {
ast_app_parse_options(read_app_options, &flags, NULL, arglist.options);
}
if (!ast_strlen_zero(arglist.attempts)) {
tries = atoi(arglist.attempts);
if (tries <= 0)
tries = 1;
}
if (!ast_strlen_zero(arglist.timeout)) {
tosec = atof(arglist.timeout);
if (tosec <= 0)
to = 0;
else
to = tosec * 1000.0;
}
if (ast_strlen_zero(arglist.filename)) {
arglist.filename = NULL;
}
if (!ast_strlen_zero(arglist.maxdigits)) {
maxdigits = atoi(arglist.maxdigits);
if ((maxdigits < 1) || (maxdigits > 255)) {
maxdigits = 255;
} else
ast_verb(3, "Accepting a maximum of %d digits.\n", maxdigits);
}
if (ast_strlen_zero(arglist.variable)) {
ast_log(LOG_WARNING, "Invalid! Usage: Read(variable[,filename][,maxdigits][,option][,attempts][,timeout])\n\n");
return 0;
}
if (ast_test_flag(&flags, OPT_INDICATION)) {
Merge a large set of updates to the Asterisk indications API. This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@176627 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17 20:41:24 +00:00
if (!ast_strlen_zero(arglist.filename)) {
ts = ast_get_indication_tone(chan->zone, arglist.filename);
}
}
if (chan->_state != AST_STATE_UP) {
if (ast_test_flag(&flags, OPT_SKIP)) {
/* At the user's option, skip if the line is not up */
pbx_builtin_setvar_helper(chan, arglist.variable, "");
pbx_builtin_setvar_helper(chan, "READSTATUS", "SKIPPED");
return 0;
} else if (!ast_test_flag(&flags, OPT_NOANSWER)) {
/* Otherwise answer unless we're supposed to read while on-hook */
res = ast_answer(chan);
}
}
if (!res) {
while (tries && !res) {
ast_stopstream(chan);
if (ts && ts->data[0]) {
if (!to)
to = chan->pbx ? chan->pbx->rtimeoutms : 6000;
res = ast_playtones_start(chan, 0, ts->data, 0);
for (x = 0; x < maxdigits; ) {
res = ast_waitfordigit(chan, to);
ast_playtones_stop(chan);
if (res < 1) {
if (res == 0)
status = "TIMEOUT";
tmp[x]='\0';
break;
}
tmp[x++] = res;
if (tmp[x-1] == '#') {
tmp[x-1] = '\0';
status = "OK";
break;
}
if (x >= maxdigits) {
status = "OK";
}
}
} else {
res = ast_app_getdata(chan, arglist.filename, tmp, maxdigits, to);
if (res == AST_GETDATA_COMPLETE || res == AST_GETDATA_EMPTY_END_TERMINATED)
status = "OK";
else if (res == AST_GETDATA_TIMEOUT)
status = "TIMEOUT";
else if (res == AST_GETDATA_INTERRUPTED)
status = "INTERRUPTED";
}
if (res > -1) {
pbx_builtin_setvar_helper(chan, arglist.variable, tmp);
if (!ast_strlen_zero(tmp)) {
ast_verb(3, "User entered '%s'\n", tmp);
tries = 0;
} else {
tries--;
if (tries)
ast_verb(3, "User entered nothing, %d chance%s left\n", tries, (tries != 1) ? "s" : "");
else
ast_verb(3, "User entered nothing.\n");
}
res = 0;
} else {
pbx_builtin_setvar_helper(chan, arglist.variable, tmp);
ast_verb(3, "User disconnected\n");
}
}
}
Merge a large set of updates to the Asterisk indications API. This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@176627 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17 20:41:24 +00:00
if (ts) {
ts = ast_tone_zone_sound_unref(ts);
}
if (ast_check_hangup(chan))
status = "HANGUP";
pbx_builtin_setvar_helper(chan, "READSTATUS", status);
return 0;
}
static int unload_module(void)
{
return ast_unregister_application(app);
}
static int load_module(void)
{
return ast_register_application_xml(app, read_exec);
}
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Read Variable Application");