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"
git migration: Refactor the ASTERISK_FILE_VERSION macro Git does not support the ability to replace a token with a version string during check-in. While it does have support for replacing a token on clone, this is somewhat sub-optimal: the token is replaced with the object hash, which is not particularly easy for human consumption. What's more, in practice, the source file version was often not terribly useful. Generally, when triaging bugs, the overall version of Asterisk is far more useful than an individual SVN version of a file. As a result, this patch removes Asterisk's support for showing source file versions. Specifically, it does the following: * Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and remove passing the version in with the macro. Other facilities than 'core show file version' make use of the file names, such as setting a debug level only on a specific file. As such, the act of registering source files with the Asterisk core still has use. The macro rename now reflects the new macro purpose. * main/asterisk: - Refactor the file_version structure to reflect that it no longer tracks a version field. - Remove the "core show file version" CLI command. Without the file version, it is no longer useful. - Remove the ast_file_version_find function. The file version is no longer tracked. - Rename ast_register_file_version/ast_unregister_file_version to ast_register_file/ast_unregister_file, respectively. * main/manager: Remove value from the Version key of the ModuleCheck Action. The actual key itself has not been removed, as doing so would absolutely constitute a backwards incompatible change. However, since the file version is no longer tracked, there is no need to attempt to include it in the Version key. * UPGRADE: Add notes for: - Modification to the ModuleCheck AMI Action - Removal of the "core show file version" CLI command Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e
2015-04-12 02:38:22 +00:00
ASTERISK_REGISTER_FILE()
#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: https://origsvn.digium.com/svn/asterisk/trunk@176627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@176627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17 20:41:24 +00:00
if (!ast_strlen_zero(arglist.filename)) {
ts = ast_get_indication_tone(ast_channel_zone(chan), arglist.filename);
}
}
if (ast_channel_state(chan) != 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 = ast_channel_pbx(chan) ? ast_channel_pbx(chan)->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: https://origsvn.digium.com/svn/asterisk/trunk@176627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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");