Files
asterisk/apps/app_page.c
George Joseph a80179bfe7 docs: Add version information to application and function XML elements
* Do a git blame on the embedded XML application or function element.

* From the commit hash, grab the summary line.

* Do a git log --grep <summary> to find the cherry-pick commits in all
  branches that match.

* Do a git patch-id to ensure the commits are all related and didn't get
  a false match on the summary.

* Do a git tag --contains <commit> to find the tags that contain each
  commit.

* Weed out all tags not ..0.

* Sort and discard any .0.0 and following tags where the commit
  appeared in an earlier branch.

* The result is a single tag for each branch where the application or function
  was defined.

The applications and functions defined in the following files were done by
hand because the XML was extracted from the C source file relatively recently.
* channels/pjsip/dialplan_functions_doc.xml
* main/logger_doc.xml
* main/manager_doc.xml
* res/res_geolocation/geoloc_doc.xml
* res/res_stir_shaken/stir_shaken_doc.xml

(cherry picked from commit 85a4ab8390)
2025-01-23 18:39:42 +00:00

455 lines
14 KiB
C

/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (c) 2004 - 2006 Digium, Inc. All rights reserved.
*
* Mark Spencer <markster@digium.com>
*
* This code is released under the GNU General Public License
* version 2.0. See LICENSE for more information.
*
* 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.
*
*/
/*! \file
*
* \brief page() - Paging application
*
* \author Mark Spencer <markster@digium.com>
*
* \ingroup applications
*/
/*** MODULEINFO
<depend>app_confbridge</depend>
<support_level>core</support_level>
***/
#include "asterisk.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/file.h"
#include "asterisk/app.h"
#include "asterisk/chanvars.h"
#include "asterisk/utils.h"
#include "asterisk/devicestate.h"
#include "asterisk/dial.h"
/*** DOCUMENTATION
<application name="Page" language="en_US">
<since><version>1.6.2.0</version></since>
<synopsis>
Page series of phones
</synopsis>
<syntax>
<parameter name="Technology/Resource" required="false" argsep="&amp;">
<argument name="Technology/Resource" required="true">
<para>Specification of the device(s) to dial. These must be in the format of
<literal>Technology/Resource</literal>, where <replaceable>Technology</replaceable>
represents a particular channel driver, and <replaceable>Resource</replaceable> represents a resource
available to that particular channel driver.</para>
</argument>
<argument name="Technology2/Resource2" multiple="true">
<para>Optional extra devices to dial in parallel</para>
<para>If you need more than one, enter them as Technology2/Resource2&amp;
Technology3/Resource3&amp;.....</para>
</argument>
</parameter>
<parameter name="options">
<optionlist>
<option name="b" argsep="^">
<para>Before initiating an outgoing call, Gosub to the specified
location using the newly created channel. The Gosub will be
executed for each destination channel.</para>
<argument name="context" required="false" />
<argument name="exten" required="false" />
<argument name="priority" required="true" hasparams="optional" argsep="^">
<argument name="arg1" multiple="true" required="true" />
<argument name="argN" />
</argument>
</option>
<option name="B" argsep="^">
<para>Before initiating the outgoing call(s), Gosub to the specified
location using the current channel.</para>
<argument name="context" required="false" />
<argument name="exten" required="false" />
<argument name="priority" required="true" hasparams="optional" argsep="^">
<argument name="arg1" multiple="true" required="true" />
<argument name="argN" />
</argument>
</option>
<option name="d">
<para>Full duplex audio</para>
</option>
<option name="i">
<para>Ignore attempts to forward the call</para>
</option>
<option name="q">
<para>Quiet, do not play beep to caller</para>
</option>
<option name="r">
<para>Record the page into a file (<literal>CONFBRIDGE(bridge,record_conference)</literal>)</para>
</option>
<option name="s">
<para>Only dial a channel if its device state says that it is <literal>NOT_INUSE</literal></para>
</option>
<option name="A">
<argument name="x" required="true">
<para>The announcement to playback to all devices</para>
</argument>
<para>Play an announcement to all paged participants</para>
</option>
<option name="n">
<para>Do not play announcement to caller (alters <literal>A(x)</literal> behavior)</para>
</option>
</optionlist>
</parameter>
<parameter name="timeout">
<para>Specify the length of time that the system will attempt to connect a call.
After this duration, any page calls that have not been answered will be hung up by the
system.</para>
</parameter>
</syntax>
<description>
<para>Places outbound calls to the given <replaceable>technology</replaceable>/<replaceable>resource</replaceable>
and dumps them into a conference bridge as muted participants. The original
caller is dumped into the conference as a speaker and the room is
destroyed when the original caller leaves.</para>
</description>
<see-also>
<ref type="application">ConfBridge</ref>
</see-also>
</application>
***/
static const char * const app_page= "Page";
enum page_opt_flags {
PAGE_DUPLEX = (1 << 0),
PAGE_QUIET = (1 << 1),
PAGE_RECORD = (1 << 2),
PAGE_SKIP = (1 << 3),
PAGE_IGNORE_FORWARDS = (1 << 4),
PAGE_ANNOUNCE = (1 << 5),
PAGE_NOCALLERANNOUNCE = (1 << 6),
PAGE_PREDIAL_CALLEE = (1 << 7),
PAGE_PREDIAL_CALLER = (1 << 8),
};
enum {
OPT_ARG_ANNOUNCE = 0,
OPT_ARG_PREDIAL_CALLEE = 1,
OPT_ARG_PREDIAL_CALLER = 2,
OPT_ARG_ARRAY_SIZE = 3,
};
AST_APP_OPTIONS(page_opts, {
AST_APP_OPTION_ARG('b', PAGE_PREDIAL_CALLEE, OPT_ARG_PREDIAL_CALLEE),
AST_APP_OPTION_ARG('B', PAGE_PREDIAL_CALLER, OPT_ARG_PREDIAL_CALLER),
AST_APP_OPTION('d', PAGE_DUPLEX),
AST_APP_OPTION('q', PAGE_QUIET),
AST_APP_OPTION('r', PAGE_RECORD),
AST_APP_OPTION('s', PAGE_SKIP),
AST_APP_OPTION('i', PAGE_IGNORE_FORWARDS),
AST_APP_OPTION_ARG('A', PAGE_ANNOUNCE, OPT_ARG_ANNOUNCE),
AST_APP_OPTION('n', PAGE_NOCALLERANNOUNCE),
});
#define PAGE_BEEP "beep"
/* We use this structure as a way to pass this to all dialed channels */
struct page_options {
char *opts[OPT_ARG_ARRAY_SIZE];
struct ast_flags flags;
};
/*!
* \internal
* \brief Setup the page bridge profile.
*
* \param chan Setup bridge profile on this channel.
* \param options Options to setup bridge profile.
*/
static void setup_profile_bridge(struct ast_channel *chan, struct page_options *options)
{
/* Use default_bridge as a starting point */
ast_func_write(chan, "CONFBRIDGE(bridge,template)", "");
if (ast_test_flag(&options->flags, PAGE_RECORD)) {
ast_func_write(chan, "CONFBRIDGE(bridge,record_conference)", "yes");
}
}
/*!
* \internal
* \brief Setup the paged user profile.
*
* \param chan Setup user profile on this channel.
* \param options Options to setup paged user profile.
*/
static void setup_profile_paged(struct ast_channel *chan, struct page_options *options)
{
/* Use default_user as a starting point */
ast_func_write(chan, "CONFBRIDGE(user,template)", "");
ast_func_write(chan, "CONFBRIDGE(user,quiet)", "yes");
ast_func_write(chan, "CONFBRIDGE(user,end_marked)", "yes");
if (!ast_test_flag(&options->flags, PAGE_DUPLEX)) {
ast_func_write(chan, "CONFBRIDGE(user,startmuted)", "yes");
}
if (ast_test_flag(&options->flags, PAGE_ANNOUNCE)
&& !ast_strlen_zero(options->opts[OPT_ARG_ANNOUNCE])) {
ast_func_write(chan, "CONFBRIDGE(user,announcement)", options->opts[OPT_ARG_ANNOUNCE]);
}
}
/*!
* \internal
* \brief Setup the caller user profile.
*
* \param chan Setup user profile on this channel.
* \param options Options to setup caller user profile.
*/
static void setup_profile_caller(struct ast_channel *chan, struct page_options *options)
{
/* Use default_user as a starting point if not already setup. */
ast_func_write(chan, "CONFBRIDGE(user,template)", "");
ast_func_write(chan, "CONFBRIDGE(user,quiet)", "yes");
ast_func_write(chan, "CONFBRIDGE(user,marked)", "yes");
if (!ast_test_flag(&options->flags, PAGE_NOCALLERANNOUNCE)
&& ast_test_flag(&options->flags, PAGE_ANNOUNCE)
&& !ast_strlen_zero(options->opts[OPT_ARG_ANNOUNCE])) {
ast_func_write(chan, "CONFBRIDGE(user,announcement)", options->opts[OPT_ARG_ANNOUNCE]);
}
}
static void page_state_callback(struct ast_dial *dial)
{
struct ast_channel *chan;
struct page_options *options;
if (ast_dial_state(dial) != AST_DIAL_RESULT_ANSWERED ||
!(chan = ast_dial_answered(dial)) ||
!(options = ast_dial_get_user_data(dial))) {
return;
}
setup_profile_bridge(chan, options);
setup_profile_paged(chan, options);
}
static int page_exec(struct ast_channel *chan, const char *data)
{
char *tech;
char *resource;
char *tmp;
char *predial_callee = NULL;
char confbridgeopts[128];
char originator[AST_CHANNEL_NAME];
struct page_options options = { { 0, }, { 0, } };
unsigned int confid = ast_random();
struct ast_app *app;
int res = 0;
int pos = 0;
int i = 0;
struct ast_dial **dial_list;
unsigned int num_dials;
int timeout = 0;
char *parse;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(devices);
AST_APP_ARG(options);
AST_APP_ARG(timeout);
);
if (!(app = pbx_findapp("ConfBridge"))) {
ast_log(LOG_WARNING, "There is no ConfBridge application available!\n");
return -1;
};
parse = ast_strdupa(data ?: "");
AST_STANDARD_APP_ARGS(args, parse);
ast_copy_string(originator, ast_channel_name(chan), sizeof(originator));
if ((tmp = strchr(originator, '-'))) {
*tmp = '\0';
}
if (!ast_strlen_zero(args.options)) {
ast_app_parse_options(page_opts, &options.flags, options.opts, args.options);
}
if (!ast_strlen_zero(args.timeout)) {
timeout = atoi(args.timeout);
}
snprintf(confbridgeopts, sizeof(confbridgeopts), "ConfBridge,%u", confid);
/* Count number of extensions in list by number of ampersands + 1 */
num_dials = 1;
tmp = args.devices ?: "";
while (*tmp) {
if (*tmp == '&') {
num_dials++;
}
tmp++;
}
if (!(dial_list = ast_calloc(num_dials, sizeof(struct ast_dial *)))) {
ast_log(LOG_ERROR, "Can't allocate %ld bytes for dial list\n", (long)(sizeof(struct ast_dial *) * num_dials));
return -1;
}
/* PREDIAL: Preprocess any callee gosub arguments. */
if (ast_test_flag(&options.flags, PAGE_PREDIAL_CALLEE)
&& !ast_strlen_zero(options.opts[OPT_ARG_PREDIAL_CALLEE])) {
ast_replace_subargument_delimiter(options.opts[OPT_ARG_PREDIAL_CALLEE]);
predial_callee =
(char *) ast_app_expand_sub_args(chan, options.opts[OPT_ARG_PREDIAL_CALLEE]);
}
/* PREDIAL: Run gosub on the caller's channel */
if (ast_test_flag(&options.flags, PAGE_PREDIAL_CALLER)
&& !ast_strlen_zero(options.opts[OPT_ARG_PREDIAL_CALLER])) {
ast_replace_subargument_delimiter(options.opts[OPT_ARG_PREDIAL_CALLER]);
ast_app_exec_sub(NULL, chan, options.opts[OPT_ARG_PREDIAL_CALLER], 0);
}
/* Go through parsing/calling each device */
while ((tech = strsep(&args.devices, "&"))) {
int state = 0;
struct ast_dial *dial = NULL;
tech = ast_strip(tech);
if (ast_strlen_zero(tech)) {
/* No tech/resource in this position. */
continue;
}
/* don't call the originating device */
if (!strcasecmp(tech, originator)) {
continue;
}
/* If no resource is available, continue on */
if (!(resource = strchr(tech, '/'))) {
ast_log(LOG_WARNING, "Incomplete destination: '%s' supplied.\n", tech);
continue;
}
/* Ensure device is not in use if skip option is enabled */
if (ast_test_flag(&options.flags, PAGE_SKIP)) {
state = ast_device_state(tech);
if (state == AST_DEVICE_UNKNOWN) {
ast_verb(3, "Destination '%s' has device state '%s'. Paging anyway.\n",
tech, ast_devstate2str(state));
} else if (state != AST_DEVICE_NOT_INUSE) {
ast_verb(3, "Destination '%s' has device state '%s'.\n",
tech, ast_devstate2str(state));
continue;
}
}
*resource++ = '\0';
/* Create a dialing structure */
if (!(dial = ast_dial_create())) {
ast_log(LOG_WARNING, "Failed to create dialing structure.\n");
continue;
}
/* Append technology and resource */
if (ast_dial_append(dial, tech, resource, NULL) == -1) {
ast_log(LOG_ERROR, "Failed to add %s/%s to outbound dial\n", tech, resource);
ast_dial_destroy(dial);
continue;
}
/* Set ANSWER_EXEC as global option */
ast_dial_option_global_enable(dial, AST_DIAL_OPTION_ANSWER_EXEC, confbridgeopts);
if (predial_callee) {
ast_dial_option_global_enable(dial, AST_DIAL_OPTION_PREDIAL, predial_callee);
}
if (timeout) {
ast_dial_set_global_timeout(dial, timeout * 1000);
}
if (ast_test_flag(&options.flags, PAGE_IGNORE_FORWARDS)) {
ast_dial_option_global_enable(dial, AST_DIAL_OPTION_DISABLE_CALL_FORWARDING, NULL);
}
ast_dial_set_state_callback(dial, &page_state_callback);
ast_dial_set_user_data(dial, &options);
/* Run this dial in async mode */
ast_dial_run(dial, chan, 1);
/* Put in our dialing array */
dial_list[pos++] = dial;
}
ast_free(predial_callee);
if (!ast_test_flag(&options.flags, PAGE_QUIET)) {
if (!ast_fileexists(PAGE_BEEP, NULL, NULL)) {
ast_log(LOG_WARNING, "Missing required sound file: '" PAGE_BEEP "'\n");
} else {
res = ast_streamfile(chan, PAGE_BEEP, ast_channel_language(chan));
if (!res) {
res = ast_waitstream(chan, "");
}
}
}
if (!res) {
setup_profile_bridge(chan, &options);
setup_profile_caller(chan, &options);
snprintf(confbridgeopts, sizeof(confbridgeopts), "%u", confid);
pbx_exec(chan, app, confbridgeopts);
}
/* Go through each dial attempt cancelling, joining, and destroying */
for (i = 0; i < pos; i++) {
struct ast_dial *dial = dial_list[i];
/* We have to wait for the async thread to exit as it's possible ConfBridge won't throw them out immediately */
ast_dial_join(dial);
/* Hangup all channels */
ast_dial_hangup(dial);
/* Destroy dialing structure */
ast_dial_destroy(dial);
}
ast_free(dial_list);
return -1;
}
static int unload_module(void)
{
return ast_unregister_application(app_page);
}
static int load_module(void)
{
return ast_register_application_xml(app_page, page_exec);
}
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "Page Multiple Phones",
.support_level = AST_MODULE_SUPPORT_CORE,
.load = load_module,
.unload = unload_module,
.requires = "app_confbridge",
);