mirror of
				https://github.com/asterisk/asterisk.git
				synced 2025-10-27 06:31:54 +00:00 
			
		
		
		
	Also updates the 'since' of applications/functions that existed before
XML documentation was introduced (1.6.2.0).
(cherry picked from commit 67e89b3e77)
		
	
		
			
				
	
	
		
			457 lines
		
	
	
		
			14 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
			
		
		
	
	
			457 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>1.2.0</version>
 | |
| 		</since>
 | |
| 		<synopsis>
 | |
| 			Page series of phones
 | |
| 		</synopsis>
 | |
| 		<syntax>
 | |
| 			<parameter name="Technology/Resource" required="false" argsep="&">
 | |
| 				<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&
 | |
| 					Technology3/Resource3&.....</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",
 | |
| );
 |