https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r54066 | russell | 2007-02-12 11:58:43 -0600 (Mon, 12 Feb 2007) | 4 lines - Add the ability to register a callback to monitor state changes in an asynchronous dial operation. - Rename the various references to "status" to "state" in the dial API ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54067 65c4cc65-6c06-0410-ace0-fbb531ad65f3
214 lines
5.6 KiB
C
214 lines
5.6 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (c) 2004 - 2006 Digium, Inc. All rights reserved.
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*
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* Mark Spencer <markster@digium.com>
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*
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* This code is released under the GNU General Public License
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* version 2.0. See LICENSE for more information.
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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*/
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/*! \file
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*
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* \brief page() - Paging application
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*
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* \author Mark Spencer <markster@digium.com>
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*
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* \ingroup applications
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*/
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/*** MODULEINFO
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<depend>zaptel</depend>
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***/
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#include "asterisk.h"
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#include <stdio.h>
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#include <stdlib.h>
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#include <unistd.h>
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#include <string.h>
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#include <errno.h>
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#include "asterisk/options.h"
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#include "asterisk/logger.h"
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#include "asterisk/channel.h"
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#include "asterisk/pbx.h"
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#include "asterisk/module.h"
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#include "asterisk/file.h"
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#include "asterisk/app.h"
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#include "asterisk/chanvars.h"
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#include "asterisk/utils.h"
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#include "asterisk/devicestate.h"
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#include "asterisk/dial.h"
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static const char *app_page= "Page";
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static const char *page_synopsis = "Pages phones";
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static const char *page_descrip =
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"Page(Technology/Resource&Technology2/Resource2[|options])\n"
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" Places outbound calls to the given technology / resource and dumps\n"
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"them into a conference bridge as muted participants. The original\n"
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"caller is dumped into the conference as a speaker and the room is\n"
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"destroyed when the original caller leaves. Valid options are:\n"
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" d - full duplex audio\n"
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" q - quiet, do not play beep to caller\n"
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" r - record the page into a file (see 'r' for app_meetme)\n"
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" s - only dial channel if devicestate says it is not in use\n";
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enum {
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PAGE_DUPLEX = (1 << 0),
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PAGE_QUIET = (1 << 1),
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PAGE_RECORD = (1 << 2),
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PAGE_SKIP = (1 << 3),
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} page_opt_flags;
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AST_APP_OPTIONS(page_opts, {
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AST_APP_OPTION('d', PAGE_DUPLEX),
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AST_APP_OPTION('q', PAGE_QUIET),
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AST_APP_OPTION('r', PAGE_RECORD),
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AST_APP_OPTION('s', PAGE_SKIP),
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});
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#define MAX_DIALS 128
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static int page_exec(struct ast_channel *chan, void *data)
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{
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struct ast_module_user *u;
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char *options, *tech, *resource, *tmp;
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char meetmeopts[88], originator[AST_CHANNEL_NAME];
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struct ast_flags flags = { 0 };
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unsigned int confid = ast_random();
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struct ast_app *app;
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int res = 0, pos = 0, i = 0;
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struct ast_dial *dials[MAX_DIALS];
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if (ast_strlen_zero(data)) {
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ast_log(LOG_WARNING, "This application requires at least one argument (destination(s) to page)\n");
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return -1;
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}
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u = ast_module_user_add(chan);
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if (!(app = pbx_findapp("MeetMe"))) {
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ast_log(LOG_WARNING, "There is no MeetMe application available!\n");
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ast_module_user_remove(u);
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return -1;
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};
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options = ast_strdupa(data);
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ast_copy_string(originator, chan->name, sizeof(originator));
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if ((tmp = strchr(originator, '-')))
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*tmp = '\0';
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tmp = strsep(&options, "|");
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if (options)
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ast_app_parse_options(page_opts, &flags, NULL, options);
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snprintf(meetmeopts, sizeof(meetmeopts), "MeetMe|%ud|%s%sqxdw(5)", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "m"),
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(ast_test_flag(&flags, PAGE_RECORD) ? "r" : "") );
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/* Go through parsing/calling each device */
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while ((tech = strsep(&tmp, "&"))) {
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int state = 0;
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struct ast_dial *dial = NULL;
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/* don't call the originating device */
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if (!strcasecmp(tech, originator))
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continue;
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/* If no resource is available, continue on */
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if (!(resource = strchr(tech, '/'))) {
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ast_log(LOG_WARNING, "Incomplete destination '%s' supplied.\n", tech);
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continue;
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}
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/* Ensure device is not in use if skip option is enabled */
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if (ast_test_flag(&flags, PAGE_SKIP) && (state = ast_device_state(tech)) != AST_DEVICE_NOT_INUSE) {
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ast_log(LOG_WARNING, "Destination '%s' has device state '%s'.\n", tech, devstate2str(state));
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continue;
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}
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*resource++ = '\0';
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/* Create a dialing structure */
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if (!(dial = ast_dial_create())) {
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ast_log(LOG_WARNING, "Failed to create dialing structure.\n");
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continue;
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}
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/* Append technology and resource */
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ast_dial_append(dial, tech, resource);
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/* Set ANSWER_EXEC as global option */
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ast_dial_option_global_enable(dial, AST_DIAL_OPTION_ANSWER_EXEC, meetmeopts);
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/* Run this dial in async mode */
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ast_dial_run(dial, chan, 1);
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/* Put in our dialing array */
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dials[pos++] = dial;
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}
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if (!ast_test_flag(&flags, PAGE_QUIET)) {
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res = ast_streamfile(chan, "beep", chan->language);
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if (!res)
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res = ast_waitstream(chan, "");
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}
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if (!res) {
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snprintf(meetmeopts, sizeof(meetmeopts), "%ud|A%s%sqxd", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "t"),
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(ast_test_flag(&flags, PAGE_RECORD) ? "r" : "") );
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pbx_exec(chan, app, meetmeopts);
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}
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/* Go through each dial attempt cancelling, joining, and destroying */
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for (i = 0; i < pos; i++) {
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struct ast_dial *dial = dials[i];
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/* If the dial is already answered, then they will/should get kicked out by Meetme */
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if (ast_dial_state(dial) != AST_DIAL_RESULT_ANSWERED)
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ast_dial_join(dial);
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/* Hangup all channels */
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ast_dial_hangup(dial);
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/* Destroy dialing structure */
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ast_dial_destroy(dial);
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}
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ast_module_user_remove(u);
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return -1;
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}
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static int unload_module(void)
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{
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int res;
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res = ast_unregister_application(app_page);
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ast_module_user_hangup_all();
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return res;
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}
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static int load_module(void)
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{
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return ast_register_application(app_page, page_exec, page_synopsis, page_descrip);
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}
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AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Page Multiple Phones");
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