apps/app_page.c

   1 /*
   2  * Asterisk -- An open source telephony toolkit.
   3  *
   4  * Copyright (c) 2004 - 2006 Digium, Inc.  All rights reserved.
   5  *
   6  * Mark Spencer <markster@digium.com>
   7  *
   8  * This code is released under the GNU General Public License
   9  * version 2.0.  See LICENSE for more information.
  10  *
  11  * See http://www.asterisk.org for more information about
  12  * the Asterisk project. Please do not directly contact
  13  * any of the maintainers of this project for assistance;
  14  * the project provides a web site, mailing lists and IRC
  15  * channels for your use.
  16  *
  17  */
  18
  19 /*! \file
  20  *
  21  * \brief page() - Paging application
  22  *
  23  * \author Mark Spencer <markster@digium.com>
  24  *
  25  * \ingroup applications
  26  */
  27
  28 /*** MODULEINFO
  29         <depend>zaptel</depend>
  30         <depend>app_meetme</depend>
  31  ***/
  32
  33 #include "asterisk.h"
  34
  35 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
  36
  37 #include <stdio.h>
  38 #include <stdlib.h>
  39 #include <unistd.h>
  40 #include <string.h>
  41 #include <errno.h>
  42
  43 #include "asterisk/options.h"
  44 #include "asterisk/logger.h"
  45 #include "asterisk/channel.h"
  46 #include "asterisk/pbx.h"
  47 #include "asterisk/module.h"
  48 #include "asterisk/file.h"
  49 #include "asterisk/app.h"
  50 #include "asterisk/chanvars.h"
  51 #include "asterisk/utils.h"
  52 #include "asterisk/dial.h"
  53 #include "asterisk/devicestate.h"
  54
  55 static const char *app_page= "Page";
  56
  57 static const char *page_synopsis = "Pages phones";
  58
  59 static const char *page_descrip =
  60 "Page(Technology/Resource&Technology2/Resource2[|options])\n"
  61 "  Places outbound calls to the given technology / resource and dumps\n"
  62 "them into a conference bridge as muted participants.  The original\n"
  63 "caller is dumped into the conference as a speaker and the room is\n"
  64 "destroyed when the original caller leaves.  Valid options are:\n"
  65 "        d - full duplex audio\n"
  66 "        q - quiet, do not play beep to caller\n"
  67 "        r - record the page into a file (see 'r' for app_meetme)\n";
  68
  69 enum {
  70         PAGE_DUPLEX = (1 << 0),
  71         PAGE_QUIET = (1 << 1),
  72         PAGE_RECORD = (1 << 2),
  73 } page_opt_flags;
  74
  75 AST_APP_OPTIONS(page_opts, {
  76         AST_APP_OPTION('d', PAGE_DUPLEX),
  77         AST_APP_OPTION('q', PAGE_QUIET),
  78         AST_APP_OPTION('r', PAGE_RECORD),
  79 });
  80
  81 #define MAX_DIALS 128
  82
  83 static int page_exec(struct ast_channel *chan, void *data)
  84 {
  85         struct ast_module_user *u;
  86         char *options, *tech, *resource, *tmp;
  87         char meetmeopts[88], originator[AST_CHANNEL_NAME];
  88         struct ast_flags flags = { 0 };
  89         unsigned int confid = ast_random();
  90         struct ast_app *app;
  91         int res = 0, pos = 0, i = 0;
  92         struct ast_dial *dials[MAX_DIALS];
  93
  94         if (ast_strlen_zero(data)) {
  95                 ast_log(LOG_WARNING, "This application requires at least one argument (destination(s) to page)\n");
  96                 return -1;
  97         }
  98
  99         u = ast_module_user_add(chan);
 100
 101         if (!(app = pbx_findapp("MeetMe"))) {
 102                 ast_log(LOG_WARNING, "There is no MeetMe application available!\n");
 103                 ast_module_user_remove(u);
 104                 return -1;
 105         };
 106
 107         options = ast_strdupa(data);
 108
 109         ast_copy_string(originator, chan->name, sizeof(originator));
 110         if ((tmp = strchr(originator, '-')))
 111                 *tmp = '\0';
 112
 113         tmp = strsep(&options, "|");
 114         if (options)
 115                 ast_app_parse_options(page_opts, &flags, NULL, options);
 116
 117         snprintf(meetmeopts, sizeof(meetmeopts), "MeetMe|%ud|%s%sqxdw(5)", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "m"),
 118                 (ast_test_flag(&flags, PAGE_RECORD) ? "r" : "") );
 119
 120         /* Go through parsing/calling each device */
 121         while ((tech = strsep(&tmp, "&"))) {
 122                 struct ast_dial *dial = NULL;
 123
 124                 /* don't call the originating device */
 125                 if (!strcasecmp(tech, originator))
 126                         continue;
 127
 128                 /* If no resource is available, continue on */
 129                 if (!(resource = strchr(tech, '/'))) {
 130                         ast_log(LOG_WARNING, "Incomplete destination '%s' supplied.\n", tech);
 131                         continue;
 132                 }
 133
 134                 *resource++ = '\0';
 135
 136                 /* Create a dialing structure */
 137                 if (!(dial = ast_dial_create())) {
 138                         ast_log(LOG_WARNING, "Failed to create dialing structure.\n");
 139                         continue;
 140                 }
 141
 142                 /* Append technology and resource */
 143                 ast_dial_append(dial, tech, resource);
 144
 145                 /* Set ANSWER_EXEC as global option */
 146                 ast_dial_option_global_enable(dial, AST_DIAL_OPTION_ANSWER_EXEC, meetmeopts);
 147
 148                 /* Run this dial in async mode */
 149                 ast_dial_run(dial, chan, 1);
 150
 151                 /* Put in our dialing array */
 152                 dials[pos++] = dial;
 153         }
 154
 155         if (!ast_test_flag(&flags, PAGE_QUIET)) {
 156                 res = ast_streamfile(chan, "beep", chan->language);
 157                 if (!res)
 158                         res = ast_waitstream(chan, "");
 159         }
 160
 161         if (!res) {
 162                 snprintf(meetmeopts, sizeof(meetmeopts), "%ud|A%s%sqxd", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "t"),
 163                         (ast_test_flag(&flags, PAGE_RECORD) ? "r" : "") );
 164                 pbx_exec(chan, app, meetmeopts);
 165         }
 166
 167         /* Go through each dial attempt cancelling, joining, and destroying */
 168         for (i = 0; i < pos; i++) {
 169                 struct ast_dial *dial = dials[i];
 170
 171                 /* We have to wait for the async thread to exit as it's possible Meetme won't throw them out immediately */
 172                 ast_dial_join(dial);
 173
 174                 /* Hangup all channels */
 175                 ast_dial_hangup(dial);
 176
 177                 /* Destroy dialing structure */
 178                 ast_dial_destroy(dial);
 179         }
 180
 181         ast_module_user_remove(u);
 182
 183         return -1;
 184 }
 185
 186 static int unload_module(void)
 187 {
 188         int res;
 189
 190         res =  ast_unregister_application(app_page);
 191
 192         ast_module_user_hangup_all();
 193
 194         return res;
 195 }
 196
 197 static int load_module(void)
 198 {
 199         return ast_register_application(app_page, page_exec, page_synopsis, page_descrip);
 200 }
 201
 202 AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Page Multiple Phones");
 203