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>dahdi</depend>
  30         <depend>app_meetme</depend>
  31  ***/
  32
  33 #include "asterisk.h"
  34
  35 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
  36
  37 #include "asterisk/channel.h"
  38 #include "asterisk/pbx.h"
  39 #include "asterisk/module.h"
  40 #include "asterisk/file.h"
  41 #include "asterisk/app.h"
  42 #include "asterisk/chanvars.h"
  43 #include "asterisk/utils.h"
  44 #include "asterisk/devicestate.h"
  45 #include "asterisk/dial.h"
  46
  47 static const char *app_page= "Page";
  48
  49 static const char *page_synopsis = "Pages phones";
  50
  51 static const char *page_descrip =
  52 "Page(Technology/Resource&Technology2/Resource2[,options])\n"
  53 "  Places outbound calls to the given technology / resource and dumps\n"
  54 "them into a conference bridge as muted participants.  The original\n"
  55 "caller is dumped into the conference as a speaker and the room is\n"
  56 "destroyed when the original caller leaves.  Valid options are:\n"
  57 "        d - full duplex audio\n"
  58 "        q - quiet, do not play beep to caller\n"
  59 "        r - record the page into a file (see 'r' for app_meetme)\n"
  60 "        s - only dial channel if devicestate says it is not in use\n";
  61
  62 enum {
  63         PAGE_DUPLEX = (1 << 0),
  64         PAGE_QUIET = (1 << 1),
  65         PAGE_RECORD = (1 << 2),
  66         PAGE_SKIP = (1 << 3),
  67 } page_opt_flags;
  68
  69 AST_APP_OPTIONS(page_opts, {
  70         AST_APP_OPTION('d', PAGE_DUPLEX),
  71         AST_APP_OPTION('q', PAGE_QUIET),
  72         AST_APP_OPTION('r', PAGE_RECORD),
  73         AST_APP_OPTION('s', PAGE_SKIP),
  74 });
  75
  76 #define MAX_DIALS 128
  77
  78 static int page_exec(struct ast_channel *chan, void *data)
  79 {
  80         char *options, *tech, *resource, *tmp;
  81         char meetmeopts[88], originator[AST_CHANNEL_NAME], *opts[0];
  82         struct ast_flags flags = { 0 };
  83         unsigned int confid = ast_random();
  84         struct ast_app *app;
  85         int res = 0, pos = 0, i = 0;
  86         struct ast_dial *dials[MAX_DIALS];
  87
  88         if (ast_strlen_zero(data)) {
  89                 ast_log(LOG_WARNING, "This application requires at least one argument (destination(s) to page)\n");
  90                 return -1;
  91         }
  92
  93         if (!(app = pbx_findapp("MeetMe"))) {
  94                 ast_log(LOG_WARNING, "There is no MeetMe application available!\n");
  95                 return -1;
  96         };
  97
  98         options = ast_strdupa(data);
  99
 100         ast_copy_string(originator, chan->name, sizeof(originator));
 101         if ((tmp = strchr(originator, '-')))
 102                 *tmp = '\0';
 103
 104         tmp = strsep(&options, ",");
 105         if (options)
 106                 ast_app_parse_options(page_opts, &flags, opts, options);
 107
 108         snprintf(meetmeopts, sizeof(meetmeopts), "MeetMe,%ud,%s%sqxdw(5)", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "m"),
 109                 (ast_test_flag(&flags, PAGE_RECORD) ? "r" : "") );
 110
 111         /* Go through parsing/calling each device */
 112         while ((tech = strsep(&tmp, "&"))) {
 113                 int state = 0;
 114                 struct ast_dial *dial = NULL;
 115
 116                 /* don't call the originating device */
 117                 if (!strcasecmp(tech, originator))
 118                         continue;
 119
 120                 /* If no resource is available, continue on */
 121                 if (!(resource = strchr(tech, '/'))) {
 122                         ast_log(LOG_WARNING, "Incomplete destination '%s' supplied.\n", tech);
 123                         continue;
 124                 }
 125
 126                 /* Ensure device is not in use if skip option is enabled */
 127                 if (ast_test_flag(&flags, PAGE_SKIP)) {
 128                         state = ast_device_state(tech);
 129                         if (state == AST_DEVICE_UNKNOWN) {
 130                                 ast_log(LOG_WARNING, "Destination '%s' has device state '%s'. Paging anyway.\n", tech, devstate2str(state));
 131                         } else if (state != AST_DEVICE_NOT_INUSE) {
 132                                 ast_log(LOG_WARNING, "Destination '%s' has device state '%s'.\n", tech, devstate2str(state));
 133                                 continue;
 134                         }
 135                 }
 136
 137                 *resource++ = '\0';
 138
 139                 /* Create a dialing structure */
 140                 if (!(dial = ast_dial_create())) {
 141                         ast_log(LOG_WARNING, "Failed to create dialing structure.\n");
 142                         continue;
 143                 }
 144
 145                 /* Append technology and resource */
 146                 ast_dial_append(dial, tech, resource);
 147
 148                 /* Set ANSWER_EXEC as global option */
 149                 ast_dial_option_global_enable(dial, AST_DIAL_OPTION_ANSWER_EXEC, meetmeopts);
 150
 151                 /* Run this dial in async mode */
 152                 ast_dial_run(dial, chan, 1);
 153
 154                 /* Put in our dialing array */
 155                 dials[pos++] = dial;
 156         }
 157
 158         if (!ast_test_flag(&flags, PAGE_QUIET)) {
 159                 res = ast_streamfile(chan, "beep", chan->language);
 160                 if (!res)
 161                         res = ast_waitstream(chan, "");
 162         }
 163
 164         if (!res) {
 165                 snprintf(meetmeopts, sizeof(meetmeopts), "%ud,A%s%sqxd", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "t"),
 166                         (ast_test_flag(&flags, PAGE_RECORD) ? "r" : "") );
 167                 pbx_exec(chan, app, meetmeopts);
 168         }
 169
 170         /* Go through each dial attempt cancelling, joining, and destroying */
 171         for (i = 0; i < pos; i++) {
 172                 struct ast_dial *dial = dials[i];
 173
 174                 /* We have to wait for the async thread to exit as it's possible Meetme won't throw them out immediately */
 175                 ast_dial_join(dial);
 176
 177                 /* Hangup all channels */
 178                 ast_dial_hangup(dial);
 179
 180                 /* Destroy dialing structure */
 181                 ast_dial_destroy(dial);
 182         }
 183
 184         return -1;
 185 }
 186
 187 static int unload_module(void)
 188 {
 189         return ast_unregister_application(app_page);
 190 }
 191
 192 static int load_module(void)
 193 {
 194         return ast_register_application(app_page, page_exec, page_synopsis, page_descrip);
 195 }
 196
 197 AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Page Multiple Phones");
 198