Fri Jun 19 12:09:26 2009

Asterisk developer's documentation


app_page.c

Go to the documentation of this file.
00001 /*
00002  * Asterisk -- An open source telephony toolkit.
00003  *
00004  * Copyright (c) 2004 - 2006 Digium, Inc.  All rights reserved.
00005  *
00006  * Mark Spencer <markster@digium.com>
00007  *
00008  * This code is released under the GNU General Public License
00009  * version 2.0.  See LICENSE for more information.
00010  *
00011  * See http://www.asterisk.org for more information about
00012  * the Asterisk project. Please do not directly contact
00013  * any of the maintainers of this project for assistance;
00014  * the project provides a web site, mailing lists and IRC
00015  * channels for your use.
00016  *
00017  */
00018 
00019 /*! \file
00020  *
00021  * \brief page() - Paging application
00022  *
00023  * \author Mark Spencer <markster@digium.com>
00024  *
00025  * \ingroup applications
00026  */
00027 
00028 /*** MODULEINFO
00029    <depend>dahdi</depend>
00030    <depend>app_meetme</depend>
00031  ***/
00032 
00033 #include "asterisk.h"
00034 
00035 ASTERISK_FILE_VERSION(__FILE__, "$Revision: 170982 $")
00036 
00037 #include "asterisk/channel.h"
00038 #include "asterisk/pbx.h"
00039 #include "asterisk/module.h"
00040 #include "asterisk/file.h"
00041 #include "asterisk/app.h"
00042 #include "asterisk/chanvars.h"
00043 #include "asterisk/utils.h"
00044 #include "asterisk/devicestate.h"
00045 #include "asterisk/dial.h"
00046 
00047 static const char *app_page= "Page";
00048 
00049 static const char *page_synopsis = "Pages phones";
00050 
00051 static const char *page_descrip =
00052 "Page(Technology/Resource&Technology2/Resource2[,options])\n"
00053 "  Places outbound calls to the given technology / resource and dumps\n"
00054 "them into a conference bridge as muted participants.  The original\n"
00055 "caller is dumped into the conference as a speaker and the room is\n"
00056 "destroyed when the original caller leaves.  Valid options are:\n"
00057 "        d - full duplex audio\n"
00058 "        q - quiet, do not play beep to caller\n"
00059 "        r - record the page into a file (see 'r' for app_meetme)\n"
00060 "        s - only dial channel if devicestate says it is not in use\n";
00061 
00062 enum {
00063    PAGE_DUPLEX = (1 << 0),
00064    PAGE_QUIET = (1 << 1),
00065    PAGE_RECORD = (1 << 2),
00066    PAGE_SKIP = (1 << 3),
00067 } page_opt_flags;
00068 
00069 AST_APP_OPTIONS(page_opts, {
00070    AST_APP_OPTION('d', PAGE_DUPLEX),
00071    AST_APP_OPTION('q', PAGE_QUIET),
00072    AST_APP_OPTION('r', PAGE_RECORD),
00073    AST_APP_OPTION('s', PAGE_SKIP),
00074 });
00075 
00076 
00077 static int page_exec(struct ast_channel *chan, void *data)
00078 {
00079    char *options, *tech, *resource, *tmp, *tmp2;
00080    char meetmeopts[88], originator[AST_CHANNEL_NAME], *opts[0];
00081    struct ast_flags flags = { 0 };
00082    unsigned int confid = ast_random();
00083    struct ast_app *app;
00084    int res = 0, pos = 0, i = 0;
00085    struct ast_dial **dial_list;
00086    unsigned int num_dials;
00087 
00088    if (ast_strlen_zero(data)) {
00089       ast_log(LOG_WARNING, "This application requires at least one argument (destination(s) to page)\n");
00090       return -1;
00091    }
00092 
00093    if (!(app = pbx_findapp("MeetMe"))) {
00094       ast_log(LOG_WARNING, "There is no MeetMe application available!\n");
00095       return -1;
00096    };
00097 
00098    options = ast_strdupa(data);
00099 
00100    ast_copy_string(originator, chan->name, sizeof(originator));
00101    if ((tmp = strchr(originator, '-')))
00102       *tmp = '\0';
00103 
00104    tmp = strsep(&options, ",");
00105    if (options)
00106       ast_app_parse_options(page_opts, &flags, opts, options);
00107 
00108    snprintf(meetmeopts, sizeof(meetmeopts), "MeetMe,%ud,%s%sqxdw(5)", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "m"),
00109       (ast_test_flag(&flags, PAGE_RECORD) ? "r" : "") );
00110 
00111    /* Count number of extensions in list by number of ampersands + 1 */
00112    num_dials = 1;
00113    tmp2 = tmp;
00114    while (*tmp2) {
00115       if (*tmp2 == '&') {
00116          num_dials++;
00117       }
00118       tmp2++;
00119    }
00120 
00121    if (!(dial_list = ast_calloc(num_dials, sizeof(struct ast_dial *)))) {
00122       ast_log(LOG_ERROR, "Can't allocate %ld bytes for dial list\n", (long)(sizeof(struct ast_dial *) * num_dials));
00123       return -1;
00124    }
00125 
00126    /* Go through parsing/calling each device */
00127    while ((tech = strsep(&tmp, "&"))) {
00128       int state = 0;
00129       struct ast_dial *dial = NULL;
00130 
00131       /* don't call the originating device */
00132       if (!strcasecmp(tech, originator))
00133          continue;
00134 
00135       /* If no resource is available, continue on */
00136       if (!(resource = strchr(tech, '/'))) {
00137          ast_log(LOG_WARNING, "Incomplete destination '%s' supplied.\n", tech);
00138          continue;
00139       }
00140 
00141       /* Ensure device is not in use if skip option is enabled */
00142       if (ast_test_flag(&flags, PAGE_SKIP)) {
00143          state = ast_device_state(tech);
00144          if (state == AST_DEVICE_UNKNOWN) {
00145             ast_log(LOG_WARNING, "Destination '%s' has device state '%s'. Paging anyway.\n", tech, devstate2str(state));
00146          } else if (state != AST_DEVICE_NOT_INUSE) {
00147             ast_log(LOG_WARNING, "Destination '%s' has device state '%s'.\n", tech, devstate2str(state));
00148             continue;
00149          }
00150       }
00151 
00152       *resource++ = '\0';
00153 
00154       /* Create a dialing structure */
00155       if (!(dial = ast_dial_create())) {
00156          ast_log(LOG_WARNING, "Failed to create dialing structure.\n");
00157          continue;
00158       }
00159 
00160       /* Append technology and resource */
00161       ast_dial_append(dial, tech, resource);
00162 
00163       /* Set ANSWER_EXEC as global option */
00164       ast_dial_option_global_enable(dial, AST_DIAL_OPTION_ANSWER_EXEC, meetmeopts);
00165 
00166       /* Run this dial in async mode */
00167       ast_dial_run(dial, chan, 1);
00168 
00169       /* Put in our dialing array */
00170       dial_list[pos++] = dial;
00171    }
00172 
00173    if (!ast_test_flag(&flags, PAGE_QUIET)) {
00174       res = ast_streamfile(chan, "beep", chan->language);
00175       if (!res)
00176          res = ast_waitstream(chan, "");
00177    }
00178 
00179    if (!res) {
00180       snprintf(meetmeopts, sizeof(meetmeopts), "%ud,A%s%sqxd", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "t"), 
00181          (ast_test_flag(&flags, PAGE_RECORD) ? "r" : "") );
00182       pbx_exec(chan, app, meetmeopts);
00183    }
00184 
00185    /* Go through each dial attempt cancelling, joining, and destroying */
00186    for (i = 0; i < pos; i++) {
00187       struct ast_dial *dial = dial_list[i];
00188 
00189       /* We have to wait for the async thread to exit as it's possible Meetme won't throw them out immediately */
00190       ast_dial_join(dial);
00191 
00192       /* Hangup all channels */
00193       ast_dial_hangup(dial);
00194 
00195       /* Destroy dialing structure */
00196       ast_dial_destroy(dial);
00197    }
00198 
00199    return -1;
00200 }
00201 
00202 static int unload_module(void)
00203 {
00204    return ast_unregister_application(app_page);
00205 }
00206 
00207 static int load_module(void)
00208 {
00209    return ast_register_application(app_page, page_exec, page_synopsis, page_descrip);
00210 }
00211 
00212 AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Page Multiple Phones");
00213 

Generated on Fri Jun 19 12:09:26 2009 for Asterisk - the Open Source PBX by  doxygen 1.4.7