Mon May 14 04:42:51 2007

Asterisk developer's documentation


app_page.c

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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>zaptel</depend>
00030  ***/
00031 
00032 #include "asterisk.h"
00033 
00034 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
00035 
00036 #include <stdio.h>
00037 #include <stdlib.h>
00038 #include <unistd.h>
00039 #include <string.h>
00040 #include <errno.h>
00041 
00042 #include "asterisk/options.h"
00043 #include "asterisk/logger.h"
00044 #include "asterisk/channel.h"
00045 #include "asterisk/pbx.h"
00046 #include "asterisk/module.h"
00047 #include "asterisk/file.h"
00048 #include "asterisk/app.h"
00049 #include "asterisk/chanvars.h"
00050 #include "asterisk/utils.h"
00051 #include "asterisk/dial.h"
00052 #include "asterisk/devicestate.h"
00053 
00054 static const char *app_page= "Page";
00055 
00056 static const char *page_synopsis = "Pages phones";
00057 
00058 static const char *page_descrip =
00059 "Page(Technology/Resource&Technology2/Resource2[|options])\n"
00060 "  Places outbound calls to the given technology / resource and dumps\n"
00061 "them into a conference bridge as muted participants.  The original\n"
00062 "caller is dumped into the conference as a speaker and the room is\n"
00063 "destroyed when the original caller leaves.  Valid options are:\n"
00064 "        d - full duplex audio\n"
00065 "        q - quiet, do not play beep to caller\n"
00066 "        r - record the page into a file (see 'r' for app_meetme)\n";
00067 
00068 enum {
00069    PAGE_DUPLEX = (1 << 0),
00070    PAGE_QUIET = (1 << 1),
00071    PAGE_RECORD = (1 << 2),
00072 } page_opt_flags;
00073 
00074 AST_APP_OPTIONS(page_opts, {
00075    AST_APP_OPTION('d', PAGE_DUPLEX),
00076    AST_APP_OPTION('q', PAGE_QUIET),
00077    AST_APP_OPTION('r', PAGE_RECORD),
00078 });
00079 
00080 #define MAX_DIALS 128
00081 
00082 static int page_exec(struct ast_channel *chan, void *data)
00083 {
00084    struct ast_module_user *u;
00085    char *options, *tech, *resource, *tmp;
00086    char meetmeopts[88], originator[AST_CHANNEL_NAME];
00087    struct ast_flags flags = { 0 };
00088    unsigned int confid = ast_random();
00089    struct ast_app *app;
00090    int res = 0, pos = 0, i = 0;
00091    struct ast_dial *dials[MAX_DIALS];
00092 
00093    if (ast_strlen_zero(data)) {
00094       ast_log(LOG_WARNING, "This application requires at least one argument (destination(s) to page)\n");
00095       return -1;
00096    }
00097 
00098    u = ast_module_user_add(chan);
00099 
00100    if (!(app = pbx_findapp("MeetMe"))) {
00101       ast_log(LOG_WARNING, "There is no MeetMe application available!\n");
00102       ast_module_user_remove(u);
00103       return -1;
00104    };
00105 
00106    options = ast_strdupa(data);
00107 
00108    ast_copy_string(originator, chan->name, sizeof(originator));
00109    if ((tmp = strchr(originator, '-')))
00110       *tmp = '\0';
00111 
00112    tmp = strsep(&options, "|");
00113    if (options)
00114       ast_app_parse_options(page_opts, &flags, NULL, options);
00115 
00116    snprintf(meetmeopts, sizeof(meetmeopts), "MeetMe|%ud|%s%sqxdw(5)", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "m"),
00117       (ast_test_flag(&flags, PAGE_RECORD) ? "r" : "") );
00118 
00119    /* Go through parsing/calling each device */
00120    while ((tech = strsep(&tmp, "&"))) {
00121       struct ast_dial *dial = NULL;
00122 
00123       /* don't call the originating device */
00124       if (!strcasecmp(tech, originator))
00125          continue;
00126 
00127       /* If no resource is available, continue on */
00128       if (!(resource = strchr(tech, '/'))) {
00129          ast_log(LOG_WARNING, "Incomplete destination '%s' supplied.\n", tech);
00130          continue;
00131       }
00132 
00133       *resource++ = '\0';
00134 
00135       /* Create a dialing structure */
00136       if (!(dial = ast_dial_create())) {
00137          ast_log(LOG_WARNING, "Failed to create dialing structure.\n");
00138          continue;
00139       }
00140 
00141       /* Append technology and resource */
00142       ast_dial_append(dial, tech, resource);
00143 
00144       /* Set ANSWER_EXEC as global option */
00145       ast_dial_option_global_enable(dial, AST_DIAL_OPTION_ANSWER_EXEC, meetmeopts);
00146 
00147       /* Run this dial in async mode */
00148       ast_dial_run(dial, chan, 1);
00149 
00150       /* Put in our dialing array */
00151       dials[pos++] = dial;
00152    }
00153 
00154    if (!ast_test_flag(&flags, PAGE_QUIET)) {
00155       res = ast_streamfile(chan, "beep", chan->language);
00156       if (!res)
00157          res = ast_waitstream(chan, "");
00158    }
00159 
00160    if (!res) {
00161       snprintf(meetmeopts, sizeof(meetmeopts), "%ud|A%s%sqxd", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "t"), 
00162          (ast_test_flag(&flags, PAGE_RECORD) ? "r" : "") );
00163       pbx_exec(chan, app, meetmeopts);
00164    }
00165 
00166    /* Go through each dial attempt cancelling, joining, and destroying */
00167    for (i = 0; i < pos; i++) {
00168       struct ast_dial *dial = dials[i];
00169 
00170       /* We have to wait for the async thread to exit as it's possible Meetme won't throw them out immediately */
00171       ast_dial_join(dial);
00172 
00173       /* Hangup all channels */
00174       ast_dial_hangup(dial);
00175 
00176       /* Destroy dialing structure */
00177       ast_dial_destroy(dial);
00178    }
00179 
00180    ast_module_user_remove(u);
00181 
00182    return -1;
00183 }
00184 
00185 static int unload_module(void)
00186 {
00187    int res;
00188 
00189    res =  ast_unregister_application(app_page);
00190 
00191    ast_module_user_hangup_all();
00192 
00193    return res;
00194 }
00195 
00196 static int load_module(void)
00197 {
00198    return ast_register_application(app_page, page_exec, page_synopsis, page_descrip);
00199 }
00200 
00201 AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Page Multiple Phones");
00202 

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