Mon Jul 14 17:24:48 2008

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    <depend>app_meetme</depend>
00031  ***/
00032 
00033 #include "asterisk.h"
00034 
00035 ASTERISK_FILE_VERSION(__FILE__, "$Revision: 66879 $")
00036 
00037 #include <stdio.h>
00038 #include <stdlib.h>
00039 #include <unistd.h>
00040 #include <string.h>
00041 #include <errno.h>
00042 
00043 #include "asterisk/options.h"
00044 #include "asterisk/logger.h"
00045 #include "asterisk/channel.h"
00046 #include "asterisk/pbx.h"
00047 #include "asterisk/module.h"
00048 #include "asterisk/file.h"
00049 #include "asterisk/app.h"
00050 #include "asterisk/chanvars.h"
00051 #include "asterisk/utils.h"
00052 #include "asterisk/dial.h"
00053 #include "asterisk/devicestate.h"
00054 
00055 static const char *app_page= "Page";
00056 
00057 static const char *page_synopsis = "Pages phones";
00058 
00059 static const char *page_descrip =
00060 "Page(Technology/Resource&Technology2/Resource2[|options])\n"
00061 "  Places outbound calls to the given technology / resource and dumps\n"
00062 "them into a conference bridge as muted participants.  The original\n"
00063 "caller is dumped into the conference as a speaker and the room is\n"
00064 "destroyed when the original caller leaves.  Valid options are:\n"
00065 "        d - full duplex audio\n"
00066 "        q - quiet, do not play beep to caller\n"
00067 "        r - record the page into a file (see 'r' for app_meetme)\n";
00068 
00069 enum {
00070    PAGE_DUPLEX = (1 << 0),
00071    PAGE_QUIET = (1 << 1),
00072    PAGE_RECORD = (1 << 2),
00073 } page_opt_flags;
00074 
00075 AST_APP_OPTIONS(page_opts, {
00076    AST_APP_OPTION('d', PAGE_DUPLEX),
00077    AST_APP_OPTION('q', PAGE_QUIET),
00078    AST_APP_OPTION('r', PAGE_RECORD),
00079 });
00080 
00081 #define MAX_DIALS 128
00082 
00083 static int page_exec(struct ast_channel *chan, void *data)
00084 {
00085    struct ast_module_user *u;
00086    char *options, *tech, *resource, *tmp;
00087    char meetmeopts[88], originator[AST_CHANNEL_NAME];
00088    struct ast_flags flags = { 0 };
00089    unsigned int confid = ast_random();
00090    struct ast_app *app;
00091    int res = 0, pos = 0, i = 0;
00092    struct ast_dial *dials[MAX_DIALS];
00093 
00094    if (ast_strlen_zero(data)) {
00095       ast_log(LOG_WARNING, "This application requires at least one argument (destination(s) to page)\n");
00096       return -1;
00097    }
00098 
00099    u = ast_module_user_add(chan);
00100 
00101    if (!(app = pbx_findapp("MeetMe"))) {
00102       ast_log(LOG_WARNING, "There is no MeetMe application available!\n");
00103       ast_module_user_remove(u);
00104       return -1;
00105    };
00106 
00107    options = ast_strdupa(data);
00108 
00109    ast_copy_string(originator, chan->name, sizeof(originator));
00110    if ((tmp = strchr(originator, '-')))
00111       *tmp = '\0';
00112 
00113    tmp = strsep(&options, "|");
00114    if (options)
00115       ast_app_parse_options(page_opts, &flags, NULL, options);
00116 
00117    snprintf(meetmeopts, sizeof(meetmeopts), "MeetMe|%ud|%s%sqxdw(5)", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "m"),
00118       (ast_test_flag(&flags, PAGE_RECORD) ? "r" : "") );
00119 
00120    /* Go through parsing/calling each device */
00121    while ((tech = strsep(&tmp, "&"))) {
00122       struct ast_dial *dial = NULL;
00123 
00124       /* don't call the originating device */
00125       if (!strcasecmp(tech, originator))
00126          continue;
00127 
00128       /* If no resource is available, continue on */
00129       if (!(resource = strchr(tech, '/'))) {
00130          ast_log(LOG_WARNING, "Incomplete destination '%s' supplied.\n", tech);
00131          continue;
00132       }
00133 
00134       *resource++ = '\0';
00135 
00136       /* Create a dialing structure */
00137       if (!(dial = ast_dial_create())) {
00138          ast_log(LOG_WARNING, "Failed to create dialing structure.\n");
00139          continue;
00140       }
00141 
00142       /* Append technology and resource */
00143       ast_dial_append(dial, tech, resource);
00144 
00145       /* Set ANSWER_EXEC as global option */
00146       ast_dial_option_global_enable(dial, AST_DIAL_OPTION_ANSWER_EXEC, meetmeopts);
00147 
00148       /* Run this dial in async mode */
00149       ast_dial_run(dial, chan, 1);
00150 
00151       /* Put in our dialing array */
00152       dials[pos++] = dial;
00153    }
00154 
00155    if (!ast_test_flag(&flags, PAGE_QUIET)) {
00156       res = ast_streamfile(chan, "beep", chan->language);
00157       if (!res)
00158          res = ast_waitstream(chan, "");
00159    }
00160 
00161    if (!res) {
00162       snprintf(meetmeopts, sizeof(meetmeopts), "%ud|A%s%sqxd", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "t"), 
00163          (ast_test_flag(&flags, PAGE_RECORD) ? "r" : "") );
00164       pbx_exec(chan, app, meetmeopts);
00165    }
00166 
00167    /* Go through each dial attempt cancelling, joining, and destroying */
00168    for (i = 0; i < pos; i++) {
00169       struct ast_dial *dial = dials[i];
00170 
00171       /* We have to wait for the async thread to exit as it's possible Meetme won't throw them out immediately */
00172       ast_dial_join(dial);
00173 
00174       /* Hangup all channels */
00175       ast_dial_hangup(dial);
00176 
00177       /* Destroy dialing structure */
00178       ast_dial_destroy(dial);
00179    }
00180 
00181    ast_module_user_remove(u);
00182 
00183    return -1;
00184 }
00185 
00186 static int unload_module(void)
00187 {
00188    int res;
00189 
00190    res =  ast_unregister_application(app_page);
00191 
00192    ast_module_user_hangup_all();
00193 
00194    return res;
00195 }
00196 
00197 static int load_module(void)
00198 {
00199    return ast_register_application(app_page, page_exec, page_synopsis, page_descrip);
00200 }
00201 
00202 AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Page Multiple Phones");
00203 

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