Sat Aug 6 00:39:20 2011

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>dahdi</depend>
00030    <depend>app_meetme</depend>
00031  ***/
00032 
00033 #include "asterisk.h"
00034 
00035 ASTERISK_FILE_VERSION(__FILE__, "$Revision: 170979 $")
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 "        A(x) - Play an announce simultaneously to all paged participants,\n"
00069 "               and also to the original caller. Use 'x' as the file.\n"
00070 "        n    - Not to play simultaneous announce to caller (implies A(x)).\n";
00071 
00072 
00073 enum {
00074    PAGE_DUPLEX = (1 << 0),
00075    PAGE_QUIET = (1 << 1),
00076    PAGE_RECORD = (1 << 2),
00077    PAGE_ANNOUNCE = (1 << 3),
00078    PAGE_NOCALLERANNOUNCE = (1 << 4),
00079 } page_opt_flags;
00080 
00081 enum {
00082    OPT_ARG_ANNOUNCE = 0,
00083    OPT_ARG_ARRAY_SIZE = 1,
00084 };
00085 
00086 AST_APP_OPTIONS(page_opts, {
00087    AST_APP_OPTION('d', PAGE_DUPLEX),
00088    AST_APP_OPTION('q', PAGE_QUIET),
00089    AST_APP_OPTION('r', PAGE_RECORD),
00090    AST_APP_OPTION_ARG('A', PAGE_ANNOUNCE, OPT_ARG_ANNOUNCE),
00091    AST_APP_OPTION('n', PAGE_NOCALLERANNOUNCE),
00092 });
00093 
00094 
00095 static int page_exec(struct ast_channel *chan, void *data)
00096 {
00097    struct ast_module_user *u;
00098    char *options, *tech, *resource, *tmp, *tmp2;
00099    char meetmeopts[128], originator[AST_CHANNEL_NAME];
00100    struct ast_flags flags = { 0 };
00101    char *flag_args[OPT_ARG_ARRAY_SIZE];
00102    unsigned int confid = ast_random();
00103    struct ast_app *app;
00104    int res = 0, pos = 0, i = 0;
00105    struct ast_dial **dial_list;
00106    unsigned int num_dials;
00107 
00108    if (ast_strlen_zero(data)) {
00109       ast_log(LOG_WARNING, "This application requires at least one argument (destination(s) to page)\n");
00110       return -1;
00111    }
00112 
00113    u = ast_module_user_add(chan);
00114 
00115    if (!(app = pbx_findapp("MeetMe"))) {
00116       ast_log(LOG_WARNING, "There is no MeetMe application available!\n");
00117       ast_module_user_remove(u);
00118       return -1;
00119    };
00120 
00121    options = ast_strdupa(data);
00122 
00123    ast_copy_string(originator, chan->name, sizeof(originator));
00124    if ((tmp = strchr(originator, '-')))
00125       *tmp = '\0';
00126 
00127    tmp = strsep(&options, "|");
00128    if (options)
00129        ast_app_parse_options(page_opts, &flags, flag_args, options);
00130   
00131    if (ast_test_flag(&flags, PAGE_ANNOUNCE) && !ast_strlen_zero(flag_args[OPT_ARG_ANNOUNCE])) {
00132        snprintf(meetmeopts, sizeof(meetmeopts), "MeetMe|%ud|%s%sqxdw(5)G(%s)", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "m"),
00133            (ast_test_flag(&flags, PAGE_RECORD) ? "r" : ""), flag_args[OPT_ARG_ANNOUNCE] );
00134  
00135    } else {
00136        snprintf(meetmeopts, sizeof(meetmeopts), "MeetMe|%ud|%s%sqxdw(5)", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "m"),
00137            (ast_test_flag(&flags, PAGE_RECORD) ? "r" : "") );
00138    }
00139 
00140    /* Count number of extensions in list by number of ampersands + 1 */
00141    num_dials = 1;
00142    tmp2 = tmp;
00143    while (*tmp2) {
00144       if (*tmp2 == '&') {
00145          num_dials++;
00146       }
00147       tmp2++;
00148    }
00149 
00150    if (!(dial_list = ast_calloc(num_dials, sizeof(struct ast_dial *)))) {
00151       ast_log(LOG_ERROR, "Can't allocate %ld bytes for dial list\n", (long)(sizeof(struct ast_dial *) * num_dials));
00152       ast_module_user_remove(u);
00153       return -1;
00154    }
00155 
00156    /* Go through parsing/calling each device */
00157    while ((tech = strsep(&tmp, "&"))) {
00158       struct ast_dial *dial = NULL;
00159 
00160       /* don't call the originating device */
00161       if (!strcasecmp(tech, originator))
00162          continue;
00163 
00164       /* If no resource is available, continue on */
00165       if (!(resource = strchr(tech, '/'))) {
00166          ast_log(LOG_WARNING, "Incomplete destination '%s' supplied.\n", tech);
00167          continue;
00168       }
00169 
00170       *resource++ = '\0';
00171 
00172       /* Create a dialing structure */
00173       if (!(dial = ast_dial_create())) {
00174          ast_log(LOG_WARNING, "Failed to create dialing structure.\n");
00175          continue;
00176       }
00177 
00178       /* Append technology and resource */
00179       ast_dial_append(dial, tech, resource);
00180 
00181       /* Set ANSWER_EXEC as global option */
00182       ast_dial_option_global_enable(dial, AST_DIAL_OPTION_ANSWER_EXEC, meetmeopts);
00183 
00184       /* Run this dial in async mode */
00185       ast_dial_run(dial, chan, 1);
00186 
00187       /* Put in our dialing array */
00188       dial_list[pos++] = dial;
00189    }
00190 
00191    if (!ast_test_flag(&flags, PAGE_QUIET)) {
00192       res = ast_streamfile(chan, "beep", chan->language);
00193       if (!res)
00194          res = ast_waitstream(chan, "");
00195    }
00196 
00197    if (!res) {
00198        /* Default behaviour */
00199        snprintf(meetmeopts, sizeof(meetmeopts), "%ud|A%s%sqxd", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "t"), 
00200            (ast_test_flag(&flags, PAGE_RECORD) ? "r" : "") );
00201        if (ast_test_flag(&flags, PAGE_ANNOUNCE) && !ast_strlen_zero(flag_args[OPT_ARG_ANNOUNCE]) &&
00202       !ast_test_flag(&flags, PAGE_NOCALLERANNOUNCE)) {
00203       snprintf(meetmeopts, sizeof(meetmeopts), "%ud|A%s%sqxdG(%s)", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "t"), 
00204           (ast_test_flag(&flags, PAGE_RECORD) ? "r" : ""), flag_args[OPT_ARG_ANNOUNCE] );
00205        }
00206        pbx_exec(chan, app, meetmeopts);
00207    }
00208 
00209    /* Go through each dial attempt cancelling, joining, and destroying */
00210    for (i = 0; i < pos; i++) {
00211       struct ast_dial *dial = dial_list[i];
00212 
00213       /* We have to wait for the async thread to exit as it's possible Meetme won't throw them out immediately */
00214       ast_dial_join(dial);
00215 
00216       /* Hangup all channels */
00217       ast_dial_hangup(dial);
00218 
00219       /* Destroy dialing structure */
00220       ast_dial_destroy(dial);
00221    }
00222 
00223    ast_free(dial_list);
00224    ast_module_user_remove(u);
00225 
00226    return -1;
00227 }
00228 
00229 static int unload_module(void)
00230 {
00231    int res;
00232 
00233    res =  ast_unregister_application(app_page);
00234 
00235    ast_module_user_hangup_all();
00236 
00237    return res;
00238 }
00239 
00240 static int load_module(void)
00241 {
00242    return ast_register_application(app_page, page_exec, page_synopsis, page_descrip);
00243 }
00244 
00245 AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Page Multiple Phones");
00246 

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