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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: 170980 $")
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 /*** DOCUMENTATION
00048    <application name="Page" language="en_US">
00049       <synopsis>
00050          Page series of phones
00051       </synopsis>
00052       <syntax>
00053          <parameter name="Technology/Resource" required="true" argsep="&amp;">
00054             <argument name="Technology/Resource" required="true">
00055                <para>Specification of the device(s) to dial. These must be in the format of
00056                <literal>Technology/Resource</literal>, where <replaceable>Technology</replaceable>
00057                represents a particular channel driver, and <replaceable>Resource</replaceable> represents a resource
00058                available to that particular channel driver.</para>
00059             </argument>
00060             <argument name="Technology2/Resource2" multiple="true">
00061                <para>Optional extra devices to dial inparallel</para>
00062                <para>If you need more then one enter them as Technology2/Resource2&amp;
00063                Technology3/Resourse3&amp;.....</para>
00064             </argument>
00065          </parameter>
00066          <parameter name="options">
00067             <optionlist>
00068                <option name="d">
00069                   <para>Full duplex audio</para>
00070                </option>
00071                <option name="i">
00072                   <para>Ignore attempts to forward the call</para>
00073                </option>
00074                <option name="q">
00075                   <para>Quiet, do not play beep to caller</para>
00076                </option>
00077                <option name="r">
00078                   <para>Record the page into a file (meetme option <literal>r</literal>)</para>
00079                </option>
00080                <option name="s">
00081                   <para>Only dial a channel if its device state says that it is <literal>NOT_INUSE</literal></para>
00082                </option>
00083             </optionlist>
00084          </parameter>
00085          <parameter name="timeout">
00086             <para>Specify the length of time that the system will attempt to connect a call.
00087             After this duration, any intercom calls that have not been answered will be hung up by the
00088             system.</para>
00089          </parameter>
00090       </syntax>
00091       <description>
00092          <para>Places outbound calls to the given <replaceable>technology</replaceable>/<replaceable>resource</replaceable>
00093          and dumps them into a conference bridge as muted participants. The original
00094          caller is dumped into the conference as a speaker and the room is
00095          destroyed when the original callers leaves.</para>
00096       </description>
00097       <see-also>
00098          <ref type="application">MeetMe</ref>
00099       </see-also>
00100    </application>
00101  ***/
00102 static const char *app_page= "Page";
00103 
00104 enum {
00105    PAGE_DUPLEX = (1 << 0),
00106    PAGE_QUIET = (1 << 1),
00107    PAGE_RECORD = (1 << 2),
00108    PAGE_SKIP = (1 << 3),
00109    PAGE_IGNORE_FORWARDS = (1 << 4),
00110 } page_opt_flags;
00111 
00112 AST_APP_OPTIONS(page_opts, {
00113    AST_APP_OPTION('d', PAGE_DUPLEX),
00114    AST_APP_OPTION('q', PAGE_QUIET),
00115    AST_APP_OPTION('r', PAGE_RECORD),
00116    AST_APP_OPTION('s', PAGE_SKIP),
00117    AST_APP_OPTION('i', PAGE_IGNORE_FORWARDS),
00118 });
00119 
00120 
00121 static int page_exec(struct ast_channel *chan, void *data)
00122 {
00123    char *tech, *resource, *tmp;
00124    char meetmeopts[88], originator[AST_CHANNEL_NAME], *opts[0];
00125    struct ast_flags flags = { 0 };
00126    unsigned int confid = ast_random();
00127    struct ast_app *app;
00128    int res = 0, pos = 0, i = 0;
00129    struct ast_dial **dial_list;
00130    unsigned int num_dials;
00131    int timeout = 0;
00132    char *parse;
00133 
00134    AST_DECLARE_APP_ARGS(args,
00135       AST_APP_ARG(devices);
00136       AST_APP_ARG(options);
00137       AST_APP_ARG(timeout);
00138    );
00139 
00140    if (ast_strlen_zero(data)) {
00141       ast_log(LOG_WARNING, "This application requires at least one argument (destination(s) to page)\n");
00142       return -1;
00143    }
00144 
00145    if (!(app = pbx_findapp("MeetMe"))) {
00146       ast_log(LOG_WARNING, "There is no MeetMe application available!\n");
00147       return -1;
00148    };
00149 
00150    parse = ast_strdupa(data);
00151 
00152    AST_STANDARD_APP_ARGS(args, parse);
00153 
00154    ast_copy_string(originator, chan->name, sizeof(originator));
00155    if ((tmp = strchr(originator, '-'))) {
00156       *tmp = '\0';
00157    }
00158 
00159    if (!ast_strlen_zero(args.options)) {
00160       ast_app_parse_options(page_opts, &flags, opts, args.options);
00161    }
00162 
00163    if (!ast_strlen_zero(args.timeout)) {
00164       timeout = atoi(args.timeout);
00165    }
00166 
00167    snprintf(meetmeopts, sizeof(meetmeopts), "MeetMe,%ud,%s%sqxdw(5)", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "m"),
00168       (ast_test_flag(&flags, PAGE_RECORD) ? "r" : "") );
00169 
00170    /* Count number of extensions in list by number of ampersands + 1 */
00171    num_dials = 1;
00172    tmp = args.devices;
00173    while (*tmp) {
00174       if (*tmp == '&') {
00175          num_dials++;
00176       }
00177       tmp++;
00178    }
00179 
00180    if (!(dial_list = ast_calloc(num_dials, sizeof(struct ast_dial *)))) {
00181       ast_log(LOG_ERROR, "Can't allocate %ld bytes for dial list\n", (long)(sizeof(struct ast_dial *) * num_dials));
00182       return -1;
00183    }
00184 
00185    /* Go through parsing/calling each device */
00186    while ((tech = strsep(&args.devices, "&"))) {
00187       int state = 0;
00188       struct ast_dial *dial = NULL;
00189 
00190       /* don't call the originating device */
00191       if (!strcasecmp(tech, originator))
00192          continue;
00193 
00194       /* If no resource is available, continue on */
00195       if (!(resource = strchr(tech, '/'))) {
00196          ast_log(LOG_WARNING, "Incomplete destination '%s' supplied.\n", tech);
00197          continue;
00198       }
00199 
00200       /* Ensure device is not in use if skip option is enabled */
00201       if (ast_test_flag(&flags, PAGE_SKIP)) {
00202          state = ast_device_state(tech);
00203          if (state == AST_DEVICE_UNKNOWN) {
00204             ast_log(LOG_WARNING, "Destination '%s' has device state '%s'. Paging anyway.\n", tech, ast_devstate2str(state));
00205          } else if (state != AST_DEVICE_NOT_INUSE) {
00206             ast_log(LOG_WARNING, "Destination '%s' has device state '%s'.\n", tech, ast_devstate2str(state));
00207             continue;
00208          }
00209       }
00210 
00211       *resource++ = '\0';
00212 
00213       /* Create a dialing structure */
00214       if (!(dial = ast_dial_create())) {
00215          ast_log(LOG_WARNING, "Failed to create dialing structure.\n");
00216          continue;
00217       }
00218 
00219       /* Append technology and resource */
00220       if (ast_dial_append(dial, tech, resource) == -1) {
00221          ast_log(LOG_ERROR, "Failed to add %s to outbound dial\n", tech);
00222          continue;
00223       }
00224 
00225       /* Set ANSWER_EXEC as global option */
00226       ast_dial_option_global_enable(dial, AST_DIAL_OPTION_ANSWER_EXEC, meetmeopts);
00227 
00228       if (timeout) {
00229          ast_dial_set_global_timeout(dial, timeout * 1000);
00230       }
00231 
00232       if (ast_test_flag(&flags, PAGE_IGNORE_FORWARDS)) {
00233          ast_dial_option_global_enable(dial, AST_DIAL_OPTION_DISABLE_CALL_FORWARDING, NULL);
00234       }
00235 
00236       /* Run this dial in async mode */
00237       ast_dial_run(dial, chan, 1);
00238 
00239       /* Put in our dialing array */
00240       dial_list[pos++] = dial;
00241    }
00242 
00243    if (!ast_test_flag(&flags, PAGE_QUIET)) {
00244       res = ast_streamfile(chan, "beep", chan->language);
00245       if (!res)
00246          res = ast_waitstream(chan, "");
00247    }
00248 
00249    if (!res) {
00250       snprintf(meetmeopts, sizeof(meetmeopts), "%ud,A%s%sqxd", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "t"), 
00251          (ast_test_flag(&flags, PAGE_RECORD) ? "r" : "") );
00252       pbx_exec(chan, app, meetmeopts);
00253    }
00254 
00255    /* Go through each dial attempt cancelling, joining, and destroying */
00256    for (i = 0; i < pos; i++) {
00257       struct ast_dial *dial = dial_list[i];
00258 
00259       /* We have to wait for the async thread to exit as it's possible Meetme won't throw them out immediately */
00260       ast_dial_join(dial);
00261 
00262       /* Hangup all channels */
00263       ast_dial_hangup(dial);
00264 
00265       /* Destroy dialing structure */
00266       ast_dial_destroy(dial);
00267    }
00268 
00269    return -1;
00270 }
00271 
00272 static int unload_module(void)
00273 {
00274    return ast_unregister_application(app_page);
00275 }
00276 
00277 static int load_module(void)
00278 {
00279    return ast_register_application_xml(app_page, page_exec);
00280 }
00281 
00282 AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Page Multiple Phones");
00283