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-rw-r--r--trunk/apps/app_page.c193
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diff --git a/trunk/apps/app_page.c b/trunk/apps/app_page.c
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+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (c) 2004 - 2006 Digium, Inc. All rights reserved.
+ *
+ * Mark Spencer <markster@digium.com>
+ *
+ * This code is released under the GNU General Public License
+ * version 2.0. See LICENSE for more information.
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ */
+
+/*! \file
+ *
+ * \brief page() - Paging application
+ *
+ * \author Mark Spencer <markster@digium.com>
+ *
+ * \ingroup applications
+ */
+
+/*** MODULEINFO
+ <depend>zaptel</depend>
+ <depend>app_meetme</depend>
+ ***/
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/file.h"
+#include "asterisk/app.h"
+#include "asterisk/chanvars.h"
+#include "asterisk/utils.h"
+#include "asterisk/devicestate.h"
+#include "asterisk/dial.h"
+
+static const char *app_page= "Page";
+
+static const char *page_synopsis = "Pages phones";
+
+static const char *page_descrip =
+"Page(Technology/Resource&Technology2/Resource2[,options])\n"
+" Places outbound calls to the given technology / resource and dumps\n"
+"them into a conference bridge as muted participants. The original\n"
+"caller is dumped into the conference as a speaker and the room is\n"
+"destroyed when the original caller leaves. Valid options are:\n"
+" d - full duplex audio\n"
+" q - quiet, do not play beep to caller\n"
+" r - record the page into a file (see 'r' for app_meetme)\n"
+" s - only dial channel if devicestate says it is not in use\n";
+
+enum {
+ PAGE_DUPLEX = (1 << 0),
+ PAGE_QUIET = (1 << 1),
+ PAGE_RECORD = (1 << 2),
+ PAGE_SKIP = (1 << 3),
+} page_opt_flags;
+
+AST_APP_OPTIONS(page_opts, {
+ AST_APP_OPTION('d', PAGE_DUPLEX),
+ AST_APP_OPTION('q', PAGE_QUIET),
+ AST_APP_OPTION('r', PAGE_RECORD),
+ AST_APP_OPTION('s', PAGE_SKIP),
+});
+
+#define MAX_DIALS 128
+
+static int page_exec(struct ast_channel *chan, void *data)
+{
+ char *options, *tech, *resource, *tmp;
+ char meetmeopts[88], originator[AST_CHANNEL_NAME], *opts[0];
+ struct ast_flags flags = { 0 };
+ unsigned int confid = ast_random();
+ struct ast_app *app;
+ int res = 0, pos = 0, i = 0;
+ struct ast_dial *dials[MAX_DIALS];
+
+ if (ast_strlen_zero(data)) {
+ ast_log(LOG_WARNING, "This application requires at least one argument (destination(s) to page)\n");
+ return -1;
+ }
+
+ if (!(app = pbx_findapp("MeetMe"))) {
+ ast_log(LOG_WARNING, "There is no MeetMe application available!\n");
+ return -1;
+ };
+
+ options = ast_strdupa(data);
+
+ ast_copy_string(originator, chan->name, sizeof(originator));
+ if ((tmp = strchr(originator, '-')))
+ *tmp = '\0';
+
+ tmp = strsep(&options, ",");
+ if (options)
+ ast_app_parse_options(page_opts, &flags, opts, options);
+
+ snprintf(meetmeopts, sizeof(meetmeopts), "MeetMe,%ud,%s%sqxdw(5)", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "m"),
+ (ast_test_flag(&flags, PAGE_RECORD) ? "r" : "") );
+
+ /* Go through parsing/calling each device */
+ while ((tech = strsep(&tmp, "&"))) {
+ int state = 0;
+ struct ast_dial *dial = NULL;
+
+ /* don't call the originating device */
+ if (!strcasecmp(tech, originator))
+ continue;
+
+ /* If no resource is available, continue on */
+ if (!(resource = strchr(tech, '/'))) {
+ ast_log(LOG_WARNING, "Incomplete destination '%s' supplied.\n", tech);
+ continue;
+ }
+
+ /* Ensure device is not in use if skip option is enabled */
+ if (ast_test_flag(&flags, PAGE_SKIP) && (state = ast_device_state(tech)) != AST_DEVICE_NOT_INUSE) {
+ ast_log(LOG_WARNING, "Destination '%s' has device state '%s'.\n", tech, devstate2str(state));
+ continue;
+ }
+
+ *resource++ = '\0';
+
+ /* Create a dialing structure */
+ if (!(dial = ast_dial_create())) {
+ ast_log(LOG_WARNING, "Failed to create dialing structure.\n");
+ continue;
+ }
+
+ /* Append technology and resource */
+ ast_dial_append(dial, tech, resource);
+
+ /* Set ANSWER_EXEC as global option */
+ ast_dial_option_global_enable(dial, AST_DIAL_OPTION_ANSWER_EXEC, meetmeopts);
+
+ /* Run this dial in async mode */
+ ast_dial_run(dial, chan, 1);
+
+ /* Put in our dialing array */
+ dials[pos++] = dial;
+ }
+
+ if (!ast_test_flag(&flags, PAGE_QUIET)) {
+ res = ast_streamfile(chan, "beep", chan->language);
+ if (!res)
+ res = ast_waitstream(chan, "");
+ }
+
+ if (!res) {
+ snprintf(meetmeopts, sizeof(meetmeopts), "%ud,A%s%sqxd", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "t"),
+ (ast_test_flag(&flags, PAGE_RECORD) ? "r" : "") );
+ pbx_exec(chan, app, meetmeopts);
+ }
+
+ /* Go through each dial attempt cancelling, joining, and destroying */
+ for (i = 0; i < pos; i++) {
+ struct ast_dial *dial = dials[i];
+
+ /* We have to wait for the async thread to exit as it's possible Meetme won't throw them out immediately */
+ ast_dial_join(dial);
+
+ /* Hangup all channels */
+ ast_dial_hangup(dial);
+
+ /* Destroy dialing structure */
+ ast_dial_destroy(dial);
+ }
+
+ return -1;
+}
+
+static int unload_module(void)
+{
+ return ast_unregister_application(app_page);
+}
+
+static int load_module(void)
+{
+ return ast_register_application(app_page, page_exec, page_synopsis, page_descrip);
+}
+
+AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Page Multiple Phones");
+