Fix "bug" of not being able to page multiple phones

git-svn-id: http://svn.digium.com/svn/asterisk/trunk@6753 f38db490-d61c-443f-a65b-d21fe96a405b

2 files changed, 126 insertions, 1 deletions

diff --git a/apps/Makefile b/apps/Makefile
index 5bc26a91a..9063e20c1 100755
--- a/apps/Makefile
+++ b/apps/Makefile
@@ -28,7 +28,7 @@ APPS=app_dial.so app_playback.so app_voicemail.so app_directory.so app_mp3.so\
      app_test.so app_forkcdr.so app_math.so app_realtime.so \
      app_dumpchan.so app_waitforsilence.so app_while.so app_setrdnis.so \
      app_md5.so app_readfile.so app_chanspy.so app_settransfercapability.so \
-     app_dictate.so app_externalivr.so app_directed_pickup.so
+     app_dictate.so app_externalivr.so app_directed_pickup.so app_page.so
 
 #
 # Obsolete things...
diff --git a/apps/app_page.c b/apps/app_page.c
new file mode 100755
index 000000000..f84e0b346
--- /dev/null
+++ b/apps/app_page.c
@@ -0,0 +1,125 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (c) 2004 - 2005 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.
+ *
+ */
+
+/*
+ *
+ * Page application
+ *
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/options.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+
+
+static char *tdesc = "Page Multiple Phones";
+
+static char *app_page= "Page";
+
+static char *page_synopsis = "Pages phones";
+
+static char *page_descrip =
+"Page(Technology/Resource&Technology2/Resource2)\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.  Always returns -1.\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int page_exec(struct ast_channel *chan, void *data)
+{
+	char *options;
+	char *tech, *resource;
+	char meetmeopts[80];
+	unsigned int confid = rand();
+	struct ast_app *app;
+
+	if (data) {
+		options = ast_strdupa((char *)data);
+		if (options) {
+			char *tmp = strsep(&options, "|,");
+			if (options) {
+				/* XXX Parse options if we had any XXX */
+			}
+			snprintf(meetmeopts, sizeof(meetmeopts), "%ud|mqxdw", confid);
+			while(tmp && !ast_strlen_zero(tmp)) {
+				tech = strsep(&tmp, "&");
+				if (tech) {
+					resource = strchr(tech, '/');
+					if (resource) {
+						*resource = '\0';
+						resource++;
+						ast_pbx_outgoing_app(tech, AST_FORMAT_SLINEAR, resource, 30000, "MeetMe", meetmeopts, NULL, 0, chan->cid.cid_num, chan->cid.cid_name, NULL, NULL);
+					}
+				}
+			}
+			snprintf(meetmeopts, sizeof(meetmeopts), "%ud|Atqxd", confid);
+			app = pbx_findapp("Meetme");
+			if (app) {
+				pbx_exec(chan, app, meetmeopts, 1);
+			} else
+				ast_log(LOG_WARNING, "Whoa, meetme doesn't exist!\n");
+		} else {
+			ast_log(LOG_ERROR, "Out of memory\n");
+		}
+	}
+
+	return -1;
+}
+
+int unload_module(void)
+{
+	STANDARD_HANGUP_LOCALUSERS;
+	return ast_unregister_application(app_page);
+}
+
+int load_module(void)
+{
+	return ast_register_application(app_page, page_exec, page_synopsis, page_descrip);
+}
+
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}