* Fixed timeout logic in the dialing API as setting timeouts

  had no effect
* Updated dialing API documentation to indicate that timeouts
  are specified in milliseconds
* Added a new timeout argument to the Page application. If time
  expires, any endpoints which have not answered will be hung up.



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@153223 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 35 insertions, 10 deletions

diff --git a/apps/app_page.c b/apps/app_page.c
index 9f94b491d..33ff1b489 100644
--- a/apps/app_page.c
+++ b/apps/app_page.c
@@ -49,7 +49,7 @@ 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"
+"Page(Technology/Resource&Technology2/Resource2[,options][,timeout])\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"
@@ -57,7 +57,10 @@ static const char *page_descrip =
 "        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";
+"        s - only dial channel if devicestate says it is not in use\n"
+"The timeout parameter specifies the length of time that the system\n"
+"will attempt to connect a call. After this duration, any intercom\n"
+"calls that have not been answered will be hung up by the system.\n";
 
 enum {
 	PAGE_DUPLEX = (1 << 0),
@@ -77,13 +80,21 @@ AST_APP_OPTIONS(page_opts, {
 
 static int page_exec(struct ast_channel *chan, void *data)
 {
-	char *options, *tech, *resource, *tmp;
+	char *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];
+	int timeout = 0;
+	char *parse;
+
+	AST_DECLARE_APP_ARGS(args,
+		AST_APP_ARG(devices);
+		AST_APP_ARG(options);
+		AST_APP_ARG(timeout);
+	);
 
 	if (ast_strlen_zero(data)) {
 		ast_log(LOG_WARNING, "This application requires at least one argument (destination(s) to page)\n");
@@ -95,21 +106,28 @@ static int page_exec(struct ast_channel *chan, void *data)
 		return -1;
 	};
 
-	options = ast_strdupa(data);
+	parse = ast_strdupa(data);
+
+	AST_STANDARD_APP_ARGS(args, parse);
 
 	ast_copy_string(originator, chan->name, sizeof(originator));
-	if ((tmp = strchr(originator, '-')))
+	if ((tmp = strchr(originator, '-'))) {
 		*tmp = '\0';
+	}
+
+	if (!ast_strlen_zero(args.options)) {
+		ast_app_parse_options(page_opts, &flags, opts, args.options);
+	}
 
-	tmp = strsep(&options, ",");
-	if (options)
-		ast_app_parse_options(page_opts, &flags, opts, options);
+	if (!ast_strlen_zero(args.timeout)) {
+		timeout = atoi(args.timeout);
+	}
 
 	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, "&"))) {
+	while ((tech = strsep(&args.devices, "&"))) {
 		int state = 0;
 		struct ast_dial *dial = NULL;
 
@@ -143,11 +161,18 @@ static int page_exec(struct ast_channel *chan, void *data)
 		}
 
 		/* Append technology and resource */
-		ast_dial_append(dial, tech, resource);
+		if (ast_dial_append(dial, tech, resource) == -1) {
+			ast_log(LOG_ERROR, "Failed to add %s to outbound dial\n", tech);
+			continue;
+		}
 
 		/* Set ANSWER_EXEC as global option */
 		ast_dial_option_global_enable(dial, AST_DIAL_OPTION_ANSWER_EXEC, meetmeopts);
 
+		if (timeout) {
+			ast_dial_set_global_timeout(dial, timeout * 1000);
+		}
+
 		/* Run this dial in async mode */
 		ast_dial_run(dial, chan, 1);