[svn-commits] mmichelson: trunk r153223 - in /trunk: ./ apps/ include/asterisk/ main/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Fri Oct 31 15:05:46 CDT 2008
Author: mmichelson
Date: Fri Oct 31 15:05:46 2008
New Revision: 153223
URL: http://svn.digium.com/view/asterisk?view=rev&rev=153223
Log:
* 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.
Modified:
trunk/CHANGES
trunk/apps/app_page.c
trunk/include/asterisk/dial.h
trunk/main/dial.c
Modified: trunk/CHANGES
URL: http://svn.digium.com/view/asterisk/trunk/CHANGES?view=diff&rev=153223&r1=153222&r2=153223
==============================================================================
--- trunk/CHANGES (original)
+++ trunk/CHANGES Fri Oct 31 15:05:46 2008
@@ -685,6 +685,7 @@
WaitForRing() now takes floating pt timeout arg.
SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
* Added 's' option to Page application.
+ * Added an optional timeout argument to the Page application.
* Added 'E', 'V', and 'P' commands to ExternalIVR.
* Added 'o' and 'X' options to Chanspy.
* Added a new dialplan application, Bridge, which allows you to bridge the
Modified: trunk/apps/app_page.c
URL: http://svn.digium.com/view/asterisk/trunk/apps/app_page.c?view=diff&rev=153223&r1=153222&r2=153223
==============================================================================
--- trunk/apps/app_page.c (original)
+++ trunk/apps/app_page.c Fri Oct 31 15:05:46 2008
@@ -49,7 +49,7 @@
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 @@
" 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 @@
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 @@
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';
-
- tmp = strsep(&options, ",");
- if (options)
- ast_app_parse_options(page_opts, &flags, opts, options);
+ }
+
+ if (!ast_strlen_zero(args.options)) {
+ ast_app_parse_options(page_opts, &flags, opts, args.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;
@@ -143,10 +161,17 @@
}
/* 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);
Modified: trunk/include/asterisk/dial.h
URL: http://svn.digium.com/view/asterisk/trunk/include/asterisk/dial.h?view=diff&rev=153223&r1=153222&r2=153223
==============================================================================
--- trunk/include/asterisk/dial.h (original)
+++ trunk/include/asterisk/dial.h Fri Oct 31 15:05:46 2008
@@ -154,7 +154,7 @@
/*! \brief Set the maximum time (globally) allowed for trying to ring phones
* \param dial The dial structure to apply the time limit to
- * \param timeout Maximum time allowed
+ * \param timeout Maximum time allowed in milliseconds
* \return nothing
*/
void ast_dial_set_global_timeout(struct ast_dial *dial, int timeout);
@@ -162,7 +162,7 @@
/*! \brief Set the maximum time (per channel) allowed for trying to ring the phone
* \param dial The dial structure the channel belongs to
* \param num Channel number to set timeout on
- * \param timeout Maximum time allowed
+ * \param timeout Maximum time allowed in milliseconds
* \return nothing
*/
void ast_dial_set_timeout(struct ast_dial *dial, int num, int timeout);
Modified: trunk/main/dial.c
URL: http://svn.digium.com/view/asterisk/trunk/main/dial.c?view=diff&rev=153223&r1=153222&r2=153223
==============================================================================
--- trunk/main/dial.c (original)
+++ trunk/main/dial.c Fri Oct 31 15:05:46 2008
@@ -1038,7 +1038,7 @@
{
dial->timeout = timeout;
- if (dial->timeout > 0 && dial->actual_timeout > dial->timeout)
+ if (dial->timeout > 0 && (dial->actual_timeout > dial->timeout || dial->actual_timeout == -1))
dial->actual_timeout = dial->timeout;
return;
@@ -1059,7 +1059,7 @@
channel->timeout = timeout;
- if (channel->timeout > 0 && dial->actual_timeout > channel->timeout)
+ if (channel->timeout > 0 && (dial->actual_timeout > channel->timeout || dial->actual_timeout == -1))
dial->actual_timeout = channel->timeout;
return;
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