[svn-commits] russell: branch russell/issue_5841 r61117 -
/team/russell/issue_5841/res/
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Mon Apr 9 16:07:22 MST 2007
Author: russell
Date: Mon Apr 9 18:07:21 2007
New Revision: 61117
URL: http://svn.digium.com/view/asterisk?view=rev&rev=61117
Log:
some more code tweaks
Modified:
team/russell/issue_5841/res/res_features.c
Modified: team/russell/issue_5841/res/res_features.c
URL: http://svn.digium.com/view/asterisk/team/russell/issue_5841/res/res_features.c?view=diff&rev=61117&r1=61116&r2=61117
==============================================================================
--- team/russell/issue_5841/res/res_features.c (original)
+++ team/russell/issue_5841/res/res_features.c Mon Apr 9 18:07:21 2007
@@ -2054,10 +2054,10 @@
static char mandescr_bridge[] =
"Description: Bridge together two channels already in the PBX\n"
-"Variables: ( Names marked with * are required )\n"
-" *Channel1: Channel to Bridge to Channel2\n"
-" *Channel2: Channel to Bridge to Channel1\n"
-" Tone: (Yes|No) Play courtesy tone to Channel 2\n"
+"Variables: ( Headers marked with * are required )\n"
+" *Channel1: Channel to Bridge to Channel2\n"
+" *Channel2: Channel to Bridge to Channel1\n"
+" Tone: (Yes|No) Play courtesy tone to Channel 2\n"
"\n";
static void do_bridge_masquerade(struct ast_channel *chan, struct ast_channel *tmpchan)
@@ -2611,23 +2611,35 @@
static char *app_bridge = "Bridge";
static char *bridge_synopsis = "Bridge two channels";
static char *bridge_descrip =
-"Usage: Bridge(CHANNEL|PLAYTONE)\n"
+"Usage: Bridge(channel[|options])\n"
" Allows the ability to bridge two channels via the dialplan.\n"
-"The current channel is bridged to the specified CHANNEL.\n"
-"Adding PLAYTONE will play a courtesy audio tone on CHANNEL when bridged successfully.\n"
+"The current channel is bridged to the specified 'channel'.\n"
+"The following options are supported:\n"
+" p - Play a courtesy tone to 'channel'.\n"
"BRIDGERESULT dial plan variable will contain SUCCESS, FAILURE, LOOP, NONEXISTENT or INCOMPATIBLE.\n";
+
+enum {
+ BRIDGE_OPT_PLAYTONE,
+ /* this entry must be the last one. */
+ BRIDGE_OPT_ARRAY_SIZE,
+};
+
+AST_APP_OPTIONS(bridge_exec_options, BEGIN_OPTIONS
+ AST_APP_OPTION('p', BRIDGE_OPT_PLAYTONE)
+END_OPTIONS );
static int bridge_exec(struct ast_channel *chan, void *data)
{
struct ast_module_user *u;
struct ast_channel *current_dest_chan = NULL, *final_dest_chan = NULL;
int res, pbx_res = 0;
- int playtone = 0;
char *tmp_data = NULL;
+ struct ast_flags opts = { 0, };
+ struct ast_bridge_config bconfig = { { 0, }, };
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(dest_chan);
- AST_APP_ARG(playtone);
+ AST_APP_ARG(options);
);
if (ast_strlen_zero(data)) {
@@ -2638,20 +2650,14 @@
u = ast_module_user_add(chan);
tmp_data = ast_strdupa(data);
- if (!tmp_data) {
- ast_log(LOG_ERROR, "Out of memory!\n");
- ast_module_user_remove(u);
- return -1;
- }
-
AST_STANDARD_APP_ARGS(args, tmp_data);
- if (args.playtone) {
- playtone = 1;
- }
-
+ if (!ast_strlen_zero(args.options))
+ ast_app_parse_options(bridge_exec_options, &opts, NULL, args.options);
+
/* avoid bridge with ourselves */
- int cmplen = strlen(chan->name) < strlen(args.dest_chan) ? strlen(chan->name) : strlen(args.dest_chan);
- if (0 == strncmp(chan->name, args.dest_chan, cmplen)) {
+ if (!strncmp(chan->name, args.dest_chan,
+ strlen(chan->name) < strlen(args.dest_chan) ?
+ strlen(chan->name) : strlen(args.dest_chan))) {
ast_log(LOG_WARNING, "Unable to bridge channel %s with itself\n", chan->name);
manager_event(EVENT_FLAG_CALL, "BridgeExec",
"Response: Failed\r\n"
@@ -2714,23 +2720,14 @@
"Response: Success\r\n"
"Channel1: %s\r\n"
"Channel2: %s\r\n", chan->name, final_dest_chan->name);
+
/* we have 2 valid channels to bridge, now it is just a matter of setting up the bridge config and starting the bridge */
- struct ast_bridge_config bconfig;
- bconfig.play_warning = 0;
- bconfig.warning_freq = 0;
- bconfig.warning_sound = NULL;
- bconfig.end_sound = NULL;
- bconfig.start_sound = NULL;
- bconfig.firstpass = 0;
- bconfig.timelimit = 0;
- bconfig.feature_timer = 0;
- ast_clear_flag(&(bconfig.features_caller), AST_FLAGS_ALL);
- ast_clear_flag(&(bconfig.features_callee), AST_FLAGS_ALL);
-
- if (playtone && !ast_strlen_zero(xfersound))
- if (!ast_streamfile(final_dest_chan, xfersound, final_dest_chan->language))
+ if (ast_test_flag(&opts, BRIDGE_OPT_PLAYTONE) && !ast_strlen_zero(xfersound)) {
+ if (!ast_streamfile(final_dest_chan, xfersound, final_dest_chan->language)) {
if (ast_waitstream(final_dest_chan, "") < 0)
ast_log(LOG_WARNING, "Failed to play courtesy tone on %s\n", final_dest_chan->name);
+ }
+ }
/* do the bridge */
res = ast_bridge_call(chan, final_dest_chan, &bconfig);
@@ -2741,7 +2738,7 @@
ast_log(LOG_DEBUG, "starting new PBX in %s,%s,%d for chan %s\n", final_dest_chan->context, final_dest_chan->exten,
final_dest_chan->priority, final_dest_chan->name);
pbx_res = ast_pbx_start(final_dest_chan);
- if (AST_PBX_SUCCESS != pbx_res) {
+ if (pbx_res != AST_PBX_SUCCESS) {
ast_log(LOG_WARNING, "FAILED continuing PBX on dest chan %s\n", final_dest_chan->name);
ast_hangup(final_dest_chan);
} else
@@ -2750,9 +2747,12 @@
ast_log(LOG_DEBUG, "hangup chan %s since the other endpoint has hung up\n", final_dest_chan->name);
ast_hangup(final_dest_chan);
}
+
ast_module_user_remove(u);
+
return res;
}
+
static int reload(void)
{
return load_config();
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