Bridging: Fix bouncing native bridge

This fixes a situation in Asterisk 1.8 and 11 where ast_channel_bridge
could cause a bouncing native bridge. In the case of the
dial_LS_options test, this was a remote RTP bridge which caused the
audio path to continually cycle between Asterisk and the remote
endpoints generating a large number of SIP messages and delaying the
test long enough to cause it to fail (checking timing was part of the
test). The root cause was that the code to decide whether to use native
bridging was expecting a time-remaining value of 0 to be the default
instead of the actual default value of -1. A value of 0 or negative
numbers could also be generated by preceding code in some
circumstances. Both issues are addressed in this patch.

ASTERISK-24211 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3987/
........

Merged revisions 423006 from http://svn.asterisk.org/svn/asterisk/branches/1.8

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@423010 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/main/channel.c b/main/channel.c
index d450f415b38896a9c9d396939952cfeed11b106c..8be8fe7a311f0f7e2462705bac76ebb94bc7e921 100644 (file)
--- a/main/channel.c
+++ b/main/channel.c
@@ -7963,8 +7963,13 @@ enum ast_bridge_result ast_channel_bridge(struct ast_channel *c0, struct ast_cha
 
                if (config->timelimit) {
                        time_left_ms = config->timelimit - ast_tvdiff_ms(now, config->start_time);
-                       if (time_left_ms < to)
+                       if (time_left_ms < 0) {
+                               time_left_ms = 0;
+                       }
+
+                       if (time_left_ms < to) {
                                to = time_left_ms;
+                       }
 
                        if (time_left_ms <= 0) {
                                if (caller_warning && config->end_sound)
@@ -7972,7 +7977,7 @@ enum ast_bridge_result ast_channel_bridge(struct ast_channel *c0, struct ast_cha
                                if (callee_warning && config->end_sound)
                                        bridge_playfile(c1, c0, config->end_sound, 0);
                                *fo = NULL;
-                               res = 0;
+                               res = AST_BRIDGE_COMPLETE;
                                ast_test_suite_event_notify("BRIDGE_TIMELIMIT", "Channel1: %s\r\nChannel2: %s", ast_channel_name(c0), ast_channel_name(c1));
                                break;
                        }
@@ -8015,7 +8020,7 @@ enum ast_bridge_result ast_channel_bridge(struct ast_channel *c0, struct ast_cha
                if (ast_test_flag(ast_channel_flags(c0), AST_FLAG_ZOMBIE) || ast_check_hangup_locked(c0) ||
                    ast_test_flag(ast_channel_flags(c1), AST_FLAG_ZOMBIE) || ast_check_hangup_locked(c1)) {
                        *fo = NULL;
-                       res = 0;
+                       res = AST_BRIDGE_COMPLETE;
                        ast_debug(1, "Bridge stops because we're zombie or need a soft hangup: c0=%s, c1=%s, flags: %s,%s,%s,%s\n",
                                ast_channel_name(c0), ast_channel_name(c1),
                                ast_test_flag(ast_channel_flags(c0), AST_FLAG_ZOMBIE) ? "Yes" : "No",
@@ -8031,7 +8036,7 @@ enum ast_bridge_result ast_channel_bridge(struct ast_channel *c0, struct ast_cha
 
                if (ast_channel_tech(c0)->bridge &&
                        /* if < 1 ms remains use generic bridging for accurate timing */
-                       (!config->timelimit || to > 1000 || to == 0) &&
+                       (!config->timelimit || to > 1000 || to == -1) &&
                    (ast_channel_tech(c0)->bridge == ast_channel_tech(c1)->bridge) &&
                    !ast_channel_monitor(c0) && !ast_channel_monitor(c1) &&
                    !ast_channel_audiohooks(c0) && !ast_channel_audiohooks(c1) &&