]> git.ipfire.org Git - thirdparty/asterisk.git/commitdiff
Reinvite the RTP back to the Asterisk machine when the timeout happens. (issue #9888...
authorJoshua Colp <jcolp@digium.com>
Wed, 6 Jun 2007 13:28:34 +0000 (13:28 +0000)
committerJoshua Colp <jcolp@digium.com>
Wed, 6 Jun 2007 13:28:34 +0000 (13:28 +0000)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@67649 65c4cc65-6c06-0410-ace0-fbb531ad65f3

rtp.c

diff --git a/rtp.c b/rtp.c
index ce85b45510105d4ef3b217aac2926650ace3b6b6..4f0c8da26140b664275035ada0126aad06ce50a0 100644 (file)
--- a/rtp.c
+++ b/rtp.c
@@ -1707,8 +1707,13 @@ enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel
                }
                who = ast_waitfor_n(cs, 2, &timeoutms);
                if (!who) {
-                       if (!timeoutms) 
+                       if (!timeoutms) {
+                               if (pr0->set_rtp_peer(c0, NULL, NULL, 0, 0))
+                                       ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c0->name);
+                               if (pr1->set_rtp_peer(c1, NULL, NULL, 0, 0))
+                                       ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c1->name);
                                return AST_BRIDGE_RETRY;
+                       }
                        if (option_debug)
                                ast_log(LOG_DEBUG, "Ooh, empty read...\n");
                        /* check for hangup / whentohangup */