[asterisk-commits] file: branch 1.2 r67649 - /branches/1.2/rtp.c

[asterisk-commits at lists.digium.com]

Author: file
Date: Wed Jun  6 08:28:34 2007
New Revision: 67649

URL: http://svn.digium.com/view/asterisk?view=rev&rev=67649
Log:
Reinvite the RTP back to the Asterisk machine when the timeout happens. (issue #9888 reported by gasparz)

Modified:
    branches/1.2/rtp.c

Modified: branches/1.2/rtp.c
URL: http://svn.digium.com/view/asterisk/branches/1.2/rtp.c?view=diff&rev=67649&r1=67648&r2=67649
==============================================================================
--- branches/1.2/rtp.c (original)
+++ branches/1.2/rtp.c Wed Jun  6 08:28:34 2007
@@ -1707,8 +1707,13 @@
 		}
 		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 */