(closes issue #9690)

Reported by: mattv
Make rtp timeouts work even if two RTP streams are directly bridged in the RTP stack.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@81331 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 54dc38262b20e705befb0bd5705066083f8d111a..6fe82e0dcaeb347c487c2390776f0ceeecfa1a3a 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -15318,15 +15318,12 @@ restartsearch:
                                                                ast_mutex_lock(&sip->lock);
                                                        }
                                                        if (sip->owner) {
-                                                               if (!(ast_rtp_get_bridged(sip->rtp))) {
-                                                                       ast_log(LOG_NOTICE,
-                                                                               "Disconnecting call '%s' for lack of RTP activity in %ld seconds\n",
-                                                                               sip->owner->name,
-                                                                               (long) (t - sip->lastrtprx));
-                                                                       /* Issue a softhangup */
-                                                                       ast_softhangup_nolock(sip->owner, AST_SOFTHANGUP_DEV);
-                                                               } else
-                                                                       ast_log(LOG_NOTICE, "'%s' will not be disconnected in %ld seconds because it is directly bridged to another RTP stream\n", sip->owner->name, (long) (t - sip->lastrtprx));
+                                                               ast_log(LOG_NOTICE,
+                                                                       "Disconnecting call '%s' for lack of RTP activity in %ld seconds\n",
+                                                                       sip->owner->name,
+                                                                       (long) (t - sip->lastrtprx));
+                                                               /* Issue a softhangup */
+                                                               ast_softhangup_nolock(sip->owner, AST_SOFTHANGUP_DEV);
                                                                ast_channel_unlock(sip->owner);
                                                                /* forget the timeouts for this call, since a hangup
                                                                   has already been requested and we don't want to