Merged revisions 317865 via svnmerge from
[asterisk/asterisk.git] / channels / chan_sip.c
index 0712e8c..eeff890 100644 (file)
@@ -24933,10 +24933,11 @@ static void check_rtp_timeout(struct sip_pvt *dialog, time_t t)
                if (!ast_test_flag(&dialog->flags[1], SIP_PAGE2_CALL_ONHOLD) || (ast_rtp_instance_get_hold_timeout(dialog->rtp) && (t > dialog->lastrtprx + ast_rtp_instance_get_hold_timeout(dialog->rtp)))) {
                        /* Needs a hangup */
                        if (ast_rtp_instance_get_timeout(dialog->rtp)) {
                if (!ast_test_flag(&dialog->flags[1], SIP_PAGE2_CALL_ONHOLD) || (ast_rtp_instance_get_hold_timeout(dialog->rtp) && (t > dialog->lastrtprx + ast_rtp_instance_get_hold_timeout(dialog->rtp)))) {
                        /* Needs a hangup */
                        if (ast_rtp_instance_get_timeout(dialog->rtp)) {
-                               if(ast_channel_trylock(dialog->owner)) {
-                               /* Dont do a infinite deadlock avoidance loop.
-                                * Lets try this on next round (1 ms to 1000 ms later)
-                                * call is allready dead */
+                               if (!dialog->owner || ast_channel_trylock(dialog->owner)) {
+                                       /*
+                                        * Don't block, just try again later.
+                                        * If there was no owner, the call is dead already.
+                                        */
                                        return;
                                }
                                ast_log(LOG_NOTICE, "Disconnecting call '%s' for lack of RTP activity in %ld seconds\n",
                                        return;
                                }
                                ast_log(LOG_NOTICE, "Disconnecting call '%s' for lack of RTP activity in %ld seconds\n",