automerge commit

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2-netsec@19391 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Makefile

@@ -652,7 +652,6 @@ bininstall: all
 	if [ -n "$(OLDHEADERS)" ]; then \
 		rm -f $(addprefix $(DESTDIR)$(ASTHEADERDIR)/,$(OLDHEADERS)) ;\
 	fi
-	rm -f $(DESTDIR)$(ASTVARLIBDIR)/sounds/voicemail
 	mkdir -p $(DESTDIR)$(ASTVARLIBDIR)/sounds
 	mkdir -p $(DESTDIR)$(ASTLOGDIR)/cdr-csv
 	mkdir -p $(DESTDIR)$(ASTLOGDIR)/cdr-custom
@@ -671,7 +670,6 @@ bininstall: all
 	else \
 		echo "You need to do cvs update -d not just cvs update" ; \
 	fi 
-	( cd $(DESTDIR)$(ASTVARLIBDIR)/sounds  ; ln -s $(ASTSPOOLDIR)/voicemail . )
 	if [ -f mpg123-0.59r/mpg123 ]; then $(MAKE) -C mpg123-0.59r install; fi
 	@echo " +---- Asterisk Installation Complete -------+"  
 	@echo " +                                           +"

apps/app_page.c

@@ -1,7 +1,7 @@
 /*
  * Asterisk -- An open source telephony toolkit.
  *
- * Copyright (c) 2004 - 2005 Digium, Inc.  All rights reserved.
+ * Copyright (c) 2004 - 2006 Digium, Inc.  All rights reserved.
  *
  * Mark Spencer <markster@digium.com>
  *
@@ -149,6 +149,7 @@ static int page_exec(struct ast_channel *chan, void *data)
 	struct ast_app *app;
 	char *tmp;
 	int res=0;
+	char originator[AST_CHANNEL_NAME];
 
 	if (ast_strlen_zero(data)) {
 		ast_log(LOG_WARNING, "This application requires at least one argument (destination(s) to page)\n");
@@ -175,7 +176,16 @@ static int page_exec(struct ast_channel *chan, void *data)
 		ast_app_parse_options(page_opts, &flags, NULL, options);
 
 	snprintf(meetmeopts, sizeof(meetmeopts), "%ud|%sqxdw", confid, ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "m");
+
+	ast_copy_string(originator, chan->name, sizeof(originator));
+	if ((tmp = strchr(originator, '-')))
+		*tmp = '\0';
+
 	while ((tech = strsep(&tmp, "&"))) {
+		/* don't call the originating device */
+		if (!strcasecmp(tech, originator))
+			continue;
+
 		if ((resource = strchr(tech, '/'))) {
 			*resource++ = '\0';
 			launch_page(chan, meetmeopts, tech, resource);
@@ -183,6 +193,7 @@ static int page_exec(struct ast_channel *chan, void *data)
 			ast_log(LOG_WARNING, "Incomplete destination '%s' supplied.\n", tech);
 		}
 	}
+
 	if (!ast_test_flag(&flags, PAGE_QUIET)) {
 		res = ast_streamfile(chan, "beep", chan->language);
 		if (!res)

asterisk.c

@@ -615,6 +615,8 @@ static void *listener(void *unused)
 					consoles[x].fd = s;
 					if (ast_pthread_create(&consoles[x].t, &attr, netconsole, &consoles[x])) {
 						ast_log(LOG_ERROR, "Unable to spawn thread to handle connection: %s\n", strerror(errno));
+						close(consoles[x].p[0]);
+						close(consoles[x].p[1]);
 						consoles[x].fd = -1;
 						fdprint(s, "Server failed to spawn thread\n");
 						close(s);

channel.c

@@ -1,7 +1,7 @@
 /*
  * Asterisk -- An open source telephony toolkit.
  *
- * Copyright (C) 1999 - 2005, Digium, Inc.
+ * Copyright (C) 1999 - 2006, Digium, Inc.
  *
  * Mark Spencer <markster@digium.com>
  *
@@ -1135,7 +1135,6 @@ static void queue_frame_to_spies(struct ast_channel *chan, struct ast_frame *f,
 	struct ast_frame *translated_frame = NULL;
 	struct ast_channel_spy *spy;
 	struct ast_channel_spy_queue *queue;
-	struct ast_channel_spy_queue *other_queue;
 	struct channel_spy_trans *trans;
 	struct ast_frame *last;
 
@@ -1197,9 +1196,6 @@ static void queue_frame_to_spies(struct ast_channel *chan, struct ast_frame *f,
 
 		if (queue->samples > SPY_QUEUE_SAMPLE_LIMIT) {
 			if (ast_test_flag(spy, CHANSPY_TRIGGER_MODE) != CHANSPY_TRIGGER_NONE) {
-				other_queue = (dir == SPY_WRITE) ? &spy->read_queue : &spy->write_queue;
-
-				if (other_queue->samples == 0) {
 				switch (ast_test_flag(spy, CHANSPY_TRIGGER_MODE)) {
 				case CHANSPY_TRIGGER_READ:
 					if (dir == SPY_WRITE) {
@@ -1225,11 +1221,7 @@ static void queue_frame_to_spies(struct ast_channel *chan, struct ast_frame *f,
 						spy->type, chan->name);
 				ast_set_flag(spy, CHANSPY_TRIGGER_FLUSH);
 				ast_cond_signal(&spy->trigger);
-					ast_mutex_unlock(&spy->lock);
+			} else {
-					continue;
-				}
-			}
-
 				if (option_debug)
 					ast_log(LOG_DEBUG, "Spy '%s' on channel '%s' %s queue too long, dropping frames\n",
 						spy->type, chan->name, (dir == SPY_READ) ? "read" : "write");
@@ -1240,6 +1232,7 @@ static void queue_frame_to_spies(struct ast_channel *chan, struct ast_frame *f,
 					queue->head = drop->next;
 					ast_frfree(drop);
 				}
+			}
 		} else {
 			switch (ast_test_flag(spy, CHANSPY_TRIGGER_MODE)) {
 			case CHANSPY_TRIGGER_READ:

pbx/pbx_config.c

@@ -1,7 +1,7 @@
 /*
  * Asterisk -- An open source telephony toolkit.
  *
- * Copyright (C) 1999 - 2005, Digium, Inc.
+ * Copyright (C) 1999 - 2006, Digium, Inc.
  *
  * Mark Spencer <markster@digium.com>
  *
@@ -1808,7 +1808,6 @@ int load_module(void)
 
 int reload(void)
 {
-	ast_context_destroy(NULL, registrar);
 	if (clearglobalvars_config)
 		pbx_builtin_clear_globals();
 	pbx_load_module();