.gitreview

 [gerrit]
+defaultbranch=13
+#
+# Intentional padding to ensure it is possible to point a commit
+# to an alternative gerrit server/repository without breaking
+# cherry-pick between branches.
+#
 host=gerrit.asterisk.org
 port=29418
 project=asterisk.git
-defaultbranch=13

.version

-13.20.0
\ No newline at end of file
+13.22.0
\ No newline at end of file

CHANGES

@@ -8,6 +8,85 @@
 ===
 ==============================================================================
 
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 13.21.0 to Asterisk 13.22.0 ----------
+------------------------------------------------------------------------------
+
+Core
+------------------
+ * Core bridging and, more specifically, bridge_softmix have been enhanced to
+   relay received frames of type TEXT or TEXT_DATA to all participants in a
+   softmix bridge.  res_pjsip_messaging and chan_pjsip have been enhanced to
+   take advantage of this so when res_pjsip_messaging receives an in-dialog
+   MESSAGE message from a user in a conference call, it's relayed to all
+   other participants in the call.
+
+app_sendtext
+------------------
+    Support Enhanced Messaging.  SendText now accepts new channel variables
+    that can be used to override the To and From display names and set the
+    Content-Type of a message.  Since you can now set Content-Type, other
+    text/* content types are now valid.
+
+app_confbridge
+------------------
+  * ConfbridgeList now shows talking status. This utilizes the same voice
+    detection as the ConfbridgeTalking event, so bridges must be configured
+    with "talk_detection_events=yes" for this flag to have meaning.
+
+res_pjsip
+------------------
+  * Two new options have been added to the system and endpoint objects to
+    control whether, on outbound calls, Asterisk will accept updated SDP answers
+    during the initial INVITE transaction when 100rel is not in effect.
+    This usually happens when the INVITE is forked to multiple UASs and more
+    than one sends an SDP answer or when a single UAS needs to change a media
+    port to switch from custom ringback to the actual media destination.
+
+    The 'follow_early_media_forked' option sets whether Asterisk will accept
+    the updated SDP when the To tag on the subsequent response is different than
+    that on the the previous response.  This usually occurs in the forked INVITE
+    scenario. The default value is "yes" which is the current behavior.
+
+    The 'accept_multiple_sdp_answers' flag sets whether Asterisk will accept the
+    updated SDP when the To tag on the subsequent response is the same as that
+    on the previous response. This can occur when a UAS needs to switch media
+    ports from custom ringback to the final media path.  The default value is
+    "no" which is the current behavior.
+
+    These options have to be enabled system-wide in the system config section
+    of pjsip.conf as well as on individual endpoints that require the
+    functionality.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 13.20.0 to Asterisk 13.21.0 ----------
+------------------------------------------------------------------------------
+
+Build System
+------------------
+ * RADIUS backends for CEL and CDR can now also be built using the radcli
+   client library, in addition to the existing support for building them
+   using either freeradius or radiusclient-ng.
+
+Core
+------------------
+ * A new configuration option "genericplc_on_equal_codecs" was added to the
+   "plc" section of codecs.conf to allow generic packet loss concealment even
+   if no transcoding was originally needed.  Transcoding via SLIN is forced
+   in this case.
+
+res_pjproject
+------------------
+ * Added the "cache_pools" option to pjproject.conf.  Disabling the option
+   helps track down pool content mismanagement when using valgrind or
+   MALLOC_DEBUG.  The cache gets in the way of determining if the pool contents
+   are used after free and who freed it.
+
+res_pjsip_notify
+------------------
+ * Extend the PJSIPNotify AMI command to send an in-dialog notify on a
+   channel.
+
 ------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 13.19.0 to Asterisk 13.20.0 ----------
 ------------------------------------------------------------------------------

Makefile

@@ -218,7 +218,9 @@ ifeq ($(AST_DEVMODE),yes)
   endif
 endif
 
-ifneq ($(findstring BSD,$(OSARCH)),)
+ifeq ($(OSARCH),NetBSD)
+  _ASTCFLAGS+=-isystem /usr/pkg/include
+else ifneq ($(findstring BSD,$(OSARCH)),)
   _ASTCFLAGS+=-isystem /usr/local/include
 endif
 
@@ -229,7 +231,7 @@ ifeq ($(OSARCH),FreeBSD)
 endif
 
 ifeq ($(OSARCH),NetBSD)
-  _ASTCFLAGS+=-pthread -I/usr/pkg/include
+  _ASTCFLAGS+=-pthread -D__LIBPTHREAD_SOURCE__ -I/usr/pkg/include
 endif
 
 ifeq ($(OSARCH),OpenBSD)
@@ -240,7 +242,9 @@ ifeq ($(OSARCH),SunOS)
   _ASTCFLAGS+=-Wcast-align -DSOLARIS -I../include/solaris-compat -I/opt/ssl/include -I/usr/local/ssl/include -D_XPG4_2 -D__EXTENSIONS__
 endif
 
-ifneq ($(GREP),)
+ifeq ($(GREP),)
+else ifeq ($(GREP),:)
+else
   ASTERISKVERSION:=$(shell GREP=$(GREP) AWK=$(AWK) GIT=$(GIT) build_tools/make_version .)
 endif
 ifneq ($(AWK),)
@@ -283,7 +287,9 @@ else
 # These are used for all but Darwin
   SOLINK=-shared
   DYLINK=$(SOLINK)
-  ifneq ($(findstring BSD,$(OSARCH)),)
+  ifeq ($(OSARCH),NetBSD)
+    _ASTLDFLAGS+=-L/usr/pkg/lib
+  else ifneq ($(findstring BSD,$(OSARCH)),)
     _ASTLDFLAGS+=-L/usr/local/lib
   endif
 endif
@@ -468,7 +474,9 @@ endif
 		$(INSTALL) -m 644 $$x "$(DESTDIR)$(ASTDATADIR)/rest-api" ; \
 	done
 
-ifneq ($(GREP),)
+ifeq ($(GREP),)
+else ifeq ($(GREP),:)
+else
   XML_core_en_US = $(foreach dir,$(MOD_SUBDIRS),$(shell $(GREP) -l "language=\"en_US\"" $(dir)/*.c $(dir)/*.cc 2>/dev/null))
 endif
 
@@ -487,7 +495,9 @@ doc/core-en_US.xml: makeopts .lastclean $(XML_core_en_US)
 	@echo
 	@echo "</docs>" >> $@
 
-ifneq ($(GREP),)
+ifeq ($(GREP),)
+else ifeq ($(GREP),:)
+else
   XML_full_en_US = $(foreach dir,$(MOD_SUBDIRS),$(shell $(GREP) -l "language=\"en_US\"" $(dir)/*.c $(dir)/*.cc 2>/dev/null))
 endif
 
@@ -502,7 +512,7 @@ else
 	@echo "<!DOCTYPE docs SYSTEM \"appdocsxml.dtd\">" >> $@
 	@echo "<?xml-stylesheet type=\"text/xsl\" href=\"appdocsxml.xslt\"?>" >> $@
 	@echo "<docs xmlns:xi=\"http://www.w3.org/2001/XInclude\">" >> $@
-	@for x in $(MOD_SUBDIRS); do \
+	@for x in $(filter-out third-party,$(MOD_SUBDIRS)); do \
 		printf "$$x " ; \
 		for i in `find $$x -name '*.c'`; do \
 			$(PYTHON) build_tools/get_documentation.py < $$i >> $@ ; \
@@ -631,7 +641,9 @@ oldmodcheck:
 	fi
 
 ld-cache-update:
-ifneq ($(LDCONFIG),)
+ifeq ($(LDCONFIG),)
+else ifeq ($(LDCONFIG),:)
+else
 ifeq ($(DESTDIR),)  # DESTDIR means binary archive creation; ldconfig should be run on postinst
 	@if [ $$(id -u) -eq 0 ] ; then \
 		$(LDCONFIG) "$(ASTLIBDIR)/" ; \
@@ -930,8 +942,20 @@ config:
 		if [ -z "$(DESTDIR)" ] ; then \
 			/sbin/chkconfig --add asterisk ; \
 		fi ; \
+	elif [ -f /etc/os-release ] && [ "opensuse" = "$(shell . /etc/os-release && echo $$ID)" ] ; then \
+		./build_tools/install_subst contrib/init.d/rc.suse.asterisk  "$(DESTDIR)/etc/init.d/asterisk"; \
+		if [ ! -f /etc/sysconfig/asterisk ] ; then \
+			$(INSTALL) -m 644 contrib/init.d/etc_default_asterisk "$(DESTDIR)/etc/sysconfig/asterisk" ; \
+		fi ; \
+		if [ -z "$(DESTDIR)" ] ; then \
+			/sbin/chkconfig --add asterisk ; \
+		fi ; \
 	elif [ -f /etc/arch-release -o -f /etc/arch-release ] ; then \
 		./build_tools/install_subst contrib/init.d/rc.archlinux.asterisk  "$(DESTDIR)/etc/init.d/asterisk"; \
+	elif [ -f /etc/slackware-version ]; then \
+		./build_tools/install_subst contrib/init.d/rc.slackware.asterisk  "$(DESTDIR)/etc/rc.d/rc.asterisk"; \
+	elif [ -f /etc/os-release ] && [ "slackware" = "$(shell . /etc/os-release && echo $$ID)" ] ; then \
+		./build_tools/install_subst contrib/init.d/rc.slackware.asterisk  "$(DESTDIR)/etc/rc.d/rc.asterisk"; \
 	elif [ -d "$(DESTDIR)/Library/LaunchDaemons" ]; then \
 		if [ ! -f "$(DESTDIR)/Library/LaunchDaemons/org.asterisk.asterisk.plist" ]; then \
 			./build_tools/install_subst contrib/init.d/org.asterisk.asterisk.plist "$(DESTDIR)/Library/LaunchDaemons/org.asterisk.asterisk.plist"; \
@@ -939,8 +963,6 @@ config:
 		if [ ! -f "$(DESTDIR)/Library/LaunchDaemons/org.asterisk.muted.plist" ]; then \
 			./build_tools/install_subst contrib/init.d/org.asterisk.muted.plist "$(DESTDIR)/Library/LaunchDaemons/org.asterisk.muted.plist"; \
 		fi; \
-	elif [ -f /etc/slackware-version ]; then \
-		echo "Slackware is not currently supported, although an init script does exist for it."; \
 	else \
 		echo "We could not install init scripts for your distribution." ; \
 	fi
@@ -977,7 +999,9 @@ ifeq ($(HAVE_DAHDI),1)
 	rm -f $(DESTDIR)$(DAHDI_UDEV_HOOK_DIR)/40-asterisk
 endif
 	$(MAKE) -C sounds uninstall
-ifneq ($(LDCONFIG),)
+ifeq ($(LDCONFIG),)
+else ifeq ($(LDCONFIG),:)
+else
 	$(LDCONFIG) "$(ASTLIBDIR)/" || :
 endif
 

UPGRADE.txt

@@ -21,6 +21,14 @@
 === UPGRADE-12.txt  -- Upgrade info for 11 to 12
 ===========================================================
 
+From 13.20.0 to 13.21.0:
+
+app_dial
+------------------
+ * The Dial application now supports early-media video (in addition to
+   audio) on both the calling as well as the called party.
+   Be aware that this is a change in behavior.
+
 From 13.19.0 to 13.20.0:
 
 app_confbridge

addons/Makefile

@@ -65,6 +65,7 @@ ifneq ($(wildcard mp3/Makefile),)
 format_mp3.so: mp3/common.o mp3/dct64_i386.o mp3/decode_ntom.o mp3/layer3.o mp3/tabinit.o mp3/interface.o
 endif
 
+chan_mobile.o: _ASTCFLAGS+=$(AST_NO_FORMAT_TRUNCATION)
 chan_ooh323.o: _ASTCFLAGS+=$(H323CFLAGS)
 chan_ooh323.so: _ASTCFLAGS+=$(H323CFLAGS)
 chan_ooh323.so: $(addprefix ooh323c/src/,$(H323OBJS)) chan_ooh323.o ooh323cDriver.o

addons/cdr_mysql.c

@@ -60,6 +60,14 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
 #define DATE_FORMAT "%Y-%m-%d %T"
 
+#ifndef MYSQL_PORT
+# ifdef MARIADB_PORT
+#  define MYSQL_PORT MARIADB_PORT
+# else
+#  define MYSQL_PORT 3306
+# endif
+#endif
+
 AST_THREADSTORAGE(sql1_buf);
 AST_THREADSTORAGE(sql2_buf);
 AST_THREADSTORAGE(escape_buf);

addons/chan_mobile.c

@@ -58,6 +58,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
 #include "asterisk/compat.h"
 #include "asterisk/lock.h"
+#include "asterisk/callerid.h"
 #include "asterisk/channel.h"
 #include "asterisk/config.h"
 #include "asterisk/logger.h"
@@ -163,6 +164,12 @@ struct mbl_pvt {
 	AST_LIST_ENTRY(mbl_pvt) entry;
 };
 
+/*! Structure used by hfp_parse_clip to return two items */
+struct cidinfo {
+	char *cnum;
+	char *cnam;
+};
+
 static AST_RWLIST_HEAD_STATIC(devices, mbl_pvt);
 
 static int handle_response_ok(struct mbl_pvt *pvt, char *buf);
@@ -208,7 +215,7 @@ static char *mblsendsms_desc =
 "  Dest - destination\n"
 "  Message - text of the message\n";
 
-static struct ast_channel *mbl_new(int state, struct mbl_pvt *pvt, char *cid_num,
+static struct ast_channel *mbl_new(int state, struct mbl_pvt *pvt, struct cidinfo *cidinfo,
 		const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor);
 static struct ast_channel *mbl_request(const char *type, struct ast_format_cap *cap,
 		const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause);
@@ -364,7 +371,8 @@ static struct hfp_hf hfp_our_brsf = {
 
 
 static int hfp_parse_ciev(struct hfp_pvt *hfp, char *buf, int *value);
-static char *hfp_parse_clip(struct hfp_pvt *hfp, char *buf);
+static struct cidinfo hfp_parse_clip(struct hfp_pvt *hfp, char *buf);
+static int parse_next_token(char string[], const int start, const char delim);
 static int hfp_parse_cmti(struct hfp_pvt *hfp, char *buf);
 static int hfp_parse_cmgr(struct hfp_pvt *hfp, char *buf, char **from_number, char **text);
 static int hfp_parse_brsf(struct hfp_pvt *hfp, const char *buf);
@@ -837,7 +845,7 @@ e_return:
 
 */
 
-static struct ast_channel *mbl_new(int state, struct mbl_pvt *pvt, char *cid_num,
+static struct ast_channel *mbl_new(int state, struct mbl_pvt *pvt, struct cidinfo *cidinfo,
 		const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor)
 {
 	struct ast_channel *chn;
@@ -853,8 +861,10 @@ static struct ast_channel *mbl_new(int state, struct mbl_pvt *pvt, char *cid_num
 	ast_smoother_reset(pvt->smoother, DEVICE_FRAME_SIZE);
 	ast_dsp_digitreset(pvt->dsp);
 
-	chn = ast_channel_alloc(1, state, cid_num, pvt->id, 0, 0, pvt->context,
-			assignedids, requestor, 0,
+	chn = ast_channel_alloc(1, state,
+		cidinfo ? cidinfo->cnum : NULL,
+		cidinfo ? cidinfo->cnam : NULL,
+		0, 0, pvt->context, assignedids, requestor, 0,
 		"Mobile/%s-%04lx", pvt->id, ast_random() & 0xffff);
 	if (!chn) {
 		goto e_return;
@@ -2203,45 +2213,103 @@ static int hfp_parse_ciev(struct hfp_pvt *hfp, char *buf, int *value)
  * \param hfp an hfp_pvt struct
  * \param buf the buffer to parse (null terminated)
  * \note buf will be modified when the CID string is parsed
- * \return NULL on error (parse error) or a pointer to the caller id
- * information in buf
+ * \return a cidinfo structure pointing to the cnam and cnum
+ * data in buf.  On parse errors, either or both pointers
+ * will point to null strings
  */
-static char *hfp_parse_clip(struct hfp_pvt *hfp, char *buf)
+static struct cidinfo hfp_parse_clip(struct hfp_pvt *hfp, char *buf)
 {
-	int i, state;
-	char *clip = NULL;
-	size_t s;
+	int i;
+	int tokens[6];
+	char *cnamtmp;
+	char delim = ' ';	/* First token terminates with space */
+	int invalid = 0;	/* Number of invalid chars in cnam */
+	struct cidinfo cidinfo = { NULL, NULL };
 
 	/* parse clip info in the following format:
 	 * +CLIP: "123456789",128,...
 	 */
-	state = 0;
-	s = strlen(buf);
-	for (i = 0; i < s && state != 3; i++) {
-		switch (state) {
-		case 0: /* search for start of the number (") */
-			if (buf[i] == '"') {
-				state++;
+	ast_debug(3, "[%s] hfp_parse_clip is processing \"%s\"\n", hfp->owner->id, buf);
+	tokens[0] = 0;		/* First token starts in position 0 */
+	for (i = 1; i < ARRAY_LEN(tokens); i++) {
+		tokens[i] = parse_next_token(buf, tokens[i - 1], delim);
+		delim = ',';	/* Subsequent tokens terminate with comma */
+	}
+	ast_debug(3, "[%s] hfp_parse_clip found tokens: 0=%s, 1=%s, 2=%s, 3=%s, 4=%s, 5=%s\n",
+		hfp->owner->id, &buf[tokens[0]], &buf[tokens[1]], &buf[tokens[2]],
+		&buf[tokens[3]], &buf[tokens[4]], &buf[tokens[5]]);
+
+	/* Clean up cnum, and make sure it is legitimate since it is untrusted. */
+	cidinfo.cnum = ast_strip_quoted(&buf[tokens[1]], "\"", "\"");
+	if (!ast_isphonenumber(cidinfo.cnum)) {
+		ast_debug(1, "[%s] hfp_parse_clip invalid cidinfo.cnum data \"%s\" - deleting\n",
+			hfp->owner->id, cidinfo.cnum);
+		cidinfo.cnum = "";
+	}
+
+	/*
+	 * Some docs say tokens 2 and 3 including the commas are optional.
+	 * If absent, that would move CNAM back to token 3.
+	 */
+	cidinfo.cnam = &buf[tokens[5]];	/* Assume it's in token 5 */
+	if (buf[tokens[5]] == '\0' && buf[tokens[4]] == '\0') {
+		/* Tokens 4 and 5 are empty.  See if token 3 looks like CNAM (starts with ") */
+		i = tokens[3];
+		while (buf[i] == ' ') {		/* Find the first non-blank */
+			i++;
 		}
-			break;
-		case 1: /* mark the number */
-			clip = &buf[i];
-			state++;
-			/* fall through */
-		case 2: /* search for the end of the number (") */
 		if (buf[i] == '"') {
-				buf[i] = '\0';
-				state++;
+			/* Starts with quote.  Use this for CNAM. */
+			cidinfo.cnam = &buf[i];
 		}
-			break;
+	}
+
+	/* Clean up CNAM. */
+	cidinfo.cnam = ast_strip_quoted(cidinfo.cnam, "\"", "\"");
+	for (cnamtmp = cidinfo.cnam; *cnamtmp != '\0'; cnamtmp++) {
+		if (!strchr("ABCDEFGHIJKLMNOPQRSTUVWXYZ 0123456789-,abcdefghijklmnopqrstuvwxyz_", *cnamtmp)) {
+			*cnamtmp = '_';	/* Invalid.  Replace with underscore. */
+			invalid++;
 		}
 	}
+	if (invalid) {
+		ast_debug(2, "[%s] hfp_parse_clip replaced %d invalid byte(s) in cnam data\n",
+			hfp->owner->id, invalid);
+	}
+	ast_debug(2, "[%s] hfp_parse_clip returns cnum=%s and cnam=%s\n",
+		hfp->owner->id, cidinfo.cnum, cidinfo.cnam);
 
-	if (state != 3) {
-		return NULL;
+	return cidinfo;
 }
 
-	return clip;
+/*!
+ * \brief Terminate current token and return an index to start of the next token.
+ * \param string the null-terminated string being parsed (will be altered!)
+ * \param start where the current token starts
+ * \param delim the token termination delimiter.  \0 is also considered a terminator.
+ * \return index of the next token.  May be the same as this token if the string is
+ * exhausted.
+ */
+static int parse_next_token(char string[], const int start, const char delim)
+{
+	int index;
+	int quoting = 0;
+
+	for (index = start; string[index] != 0; index++) {
+		if ((string[index] == delim) && !quoting ) {
+			/* Found the delimiter, outside of quotes.  This is the end of the token. */
+			string[index] = '\0';	/* Terminate this token. */
+			index++;		/* Point the index to the start of the next token. */
+			break;			/* We're done. */
+		} else if (string[index] == '"' && !quoting) {
+			/* Found a beginning quote mark.  Remember it. */
+			quoting = 1;
+		} else if (string[index] == '"' ) {
+			/* Found the end quote mark. */
+			quoting = 0;
+		}
+	}
+	return index;
 }
 
 /*!
@@ -3578,19 +3646,17 @@ static int handle_response_ciev(struct mbl_pvt *pvt, char *buf)
  */
 static int handle_response_clip(struct mbl_pvt *pvt, char *buf)
 {
-	char *clip;
 	struct msg_queue_entry *msg;
 	struct ast_channel *chan;
+	struct cidinfo cidinfo;
 
 	if ((msg = msg_queue_head(pvt)) && msg->expected == AT_CLIP) {
 		msg_queue_free_and_pop(pvt);
 
 		pvt->needcallerid = 0;
-		if (!(clip = hfp_parse_clip(pvt->hfp, buf))) {
-			ast_debug(1, "[%s] error parsing CLIP: %s\n", pvt->id, buf);
-		}
+		cidinfo = hfp_parse_clip(pvt->hfp, buf);
 
-		if (!(chan = mbl_new(AST_STATE_RING, pvt, clip, NULL, NULL))) {
+		if (!(chan = mbl_new(AST_STATE_RING, pvt, &cidinfo, NULL, NULL))) {
 			ast_log(LOG_ERROR, "[%s] unable to allocate channel for incoming call\n", pvt->id);
 			hfp_send_chup(pvt->hfp);
 			msg_queue_push(pvt, AT_OK, AT_CHUP);
@@ -3859,7 +3925,7 @@ static void *do_monitor_phone(void *data)
 			break;
 		}
 
-		ast_debug(1, "[%s] %s\n", pvt->id, buf);
+		ast_debug(1, "[%s] read %s\n", pvt->id, buf);
 
 		switch (at_msg) {
 		case AT_BRSF:

addons/chan_ooh323.c

@@ -1174,7 +1174,6 @@ static int ooh323_answer(struct ast_channel *ast)
 				p->alertsent = 1;
 			}
 			ast_setstate(ast, AST_STATE_UP);
-      			if (option_debug)
 			ast_debug(1, "ooh323_answer(%s)\n", ast_channel_name(ast));
 			ast_channel_unlock(ast);
 			ooAnswerCall(p->callToken);
@@ -1282,7 +1281,7 @@ static int ooh323_indicate(struct ast_channel *ast, int condition, const void *d
 
 	struct ooh323_pvt *p = (struct ooh323_pvt *) ast_channel_tech_pvt(ast);
 	char *callToken = (char *)NULL;
-	int res = -1;
+	int res = -1, rres;
 
 	if (!p) return -1;
 
@@ -1329,11 +1328,9 @@ static int ooh323_indicate(struct ast_channel *ast, int condition, const void *d
 	case AST_CONTROL_PROGRESS:
 		if (ast_channel_state(ast) != AST_STATE_UP) {
 	    		if (!p->progsent) {
+				rres = ooManualProgress(callToken);
 	     			if (gH323Debug) {
-					ast_debug(1, "Sending manual progress for %s, res = %u\n", callToken,
-             				ooManualProgress(callToken));
-				} else {
-	     				ooManualProgress(callToken);
+					ast_debug(1, "Sending manual progress for %s, res = %u\n", callToken, rres);
 				}
 	     			p->progsent = 1;
 	    		}
@@ -1342,12 +1339,9 @@ static int ooh323_indicate(struct ast_channel *ast, int condition, const void *d
       case AST_CONTROL_RINGING:
 		if (ast_channel_state(ast) == AST_STATE_RING || ast_channel_state(ast) == AST_STATE_RINGING) {
 			if (!p->alertsent) {
+				rres = ooManualRingback(callToken);
 				if (gH323Debug) {
-					ast_debug(1, "Sending manual ringback for %s, res = %u\n",
-						callToken,
-						ooManualRingback(callToken));
-				} else {
-					ooManualRingback(callToken);
+					ast_debug(1, "Sending manual ringback for %s, res = %u\n", callToken, rres);
 				}
 				p->alertsent = 1;
 			}
@@ -5054,9 +5048,7 @@ struct ast_frame *ooh323_rtp_read(struct ast_channel *ast, struct ooh323_pvt *p)
 					ast_log(LOG_NOTICE, "Failed to async goto '%s' into fax of '%s'\n", ast_channel_name(p->owner),target_context);
 				}
 				p->faxdetected = 1;
-				if (dfr) {
 				ast_frfree(dfr);
-				}
 				return &ast_null_frame;
 			}
 		}

addons/ooh323c/src/context.c

@@ -163,6 +163,7 @@ OOCTXT* newContext ()
          /* ASN1CRTFREE0 (pctxt); */
 	 free(pctxt);
          pctxt = 0;
+         return (pctxt);
       }
       pctxt->flags |= ASN1DYNCTXT;
    }

addons/ooh323c/src/memheap.c

@@ -622,7 +622,7 @@ void memHeapFreePtr (void** ppvMemHeap, void* mem_p)
             }
          }
       }
-      if (!ISLAST (pElem) && ISFREE (GETNEXT (pElem))) {
+      if (pElem && !ISLAST (pElem) && ISFREE (GETNEXT (pElem))) {
          OSMemElemDescr* nextelem_p = GETNEXT (pElem);
 
          /* +1 because the OSMemElemDescr has size ONE unit (8 bytes) */
@@ -637,7 +637,7 @@ void memHeapFreePtr (void** ppvMemHeap, void* mem_p)
       }
 
       /* correct the prevOff field of next element */
-      if (!ISLAST (pElem)) {
+      if (pElem && !ISLAST (pElem)) {
          OSMemElemDescr* nextelem_p = GETNEXT (pElem);
          pElem_prevOff (nextelem_p) = QOFFSETOF (nextelem_p, pElem);
       }
@@ -685,7 +685,7 @@ static void initNewFreeElement (OSMemBlk* pMemBlk,
    }
 
    pNextElem = GETNEXT (pNewElem);
-   if (ISFREE (pNextElem)) {
+   if (pNextElem && ISFREE (pNextElem)) {
 
       /* if the next elem is free, then unite them together */
 
@@ -819,7 +819,7 @@ void* memHeapRealloc (void** ppvMemHeap, void* mem_p, int nbytes_)
             /* look for free element after pElem */
 
             pNextElem = GETNEXT (pElem);
-            if (ISFREE (pNextElem)) {
+            if (pNextElem && ISFREE (pNextElem)) {
                /* +1 'cos sizeof (OSMemElemDescr) == 1 unit */
                sumSize += pElem_nunits (pNextElem) + 1;
                freeMem++;
@@ -1061,7 +1061,7 @@ void memHeapAddRef (void** ppvMemHeap)
 void memHeapRelease (void** ppvMemHeap)
 {
    OSMemHeap** ppMemHeap = (OSMemHeap**)ppvMemHeap;
-   OSMemHeap* pMemHeap = *ppMemHeap;
+   OSMemHeap* pMemHeap;
 
    if (ppMemHeap != 0 && *ppMemHeap != 0 && --(*ppMemHeap)->refCnt == 0) {
       OSMemLink* pMemLink, *pMemLink2;
@@ -1079,6 +1079,7 @@ void memHeapRelease (void** ppvMemHeap)
       }
 
       if ((*ppMemHeap)->flags & RT_MH_FREEHEAPDESC) {
+         pMemHeap = *ppMemHeap;
          ast_mutex_destroy(&pMemHeap->pLock);
          free (*ppMemHeap);
       }

addons/ooh323c/src/ooCalls.c

@@ -804,8 +804,7 @@ int ooAddMediaInfo(OOH323CallData *call, OOMediaInfo mediaInfo)
 
    if(!call)
    {
-      OOTRACEERR3("Error:Invalid 'call' param for ooAddMediaInfo.(%s, %s)\n",
-                   call->callType, call->callToken);
+      OOTRACEERR1("Error:Invalid 'call' param for ooAddMediaInfo.\n");
       return OO_FAILED;
    }
    newMediaInfo = (OOMediaInfo*) memAlloc(call->pctxt, sizeof(OOMediaInfo));

addons/ooh323c/src/ooCapability.c

@@ -62,8 +62,6 @@ int ooCapabilityEnableDTMFCISCO
       /*Dynamic RTP payload type range is from 96 - 127 */
       if(dynamicRTPPayloadType >= 96 && dynamicRTPPayloadType <= 127)
         gcDynamicRTPPayloadType = dynamicRTPPayloadType;
-      else
-        call->dtmfcodec = dynamicRTPPayloadType;
    }
    else{
       call->dtmfmode |= OO_CAP_DTMF_CISCO;
@@ -623,8 +621,7 @@ int ooCapabilityAddT38Capability
    else pctxt = call->pctxt;
 
    epCap = (ooH323EpCapability*)memAllocZ(pctxt, sizeof(ooH323EpCapability));
-   params = (OOCapParams*) memAlloc(pctxt, sizeof(OOCapParams));
-   memset(params, 0 , sizeof(OOCapParams));
+   params = (OOCapParams*) memAllocZ(pctxt, sizeof(OOCapParams));
    if(!epCap || !params)
    {
       OOTRACEERR1("ERROR: Memory - ooCapabilityAddT38Capability - "
@@ -808,15 +805,14 @@ void* ooCapabilityCreateDTMFCapability(int cap, int dtmfcodec, OOCTXT *pctxt)
       }
       memset(pATECap, 0, sizeof(H245AudioTelephonyEventCapability));
       pATECap->dynamicRTPPayloadType = dtmfcodec;
-      events = (char*)memAlloc(pctxt, strlen("0-16")+1);
-      memset(events, 0, strlen("0-16")+1);
+      events = (char*)memAllocZ(pctxt, strlen("0-16")+1);
       if(!events)
       {
          OOTRACEERR1("Error:Memory - ooCapabilityCreateDTMFCapability - events\n");
          memFreePtr(pctxt, pATECap);
          return NULL;
       }
-      strncpy(events, "0-16", strlen("0-16"));
+      strcpy(events, "0-16");
       pATECap->audioTelephoneEvent = events;
       return pATECap;
    case OO_CAP_DTMF_CISCO:

addons/ooh323c/src/ooGkClient.c

@@ -2332,9 +2332,8 @@ int ooGkClientSendIRR
    pIRR->m.perCallInfoPresent = TRUE;
 
    perCallInfo =
-    (H225InfoRequestResponse_perCallInfo_element *)memAlloc(pctxt,
+    (H225InfoRequestResponse_perCallInfo_element *)memAllocZ(pctxt,
      sizeof(H225InfoRequestResponse_perCallInfo_element));
-   memset(perCallInfo, 0, sizeof(H225InfoRequestResponse_perCallInfo_element));
 
    if(!perCallInfo)
    {

addons/ooh323c/src/ooh245.c

@@ -356,7 +356,6 @@ int ooSendTermCapMsg(OOH323CallData *call)
    /* pctxt = &gH323ep.msgctxt; */
    pctxt = call->msgctxt;
    ph245msg->msgType = OOTerminalCapabilitySet;
-   memset(request, 0, sizeof(H245RequestMessage));
    if(request == NULL)
    {
       OOTRACEERR3("ERROR: No memory allocated for request message (%s, %s)\n",
@@ -364,6 +363,7 @@ int ooSendTermCapMsg(OOH323CallData *call)
       return OO_FAILED;
    }
 
+   memset(request, 0, sizeof(H245RequestMessage));
    request->t = T_H245RequestMessage_terminalCapabilitySet;
    request->u.terminalCapabilitySet = (H245TerminalCapabilitySet*)
                   memAlloc(pctxt, sizeof(H245TerminalCapabilitySet));

addons/ooh323c/src/ooq931.c

@@ -2439,8 +2439,10 @@ int ooH323HandleCallFwdRequest(OOH323CallData *call)
       alias = call->pCallFwdData->aliases;
       while(alias)
       {
-         pNewAlias = (ooAliases*) memAlloc(pctxt, sizeof(ooAliases));
-         pNewAlias->value = (char*) memAlloc(pctxt, strlen(alias->value)+1);
+         pNewAlias = (ooAliases*) memAllocZ(pctxt, sizeof(ooAliases));
+         if (pNewAlias) {
+            pNewAlias->value = (char*) memAllocZ(pctxt, strlen(alias->value)+1);
+         }
          if(!pNewAlias || !pNewAlias->value)
          {
             OOTRACEERR3("Error:Memory - ooH323HandleCallFwdRequest - "

apps/Makefile

@@ -30,7 +30,11 @@ include $(ASTTOPDIR)/Makefile.moddir_rules
 clean::
 	rm -f confbridge/*.o confbridge/*.i confbridge/*.gcda confbridge/*.gcno
 
+app_confbridge.o: _ASTCFLAGS+=$(AST_NO_FORMAT_TRUNCATION)
+app_meetme.o: _ASTCFLAGS+=$(AST_NO_FORMAT_TRUNCATION)
+app_minivm.o: _ASTCFLAGS+=$(AST_NO_FORMAT_TRUNCATION)
 app_voicemail.o: _ASTCFLAGS+=$(AST_NO_FORMAT_TRUNCATION)
+app_while.o: _ASTCFLAGS+=$(AST_NO_FORMAT_TRUNCATION)
 
 app_confbridge.so: $(subst .c,.o,$(wildcard confbridge/*.c))
 $(subst .c,.o,$(wildcard confbridge/*.c)): _ASTCFLAGS+=$(call MOD_ASTCFLAGS,app_confbridge)

apps/app_agent_pool.c

@@ -440,6 +440,7 @@ static void *agent_cfg_alloc(const char *name)
 	cfg = ao2_alloc_options(sizeof(*cfg), agent_cfg_destructor,
 		AO2_ALLOC_OPT_LOCK_NOLOCK);
 	if (!cfg || ast_string_field_init(cfg, 64)) {
+		ao2_cleanup(cfg);
 		return NULL;
 	}
 	ast_string_field_set(cfg, username, name);

apps/app_amd.c

@@ -279,6 +279,12 @@ static void isAnsweringMachine(struct ast_channel *chan, const char *data)
 
 	/* Now we go into a loop waiting for frames from the channel */
 	while ((res = ast_waitfor(chan, 2 * maxWaitTimeForFrame)) > -1) {
+		int ms = 0;
+
+		/* Figure out how long we waited */
+		if (res > 0) {
+			ms = 2 * maxWaitTimeForFrame - res;
+		}
 
 		/* If we fail to read in a frame, that means they hung up */
 		if (!(f = ast_read(chan))) {
@@ -289,15 +295,22 @@ static void isAnsweringMachine(struct ast_channel *chan, const char *data)
 			break;
 		}
 
-		if (f->frametype == AST_FRAME_VOICE || f->frametype == AST_FRAME_NULL || f->frametype == AST_FRAME_CNG) {
-			/* If the total time exceeds the analysis time then give up as we are not too sure */
+		if (f->frametype == AST_FRAME_VOICE || f->frametype == AST_FRAME_CNG) {
+			/* Figure out how long the frame is in milliseconds */
 			if (f->frametype == AST_FRAME_VOICE) {
 				framelength = (ast_codec_samples_count(f) / DEFAULT_SAMPLES_PER_MS);
 			} else {
-				framelength = 2 * maxWaitTimeForFrame;
+				framelength = ms;
 			}
 
 			iTotalTime += framelength;
+
+			ast_debug(1, "AMD: Channel [%s] frametype [%s] iTotalTime [%d] framelength [%d] totalAnalysisTime [%d]\n",
+					  ast_channel_name(chan),
+					  f->frametype == AST_FRAME_VOICE ? "AST_FRAME_VOICE" : "AST_FRAME_CNG",
+					  iTotalTime, framelength, totalAnalysisTime);
+
+			/* If the total time exceeds the analysis time then give up as we are not too sure */
 			if (iTotalTime >= totalAnalysisTime) {
 				ast_verb(3, "AMD: Channel [%s]. Too long...\n", ast_channel_name(chan));
 				ast_frfree(f);
@@ -308,7 +321,7 @@ static void isAnsweringMachine(struct ast_channel *chan, const char *data)
 
 			/* Feed the frame of audio into the silence detector and see if we get a result */
 			if (f->frametype != AST_FRAME_VOICE)
-				dspsilence += 2 * maxWaitTimeForFrame;
+				dspsilence += framelength;
 			else {
 				dspsilence = 0;
 				ast_dsp_silence(silenceDetector, f, &dspsilence);