Update for 17.1.0

There are reports that this commit causes deadlocks when issuing
a "pri show" CLI command and a channel is in the process of being
hung up.  Further work is in progress to determine the cause of
the deadlock and provide a permanent fix.

ASTERISK~28605  #keeping open
Reported by: Dirk Wendland

This reverts commit c6b17b5212.

Change-Id: Iddd900c1abdd5074ff39c17cdce855f2f436cef9

.gitreview

@@ -1,5 +1,5 @@
 [gerrit]
-defaultbranch=master
+defaultbranch=17
 #
 # Intentional padding to ensure it is possible to point a commit
 # to an alternative gerrit server/repository without breaking

.lastclean

@@ -0,0 +1 @@
+40

.version

@@ -0,0 +1 @@
+17.1.0

CHANGES

@@ -12,6 +12,231 @@
 ===
 ==============================================================================
 
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 17.0.0 to Asterisk 17.1.0 ------------
+------------------------------------------------------------------------------
+
+ARI Channels
+------------------
+ * The Channel resource has a new sub-resource "externalMedia".
+   This allows an application to create a channel for the sole purpose
+   of exchanging media with an external server.  Once created, this
+   channel could be placed into a bridge with existing channels to
+   allow the external server to inject audio into the bridge or
+   receive audio from the bridge.
+   See https://wiki.asterisk.org/wiki/display/AST/External+Media+and+ARI
+   for more information.
+
+Core
+------------------
+ * H.265/HEVC is now a supported video codec and it can be used by
+   specifying "h265" in the allow line.
+   Please note however, that handling of the additional SDP parameters
+   described in RFC 7798 section 7.2 is not yet supported.
+
+app_mixmonitor
+------------------
+ * An option 'S' has been added to MixMonitor. If used in combination with
+   the r() and/or t() options, if a frame is available to write to one of
+   those files but not the other, a frame of silence if written to the file
+   that does not have an audio frame. This should prevent the two files
+   from "drifting" when mixed after the fact.
+
+app_voicemail
+------------------
+ * A feature was added in Asterisk 13.27.0 and 16.4.0 that removed lock files from
+   the Asterisk voicemail directory on startup. Some users that store their
+   voicemails on network storage devices experienced slow startup times due to the
+   relative expense of traversing the voicemail directory structure looking for
+   orphaned lock files. This feature has now been removed.
+
+   Users who require the lock files to be removed at startup should modify their
+   startup scripts to do so before starting the asterisk process.
+
+chan_pjsip
+------------------
+ * A new dialplan function, PJSIP_MOH_PASSTRHOUGH, has been added to chan_pjsip. This
+   allows the behaviour of the moh_passthrough endpoint option to be read or changed
+   in the dialplan. This allows control on a per-call basis.
+
+chan_rtp
+------------------
+ * The UnicastRTP channel driver provided by chan_rtp now accepts
+   "<hostname>:<port>" as an alternative to "<ip_address>:<port>" in the destination.
+   The first AAAA (preferred) or A record resolved will be used as the destination.
+   The lookup is synchronous so beware of possible dialplan delays if you specify a
+   hostname.
+
+func_curl
+------------------
+ * A new parameter, httpheader, has been added to CURLOPT function. This parameter
+   allows to set custom http headers for subsequent calls off CURL function.
+   Any setting of headers will replace the default curl headers
+   (e.g. "Content-type: application/x-www-form-urlencoded")
+
+func_jitterbuffer
+------------------
+ * The JITTERBUFFER dialplan function now has an option to enable video synchronization
+   support. When enabled and used with a compatible channel driver (chan_sip, chan_pjsip)
+   the video is buffered according to the size of the audio jitterbuffer and is
+   synchronized to the audio.
+
+res_musiconhold
+------------------
+ * This fix allows a realtime moh class to be unregistered from the command
+   line. This is useful when the contents of a directory referenced by a
+   realtime moh class have changed.
+   The realtime moh class is then reloaded on the next request and uses the
+   new directory contents.
+
+ * A new mode - playlist - has been added to res_musiconhold. This mode allows the
+   user to specify the files (or URLs) to play explicitly by putting them directly
+   in musiconhold.conf.
+
+taskprocessor.c
+------------------
+ * Added two new CLI commands to reset stats for taskprocessors. You can
+   reset stats for a single, specific taskprocessor ('core reset
+   taskprocessor <taskprocessor>'), or you can reset all taskprocessors
+   ('core reset taskprocessors'). These commands will reset the counter for
+   the number of tasks processed as well as the max queue size.
+
+ * Added "like" support for 'core show taskprocessors'. Now you
+   can specify a specific set of taskprocessors (or just one) by
+   adding the keyword "like" to the above command, followed by
+   your search criteria.
+
+------------------------------------------------------------------------------
+--- New functionality introduced in Asterisk 17.0.0 --------------------------
+------------------------------------------------------------------------------
+
+Bridging
+------------------
+ * The bridging core no longer uses the stasis cache for bridge
+   snapshots.  The latest bridge snapshot is now stored on the
+   ast_bridge structure itself.
+
+   The following APIs are no longer available since the stasis cache
+   is no longer used:
+     ast_bridge_topic_cached()
+     ast_bridge_topic_all_cached()
+
+   A topic pool is now used for individual bridge topics.
+
+   The ast_bridge_cache() function was removed since there's no
+   longer a separate container of snapshots.
+
+   A new function "ast_bridges()" was created to retrieve the
+   container of all bridges.  Users formerly calling
+   ast_bridge_cache() can use the new function to iterate over
+   bridges and retrieve the latest snapshot directly from the
+   bridge.
+
+   The ast_bridge_snapshot_get_latest() function was renamed to
+   ast_bridge_get_snapshot_by_uniqueid().
+
+   A new function "ast_bridge_get_snapshot()" was created to retrieve
+   the bridge snapshot directly from the bridge structure.
+
+   The ast_bridge_topic_all() function now returns a normal topic
+   not a cached one so you can't use stasis cache functions on it
+   either.
+
+   The ast_bridge_snapshot_type() stasis message now has the
+   ast_bridge_snapshot_update structure as it's data.  It contains
+   the last snapshot and the new one.
+
+Channels
+------------------
+ * The core no longer uses the stasis cache for channels snapshots.
+   The following APIs are no longer available:
+       ast_channel_topic_cached()
+       ast_channel_topic_all_cached()
+   The ast_channel_cache_all() and ast_channel_cache_by_name() functions
+   now returns an ao2_container of ast_channel_snapshots rather than a
+   container of stasis_messages therefore you can't call stasis_cache
+   functions on it.
+   The ast_channel_topic_all() function now returns a normal topic,
+   not a cached one so you can't use stasis cache functions on it either.
+   The ast_channel_snapshot_type() stasis message now has the
+   ast_channel_snapshot_update structure as it's data.
+   ast_channel_snapshot_get_latest() still returns the latest snapshot.
+
+chan_sip
+------------------
+ * The chan_sip module is now deprecated, users should migrate to the
+   replacement module chan_pjsip.  See guides at the Asterisk Wiki:
+     https://wiki.asterisk.org/wiki/x/tAHOAQ
+     https://wiki.asterisk.org/wiki/x/hYCLAQ
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 16.0.0 to Asterisk 17.0.0 ------------
+------------------------------------------------------------------------------
+
+AttendedTransfer
+------------------
+ * A new application, this will queue up attended transfer to the given extension.
+
+BlindTransfer
+------------------
+ * A new application, this will redirect all channels currently
+   bridged to the caller channel to the specified destination.
+
+ConfBridge
+------------------
+ * Add "average_all", "highest_all", and "lowest_all" values for
+   the remb_behavior option. These values operate on a bridge
+   level instead of a per-source level. This means that a single
+   REMB value is calculated and sent to every sender, instead of
+   a REMB value that is unique for the specific sender..
+
+Dial
+------------------
+ * Add RINGTIME and RINGTIME_MS variables containing respectively seconds and
+   milliseconds between creation of the dialing channel and receiving the first
+   RINGING signal
+
+   Add PROGRESSTIME and PROGRESSTIME_MS variables analogous to the above with respect to
+   the PROGRESS signal. Shorter of these two times should be equivalent to
+   the PDD (Post Dial Delay) value
+
+   Add DIALEDTIME_MS and ANSWEREDTIME_MS variables to get millisecond resolution
+   versions of DIALEDTIME and ANSWEREDTIME
+
+RTP/ICE
+------------------
+ * You can now indicate that you'd like an ice_host_candidate's local address
+   to be published as well as the mapped address.  See the sample rtp.conf
+   for more information.
+
+ReadExten
+------------------
+ * Add 'p' option to stop reading extension if user presses '#' key.
+
+pbx_dundi
+------------------
+ * The DUNDi PBX module now supports IPv4/IPv6 dual binding.
+
+res_pjsip
+------------------
+ * Added a new PJSIP global setting called norefersub.
+   Default is true to keep support working as before.
+
+   res_pjsip_refer configures PJSIP norefersub capability accordingly.
+
+   Checks the PJSIP global setting value.
+   If it is true (default) it adds the norefersub capability to PJSIP.
+   If it is false (disabled) it does not add the norefersub capability
+   to PJSIP.
+
+   This is useful for Cisco switches that do not follow RFC4488.
+
+res_rtp_asterisk
+------------------
+ * DTLS packets will now be fragmented according to the MTU as set in rtp.conf. This
+   allows larger certificates to be used for the DTLS negotiation. By default this value
+   is 1200.
+
 ------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 16.2.0 to Asterisk 16.3.0 ----------
 ------------------------------------------------------------------------------

README-SERIOUSLY.bestpractices.md

@@ -332,7 +332,7 @@ Data: echo hello world!
 ```
 
 This manager command will attempt to execute an Asterisk application, System,
-which is normally associated with the "system" class authorication.  While some
+which is normally associated with the "system" class authorization.  While some
 checks have been put into Asterisk to take this into account, certain dialplan
 configurations and/or clever manipulation of the Originate manager action can
 circumvent these checks.  For example, take the following dialplan:
@@ -365,7 +365,7 @@ set channel variables; which allows the execution of dialplan functions.
 
 Dialplan functions within Asterisk are incredibly powerful, which is wonderful
 for building applications using Asterisk. But during the read or write
-execution, certain diaplan functions do much more. For example, reading the
+execution, certain dialplan functions do much more. For example, reading the
 SHELL() function can execute arbitrary commands on the system Asterisk is
 running on. Writing to the FILE() function can change any file that Asterisk has
 write access to.

UPGRADE.txt

@@ -18,6 +18,128 @@
 ===
 ===========================================================
 
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 17.0.0 to Asterisk 17.1.0 ------------
+------------------------------------------------------------------------------
+
+AMI
+------------------
+ * The AMI Originate action, which optionally takes a dialplan application as
+   an argument, no longer accepts "Originate" as the application due to
+   security concerns.
+
+------------------------------------------------------------------------------
+--- New functionality introduced in Asterisk 17.0.0 --------------------------
+------------------------------------------------------------------------------
+
+Applications
+------------------
+ * The JabberStatus application, deprecated in Asterisk 12, has been removed.
+
+Bridging
+------------------
+ * The bridging core no longer uses the stasis cache for bridge
+   snapshots.  The latest bridge snapshot is now stored on the
+   ast_bridge structure itself.
+
+   The following APIs are no longer available since the stasis cache
+   is no longer used:
+     ast_bridge_topic_cached()
+     ast_bridge_topic_all_cached()
+
+   A topic pool is now used for individual bridge topics.
+
+   The ast_bridge_cache() function was removed since there's no
+   longer a separate container of snapshots.
+
+   A new function "ast_bridges()" was created to retrieve the
+   container of all bridges.  Users formerly calling
+   ast_bridge_cache() can use the new function to iterate over
+   bridges and retrieve the latest snapshot directly from the
+   bridge.
+
+   The ast_bridge_snapshot_get_latest() function was renamed to
+   ast_bridge_get_snapshot_by_uniqueid().
+
+   A new function "ast_bridge_get_snapshot()" was created to retrieve
+   the bridge snapshot directly from the bridge structure.
+
+   The ast_bridge_topic_all() function now returns a normal topic
+   not a cached one so you can't use stasis cache functions on it
+   either.
+
+   The ast_bridge_snapshot_type() stasis message now has the
+   ast_bridge_snapshot_update structure as it's data.  It contains
+   the last snapshot and the new one.
+
+Build
+------------------
+ * Asterisk headers are no longer installed and uninstalled automatically when
+   performing a "make install" or a "make uninstall".  To install/uninstall the
+   headers, use "make install-headers" and "make uninstall-headers".  The headers
+   also continue to be uninstalled when performing a "make uninstall-all".
+
+Channels
+------------------
+ * The core no longer uses the stasis cache for channels snapshots.
+   The following APIs are no longer available:
+       ast_channel_topic_cached()
+       ast_channel_topic_all_cached()
+   The ast_channel_cache_all() and ast_channel_cache_by_name() functions
+   now returns an ao2_container of ast_channel_snapshots rather than a
+   container of stasis_messages therefore you can't call stasis_cache
+   functions on it.
+   The ast_channel_topic_all() function now returns a normal topic,
+   not a cached one so you can't use stasis cache functions on it either.
+   The ast_channel_snapshot_type() stasis message now has the
+   ast_channel_snapshot_update structure as it's data.
+   ast_channel_snapshot_get_latest() still returns the latest snapshot.
+
+chan_sip
+------------------
+ * The chan_sip module is now deprecated, users should migrate to the
+   replacement module chan_pjsip.  See guides at the Asterisk Wiki:
+     https://wiki.asterisk.org/wiki/x/tAHOAQ
+     https://wiki.asterisk.org/wiki/x/hYCLAQ
+
+func_callerid
+------------------
+ * The CALLERPRES() dialplan function, deprecated in Asterisk 1.8, has been
+   removed.
+
+res_parking
+------------------
+ * The PARKINGSLOT channel variable, deprecated in Asterisk 12 in favor of the
+   PARKING_SPACE channel variable, will no longer be set.
+
+res_xmpp
+------------------
+ * The JabberStatus application, deprecated in Asterisk 12, has been removed.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 16.0.0 to Asterisk 17.0.0 ------------
+------------------------------------------------------------------------------
+
+Core
+------------------
+ * res_pjsip_pubsub is now required so call transfer progress can be monitored
+   and reported in the channel variable TRANSFERSTATUS.
+
+app_voicemail.c
+------------------
+ * The "Voicemail Build Options" section of menuselect has been removed along with
+   the FILE_STORAGE, ODBC_STORAGE and IMAP_STORAGE menuselect options.  All 3 variants
+   of the voicemail app can now be built at the same by enabling app_voicemail,
+   app_voicemail_imap, and app_voicemail_odbc under the "Applications" section.
+   By default, only app_voicemail is enabled.  Also, the modules.conf sample has
+   been updated to "noload" app_voicemail_imap and app_voicemail_odbc should they
+   all be built.  Packagers must update their build scripts appropriately.
+
+chan_pjsip
+------------------
+ * res_pjsip_pubsub is now required so call transfer progress can be monitored
+   and reported in the channel variable TRANSFERSTATUS.
+
 New in 16.0.0:
 
 app_fax:

addons/cdr_mysql.c

@@ -173,7 +173,9 @@ static int mysql_log(struct ast_cdr *cdr)
 {
 	struct ast_str *sql1 = ast_str_thread_get(&sql1_buf, 1024), *sql2 = ast_str_thread_get(&sql2_buf, 1024);
 	int retries = 5;
-#if MYSQL_VERSION_ID >= 50013
+#ifdef HAVE_MYSQLCLIENT_BOOL
+	bool my_bool_true = 1;
+#elif HAVE_MYSQLCLIENT_MY_BOOL
 	my_bool my_bool_true = 1;
 #endif
 
@@ -382,6 +384,13 @@ static int my_unload_module(int reload)
 {
 	struct column *entry;
 
+	if (!reload) {
+		if (ast_cdr_unregister(name)) {
+			/* If we can't unregister the backend, we can't unload the module */
+			return -1;
+		}
+	}
+
 	ast_cli_unregister_multiple(cdr_mysql_status_cli, sizeof(cdr_mysql_status_cli) / sizeof(struct ast_cli_entry));
 
 	if (connected) {
@@ -406,7 +415,8 @@ static int my_unload_module(int reload)
 	if (reload) {
 		return ast_cdr_backend_suspend(name);
 	} else {
-		return ast_cdr_unregister(name);
+		/* We unregistered earlier */
+		return 0;
 	}
 }
 
@@ -460,7 +470,9 @@ static int my_connect_db(struct ast_config *cfg)
 	MYSQL_ROW row;
 	MYSQL_RES *result;
 	char sqldesc[128];
-#if MYSQL_VERSION_ID >= 50013
+#ifdef HAVE_MYSQLCLIENT_BOOL
+	bool my_bool_true = 1;
+#elif HAVE_MYSQLCLIENT_MY_BOOL
 	my_bool my_bool_true = 1;
 #endif
 

addons/chan_ooh323.c

@@ -536,11 +536,6 @@ static struct ast_channel *ooh323_new(struct ooh323_pvt *i, int state,
 				ch = NULL;
 			}
 	 	}
-
-		if (ch) {
-			ast_publish_channel_state(ch);
-
-		}
 	} else {
 		ao2_cleanup(caps);
 		ast_log(LOG_WARNING, "Unable to allocate channel structure\n");
@@ -2225,7 +2220,7 @@ int onCallEstablished(ooCallData *call)
 			}
 
 			ast_queue_control(c, AST_CONTROL_ANSWER);
-			ast_publish_channel_state(c);
+			ast_channel_publish_snapshot(c);
 			ast_channel_unlock(p->owner);
 		}
 		ast_mutex_unlock(&p->lock);

apps/app_agent_pool.c

@@ -1457,9 +1457,9 @@ static void send_agent_logoff(struct ast_channel *chan, const char *agent, long
 
 	ast_assert(agent != NULL);
 
-	blob = ast_json_pack("{s: s, s: i}",
+	blob = ast_json_pack("{s: s, s: I}",
 		"agent", agent,
-		"logintime", logintime);
+		"logintime", (ast_json_int_t)logintime);
 	if (!blob) {
 		return;
 	}

apps/app_amd.c

@@ -162,8 +162,10 @@ static int dfltMaxWaitTimeForFrame  = 50;
 static void isAnsweringMachine(struct ast_channel *chan, const char *data)
 {
 	int res = 0;
+	int audioFrameCount = 0;
 	struct ast_frame *f = NULL;
 	struct ast_dsp *silenceDetector = NULL;
+	struct timeval amd_tvstart;
 	int dspsilence = 0, framelength = 0;
 	RAII_VAR(struct ast_format *, readFormat, NULL, ao2_cleanup);
 	int inInitialSilence = 1;
@@ -275,6 +277,9 @@ static void isAnsweringMachine(struct ast_channel *chan, const char *data)
 	/* Set silence threshold to specified value */
 	ast_dsp_set_threshold(silenceDetector, silenceThreshold);
 
+	/* Set our start time so we can tie the loop to real world time and not RTP updates */
+	amd_tvstart = ast_tvnow();
+
 	/* Now we go into a loop waiting for frames from the channel */
 	while ((res = ast_waitfor(chan, 2 * maxWaitTimeForFrame)) > -1) {
 		int ms = 0;
@@ -293,7 +298,24 @@ 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) {
+		/* Check to make sure we haven't gone over our real-world timeout in case frames get stalled for whatever reason */
+		if ( (ast_tvdiff_ms(ast_tvnow(), amd_tvstart)) > totalAnalysisTime ) {
+			ast_frfree(f);
+			strcpy(amdStatus , "NOTSURE");
+			if ( audioFrameCount == 0 ) {
+				ast_verb(3, "AMD: Channel [%s]. No audio data received in [%d] seconds.\n", ast_channel_name(chan), totalAnalysisTime);
+				sprintf(amdCause , "NOAUDIODATA-%d", iTotalTime);
+				break;
+			}
+			ast_verb(3, "AMD: Channel [%s]. Timeout...\n", ast_channel_name(chan));
+			sprintf(amdCause , "TOOLONG-%d", iTotalTime);
+			break;
+		}
+
+		if (f->frametype == AST_FRAME_VOICE || f->frametype == AST_FRAME_CNG) {
+			/* keep track of the number of audio frames we get */
+			audioFrameCount++;
+
 			/* 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);

apps/app_cdr.c

@@ -119,13 +119,13 @@ struct app_cdr_message_payload {
 	/*! The name of the channel to be manipulated */
 	const char *channel_name;
 	/*! Disable the CDR for this channel */
-	int disable:1;
+	unsigned int disable:1;
 	/*! Re-enable the CDR for this channel */
-	int reenable:1;
+	unsigned int reenable:1;
 	/*! Reset the CDR */
-	int reset:1;
+	unsigned int reset:1;
 	/*! If reseting the CDR, keep the variables */
-	int keep_variables:1;
+	unsigned int keep_variables:1;
 };
 
 static void appcdr_callback(void *data, struct stasis_subscription *sub, struct stasis_message *message)

apps/app_chanisavail.c

@@ -178,6 +178,9 @@ static int chanavail_exec(struct ast_channel *chan, const char *data)
 					snprintf(tmp, sizeof(tmp), "%d", status);
 					ast_str_append(&tmp_availcause, 0, "%s%s", ast_str_strlen(tmp_availcause) ? "&" : "", tmp);
 
+					/* Disable CDR for this temporary channel. */
+					ast_cdr_set_property(ast_channel_name(tempchan), AST_CDR_FLAG_DISABLE_ALL);
+
 					ast_hangup(tempchan);
 					tempchan = NULL;
 

apps/app_dictate.c

@@ -149,7 +149,8 @@ static int dictate_exec(struct ast_channel *chan, const char *data)
 		ast_mkdir(base, 0755);
 		len = strlen(base) + strlen(filein) + 2;
 		if (!path || len > maxlen) {
-			path = ast_alloca(len);
+			ast_free(path);
+			path = ast_malloc(len);
 			memset(path, 0, len);
 			maxlen = len;
 		} else {
@@ -334,6 +335,7 @@ static int dictate_exec(struct ast_channel *chan, const char *data)
 			ast_frfree(f);
 		}
 	}
+	ast_free(path);
 	if (oldr) {
 		ast_set_read_format(chan, oldr);
 		ao2_ref(oldr, -1);

apps/app_followme.c

@@ -408,7 +408,6 @@ static int reload_followme(int reload)
 	char *cat = NULL, *tmp;
 	struct ast_variable *var;
 	struct number *cur, *nm;
-	char *numberstr;
 	int timeout;
 	int numorder;
 	const char* enable_callee_prompt_str;
@@ -536,9 +535,12 @@ static int reload_followme(int reload)
 		while (var) {
 			if (!strcasecmp(var->name, "number")) {
 				int idx = 0;
+				char copy[strlen(var->value) + 1];
+				char *numberstr;
 
 				/* Add a new number */
-				numberstr = ast_strdupa(var->value);
+				strcpy(copy, var->value); /* safe */
+				numberstr = copy;
 				if ((tmp = strchr(numberstr, ','))) {
 					*tmp++ = '\0';
 					timeout = atoi(tmp);
@@ -762,10 +764,6 @@ static struct ast_channel *wait_for_winner(struct findme_user_listptr *findme_us
 		}
 
 		tmpto = to;
-		if (to < 0) {
-			to = 1000;
-			tmpto = 1000;
-		}
 		towas = to;
 		winner = ast_waitfor_n(watchers, pos, &to);
 		tmpto -= to;

apps/app_meetme.c

@@ -1165,7 +1165,7 @@ static int meetme_stasis_init(void)
 	STASIS_MESSAGE_TYPE_INIT(meetme_talk_request_type);
 
 	meetme_event_message_router = stasis_message_router_create(
-		ast_channel_topic_all_cached());
+		ast_channel_topic_all());
 
 	if (!meetme_event_message_router) {
 		meetme_stasis_cleanup();

apps/app_minivm.c

@@ -2652,9 +2652,10 @@ static int create_vmaccount(char *name, struct ast_variable *var, int realtime)
 			ast_copy_string(vmu->fullname, var->value, sizeof(vmu->fullname));
 		} else if (!strcasecmp(var->name, "setvar")) {
 			char *varval;
-			char *varname = ast_strdupa(var->value);
+			char varname[strlen(var->value) + 1];
 			struct ast_variable *tmpvar;
 
+			strcpy(varname, var->value); /* safe */
 			if ((varval = strchr(varname, '='))) {
 				*varval = '\0';
 				varval++;

apps/app_mixmonitor.c

@@ -115,6 +115,11 @@
 						Like with the basic filename argument, if an absolute path isn't given, it will create
 						the file in the configured monitoring directory.</para>
 					</option>
+					<option name="S">
+						<para>When combined with the <replaceable>r</replaceable> or <replaceable>t</replaceable>
+						option, inserts silence when necessary to maintain synchronization between the receive
+						and transmit audio streams.</para>
+					</option>
 					<option name="i">
 						<argument name="chanvar" required="true" />
 						<para>Stores the MixMonitor's ID on this channel variable.</para>
@@ -347,7 +352,8 @@ enum mixmonitor_flags {
 	MUXFLAG_VMRECIPIENTS = (1 << 10),
 	MUXFLAG_BEEP = (1 << 11),
 	MUXFLAG_BEEP_START = (1 << 12),
-	MUXFLAG_BEEP_STOP = (1 << 13)
+	MUXFLAG_BEEP_STOP = (1 << 13),
+	MUXFLAG_RWSYNC = (1 << 14),
 };
 
 enum mixmonitor_args {
@@ -359,6 +365,7 @@ enum mixmonitor_args {
 	OPT_ARG_UID,
 	OPT_ARG_VMRECIPIENTS,
 	OPT_ARG_BEEP_INTERVAL,
+	OPT_ARG_RWSYNC,
 	OPT_ARG_ARRAY_SIZE,	/* Always last element of the enum */
 };
 
@@ -375,6 +382,7 @@ AST_APP_OPTIONS(mixmonitor_opts, {
 	AST_APP_OPTION_ARG('t', MUXFLAG_WRITE, OPT_ARG_WRITENAME),
 	AST_APP_OPTION_ARG('i', MUXFLAG_UID, OPT_ARG_UID),
 	AST_APP_OPTION_ARG('m', MUXFLAG_VMRECIPIENTS, OPT_ARG_VMRECIPIENTS),
+	AST_APP_OPTION_ARG('S', MUXFLAG_RWSYNC, OPT_ARG_RWSYNC),
 });
 
 struct mixmonitor_ds {
@@ -962,6 +970,9 @@ static int launch_monitor_thread(struct ast_channel *chan, const char *filename,
 	}
 
 	ast_set_flag(&mixmonitor->audiohook, AST_AUDIOHOOK_TRIGGER_SYNC);
+	if ((ast_test_flag(mixmonitor, MUXFLAG_RWSYNC))) {
+		ast_set_flag(&mixmonitor->audiohook, AST_AUDIOHOOK_SUBSTITUTE_SILENCE);
+	}
 
 	if (readvol)
 		mixmonitor->audiohook.options.read_volume = readvol;

apps/app_playback.c

@@ -173,7 +173,10 @@ static int s_streamwait3(const say_args_t *a, const char *fn)
 static int do_say(say_args_t *a, const char *s, const char *options, int depth)
 {
 	struct ast_variable *v;
-	char *lang, *x, *rule = NULL;
+	char *lang;
+	char *x;
+	char *rule = NULL;
+	char *rule_head = NULL;
 	int ret = 0;
 	struct varshead head = { .first = NULL, .last = NULL };
 	struct ast_var_t *n;
@@ -195,7 +198,7 @@ static int do_say(say_args_t *a, const char *s, const char *options, int depth)
 	for (;;) {
 		for (v = ast_variable_browse(say_cfg, lang); v ; v = v->next) {
 			if (ast_extension_match(v->name, s)) {
-				rule = ast_strdupa(v->value);
+				rule_head = rule = ast_strdup(v->value);
 				break;
 			}
 		}
@@ -220,6 +223,7 @@ static int do_say(say_args_t *a, const char *s, const char *options, int depth)
 	n = ast_var_assign("SAY", s);
 	if (!n) {
 		ast_log(LOG_ERROR, "Memory allocation error in do_say\n");
+		ast_free(rule_head);
 		return -1;
 	}
 	AST_LIST_INSERT_HEAD(&head, n, entries);
@@ -281,6 +285,7 @@ static int do_say(say_args_t *a, const char *s, const char *options, int depth)
 		}
 	}
 	ast_var_delete(n);
+	ast_free(rule_head);
 	return ret;
 }
 

apps/app_queue.c

@@ -4904,7 +4904,7 @@ static void rna(int rnatime, struct queue_ent *qe, struct ast_channel *peer, cha
 				ast_verb(3, "Auto-Pausing Queue Member %s in all queues since they failed to answer on queue %s.\n",
 						interface, qe->parent->name);
 			} else {
-					ast_verb(3, "Failed to pause Queue Member %s in all queues!\n", interface);
+				ast_verb(3, "Failed to pause Queue Member %s in all queues!\n", interface);
 			}
 		}
 	}
@@ -5898,12 +5898,12 @@ static void send_agent_complete(const char *queuename, struct ast_channel_snapsh
 		break;
 	}
 
-	blob = ast_json_pack("{s: s, s: s, s: s, s: i, s: i, s: s}",
+	blob = ast_json_pack("{s: s, s: s, s: s, s: I, s: I, s: s}",
 		"Queue", queuename,
 		"Interface", member->interface,
 		"MemberName", member->membername,
-		"HoldTime", (long)(callstart - holdstart),
-		"TalkTime", (long)(time(NULL) - callstart),
+		"HoldTime", (ast_json_int_t)(callstart - holdstart),
+		"TalkTime", (ast_json_int_t)(time(NULL) - callstart),
 		"Reason", reason ?: "");
 
 	queue_publish_multi_channel_snapshot_blob(ast_queue_topic(queuename), caller, peer,
@@ -6850,8 +6850,7 @@ static int try_calling(struct queue_ent *qe, struct ast_flags opts, char **opt_a
 		tmp->lastcall = cur->lastcall;
 		tmp->lastqueue = cur->lastqueue;
 		ast_copy_string(tmp->interface, cur->interface, sizeof(tmp->interface));
-		/* Special case: If we ring everyone, go ahead and ring them, otherwise
-		   just calculate their metric for the appropriate strategy */
+		/* Calculate the metric for the appropriate strategy. */
 		if (!calc_metric(qe->parent, cur, x++, qe, tmp)) {
 			/* Put them in the list of outgoing thingies...  We're ready now.
 			   XXX If we're forcibly removed, these outgoing calls won't get
@@ -6887,6 +6886,7 @@ static int try_calling(struct queue_ent *qe, struct ast_flags opts, char **opt_a
 	orig = to;
 	++qe->pending;
 	ao2_unlock(qe->parent);
+	/* Call the queue members with the best metric now. */
 	ring_one(qe, outgoing, &numbusies);
 	lpeer = wait_for_answer(qe, outgoing, &to, &digit, numbusies,
 		ast_test_flag(&(bridge_config.features_caller), AST_FEATURE_DISCONNECT),
@@ -7174,12 +7174,12 @@ static int try_calling(struct queue_ent *qe, struct ast_flags opts, char **opt_a
 		ast_queue_log(queuename, ast_channel_uniqueid(qe->chan), member->membername, "CONNECT", "%ld|%s|%ld", (long) (time(NULL) - qe->start), ast_channel_uniqueid(peer),
 													(long)(orig - to > 0 ? (orig - to) / 1000 : 0));
 
-		blob = ast_json_pack("{s: s, s: s, s: s, s: i, s: i}",
+		blob = ast_json_pack("{s: s, s: s, s: s, s: I, s: I}",
 				     "Queue", queuename,
 				     "Interface", member->interface,
 				     "MemberName", member->membername,
-				     "HoldTime", (long) (time(NULL) - qe->start),
-				     "RingTime", (long)(orig - to > 0 ? (orig - to) / 1000 : 0));
+				     "HoldTime", (ast_json_int_t)(time(NULL) - qe->start),
+				     "RingTime", (ast_json_int_t)(orig - to > 0 ? (orig - to) / 1000 : 0));
 		queue_publish_multi_channel_blob(qe->chan, peer, queue_agent_connect_type(), blob);
 
 		ast_copy_string(oldcontext, ast_channel_context(qe->chan), sizeof(oldcontext));

apps/app_readexten.c

@@ -170,8 +170,7 @@ static int readexten_exec(struct ast_channel *chan, const char *data)
 	if (timeout <= 0)
 		timeout = ast_channel_pbx(chan) ? ast_channel_pbx(chan)->rtimeoutms : 10000;
 
-	if (digit_timeout <= 0)
-		digit_timeout = ast_channel_pbx(chan) ? ast_channel_pbx(chan)->dtimeoutms : 5000;
+	digit_timeout = ast_channel_pbx(chan) ? ast_channel_pbx(chan)->dtimeoutms : 5000;
 
 	if (ast_test_flag(&flags, OPT_INDICATION) && !ast_strlen_zero(arglist.filename)) {
 		ts = ast_get_indication_tone(ast_channel_zone(chan), arglist.filename);

apps/app_senddtmf.c

@@ -80,6 +80,9 @@
 			<parameter name="Duration" required="false">
 				<para>The duration, in milliseconds, of the digit to be played.</para>
 			</parameter>
+			<parameter name="Receive" required="false">
+				<para>Emulate receiving DTMF on this channel instead of sending it out.</para>
+			</parameter>
 		</syntax>
 		<description>
 			<para>Plays a dtmf digit on the specified channel.</para>
@@ -147,6 +150,7 @@ static int manager_play_dtmf(struct mansession *s, const struct message *m)
 	const char *channel = astman_get_header(m, "Channel");
 	const char *digit = astman_get_header(m, "Digit");
 	const char *duration = astman_get_header(m, "Duration");
+	const char *receive_s = astman_get_header(m, "Receive");
 	struct ast_channel *chan;
 	unsigned int duration_ms = 0;
 
@@ -167,7 +171,14 @@ static int manager_play_dtmf(struct mansession *s, const struct message *m)
 		return 0;
 	}
 
-	ast_senddigit_external(chan, *digit, duration_ms);
+	if (ast_true(receive_s)) {
+		struct ast_frame f = { AST_FRAME_DTMF, };
+		f.len = duration_ms;
+		f.subclass.integer = *digit;
+		ast_queue_frame(chan, &f);
+	} else {
+		ast_senddigit_external(chan, *digit, duration_ms);
+	}
 
 	chan = ast_channel_unref(chan);
 

apps/app_voicemail.c

@@ -46,39 +46,6 @@
  * \verbinclude voicemail.conf.sample
  */
 
-/*** MODULEINFO
-	<defaultenabled>yes</defaultenabled>
-	<use type="module">res_adsi</use>
-	<use type="module">res_smdi</use>
-	<support_level>core</support_level>
- ***/
-
-/*** MAKEOPTS
-<category name="MENUSELECT_OPTS_app_voicemail" displayname="Voicemail Build Options" positive_output="yes" touch_on_change="apps/app_voicemail.c apps/app_directory.c">
-	<member name="FILE_STORAGE" displayname="Storage of Voicemail using filesystem">
-		<conflict>ODBC_STORAGE</conflict>
-		<conflict>IMAP_STORAGE</conflict>
-		<defaultenabled>yes</defaultenabled>
-		<support_level>core</support_level>
-	</member>
-	<member name="ODBC_STORAGE" displayname="Storage of Voicemail using ODBC">
-		<depend>generic_odbc</depend>
-		<conflict>IMAP_STORAGE</conflict>
-		<conflict>FILE_STORAGE</conflict>
-		<defaultenabled>no</defaultenabled>
-		<support_level>core</support_level>
-	</member>
-	<member name="IMAP_STORAGE" displayname="Storage of Voicemail using IMAP4">
-		<depend>imap_tk</depend>
-		<conflict>ODBC_STORAGE</conflict>
-		<conflict>FILE_STORAGE</conflict>
-		<use type="external">openssl</use>
-		<defaultenabled>no</defaultenabled>
-		<support_level>core</support_level>
-	</member>
-</category>
-***/
-
 #include "asterisk.h"
 
 #ifdef IMAP_STORAGE
@@ -1830,7 +1797,7 @@ static void vm_change_password(struct ast_vm_user *vmu, const char *newpassword)
 	struct ast_config   *cfg = NULL;
 	struct ast_variable *var = NULL;
 	struct ast_category *cat = NULL;
-	char *category = NULL, *value = NULL, *new = NULL;
+	char *category = NULL;
 	const char *tmp = NULL;
 	struct ast_flags config_flags = { CONFIG_FLAG_WITHCOMMENTS };
 	char secretfn[PATH_MAX] = "";
@@ -1857,24 +1824,28 @@ static void vm_change_password(struct ast_vm_user *vmu, const char *newpassword)
 		if ((cfg = ast_config_load(VOICEMAIL_CONFIG, config_flags)) && valid_config(cfg)) {
 			while ((category = ast_category_browse(cfg, category))) {
 				if (!strcasecmp(category, vmu->context)) {
+					char *value = NULL;
+					char *new = NULL;
 					if (!(tmp = ast_variable_retrieve(cfg, category, vmu->mailbox))) {
 						ast_log(AST_LOG_WARNING, "We could not find the mailbox.\n");
 						break;
 					}
 					value = strstr(tmp, ",");
 					if (!value) {
-						new = ast_alloca(strlen(newpassword)+1);
+						new = ast_malloc(strlen(newpassword) + 1);
 						sprintf(new, "%s", newpassword);
 					} else {
-						new = ast_alloca((strlen(value) + strlen(newpassword) + 1));
+						new = ast_malloc((strlen(value) + strlen(newpassword) + 1));
 						sprintf(new, "%s%s", newpassword, value);
 					}
 					if (!(cat = ast_category_get(cfg, category, NULL))) {
 						ast_log(AST_LOG_WARNING, "Failed to get category structure.\n");
+						ast_free(new);
 						break;
 					}
 					ast_variable_update(cat, vmu->mailbox, new, NULL, 0);
 					found = 1;
+					ast_free(new);
 				}
 			}
 			/* save the results */
@@ -1898,13 +1869,14 @@ static void vm_change_password(struct ast_vm_user *vmu, const char *newpassword)
 			for (category = ast_category_browse(cfg, NULL); category; category = ast_category_browse(cfg, category)) {
 				ast_debug(4, "users.conf: %s\n", category);
 				if (!strcasecmp(category, vmu->mailbox)) {
+					char new[strlen(newpassword) + 1];
 					if (!ast_variable_retrieve(cfg, category, "vmsecret")) {
 						ast_debug(3, "looks like we need to make vmsecret!\n");
 						var = ast_variable_new("vmsecret", newpassword, "");
 					} else {
 						var = NULL;
 					}
-					new = ast_alloca(strlen(newpassword) + 1);
+
 					sprintf(new, "%s", newpassword);
 					if (!(cat = ast_category_get(cfg, category, NULL))) {
 						ast_debug(4, "failed to get category!\n");
@@ -2227,7 +2199,6 @@ static int imap_retrieve_greeting(const char *dir, const int msgnum, struct ast_
 	struct vm_state *vms_p;
 	char *file, *filename;
 	char dest[PATH_MAX];
-	char *attachment;
 	int i;
 	BODY *body;
 	int ret = 0;
@@ -2280,21 +2251,26 @@ static int imap_retrieve_greeting(const char *dir, const int msgnum, struct ast_
 		mail_fetchstructure(vms_p->mailstream, i + 1, &body);
 		/* We have the body, now we extract the file name of the first attachment. */
 		if (body->nested.part && body->nested.part->next && body->nested.part->next->body.parameter->value) {
-			attachment = ast_strdupa(body->nested.part->next->body.parameter->value);
+			char *attachment = body->nested.part->next->body.parameter->value;
+			char copy[strlen(attachment) + 1];
+
+			strcpy(copy, attachment); /* safe */
+			attachment = copy;
+
+			filename = strsep(&attachment, ".");
+			if (!strcmp(filename, file)) {
+				ast_copy_string(vms_p->fn, dir, sizeof(vms_p->fn));
+				vms_p->msgArray[vms_p->curmsg] = i + 1;
+				create_dirpath(dest, sizeof(dest), vmu->context, vms_p->username, "");
+				save_body(body, vms_p, "2", attachment, 0);
+				ret = 0;
+				break;
+			}
 		} else {
 			ast_log(AST_LOG_ERROR, "There is no file attached to this IMAP message.\n");
 			ret = -1;
 			break;
 		}
-		filename = strsep(&attachment, ".");
-		if (!strcmp(filename, file)) {
-			ast_copy_string(vms_p->fn, dir, sizeof(vms_p->fn));
-			vms_p->msgArray[vms_p->curmsg] = i + 1;
-			create_dirpath(dest, sizeof(dest), vmu->context, vms_p->username, "");
-			save_body(body, vms_p, "2", attachment, 0);
-			ret = 0;
-			break;
-		}
 	}
 
 	if (curr_mbox != -1) {
@@ -3778,54 +3754,6 @@ static void check_quota(struct vm_state *vms, char *mailbox) {
 
 #endif /* IMAP_STORAGE */
 
-static void cleanup_orphaned_lock_files(const char *base)
-{
-	DIR *dir;
-	struct dirent *e;
-
-	dir = opendir(base);
-	if (!dir) {
-		/* Don't complain about this too loudly */
-		ast_debug(2, "Unable to open `%s': %s\n", base, strerror(errno));
-		return;
-	}
-
-	while ((e = readdir(dir))) {
-		char *fullpath;
-		struct stat s;
-
-		/* Always skip . and .. */
-		if (!strcmp(e->d_name, ".") || !strcmp(e->d_name, "..")) {
-			continue;
-		}
-
-		/* Build up the full path (using dynamic memory because PATH_MAX is crap) */
-		if (ast_asprintf(&fullpath, "%s/%s", base, e->d_name) == -1) {
-			break;
-		}
-
-		if (lstat(fullpath, &s) < 0) {
-			ast_free(fullpath);
-			continue;
-		}
-
-		/* This is exposing an implementation detail of ast_lock_path, but it makes
-		 * our life a bit easier */
-		if (!strcmp(e->d_name, ".lock") && S_ISLNK(s.st_mode)) {
-			if (!ast_unlock_path(base)) {
-				ast_log(AST_LOG_NOTICE, "Cleaned up orphaned lock file: %s/.lock\n", base);
-			}
-		} else if (S_ISDIR(s.st_mode)) {
-			/* If it is a directory, let's dive down */
-			cleanup_orphaned_lock_files(fullpath);
-		}
-
-		ast_free(fullpath);
-	}
-
-	closedir(dir);
-}
-
 /*! \brief Lock file path
  * only return failure if ast_lock_path returns 'timeout',
  * not if the path does not exist or any other reason
@@ -5479,12 +5407,33 @@ static int add_email_attachment(FILE *p, struct ast_vm_user *vmu, char *format,
 	char sox_gain_tmpdir[PATH_MAX];
 	char *file_to_delete = NULL, *dir_to_delete = NULL;
 	int res;
+	char altfname[PATH_MAX] = "";
+	int altused = 0;
+	char altformat[80] = "";
+	char *c = NULL;
 
 	/* Eww. We want formats to tell us their own MIME type */
 	char *mime_type = (!strcasecmp(format, "ogg")) ? "application/" : "audio/x-";
 
+	/* Users of multiple file formats need special attention. */
+	snprintf(fname, sizeof(fname), "%s.%s", attach, format);
+	if (!ast_file_is_readable(fname)) {
+		ast_copy_string(altformat, vmfmts, sizeof(altformat));
+		c = strchr(altformat, '|');
+		if (c) {
+			*c = '\0';
+		}
+		ast_log(AST_LOG_WARNING, "Failed to open file: %s: %s - trying first/alternate format %s\n", fname, strerror(errno), altformat);
+		snprintf(altfname, sizeof(altfname), "%s.%s", attach, altformat);
+		if (!ast_file_is_readable(altfname)) {
+			ast_log(AST_LOG_WARNING, "Failed to open file: %s: %s - alternate format %s failure\n", altfname, strerror(errno), altformat);
+		} else {
+			altused = 1;
+		}
+	}
+
 	/* This 'while' loop will only execute once. We use it so that we can 'break' */
-	while (vmu->volgain < -.001 || vmu->volgain > .001) {
+	while (vmu->volgain < -.001 || vmu->volgain > .001 || altused) {
 		char tmpdir[PATH_MAX];
 
 		create_dirpath(tmpdir, sizeof(tmpdir), vmu->context, vmu->mailbox, "tmp");
@@ -5510,8 +5459,29 @@ static int add_email_attachment(FILE *p, struct ast_vm_user *vmu, char *format,
 				break;
 			}
 
-			res = snprintf(sox_gain_cmd, sizeof(sox_gain_cmd), "sox -v %.4f %s.%s %s",
-						   vmu->volgain, attach, format, fname);
+			if (!altused) {
+				res = snprintf(sox_gain_cmd, sizeof(sox_gain_cmd), "sox -v %.4f %s.%s %s",
+							   vmu->volgain, attach, format, fname);
+			} else {
+				if (!strcasecmp(format, "wav")) {
+					if (vmu->volgain < -.001 || vmu->volgain > .001) {
+						res = snprintf(sox_gain_cmd, sizeof(sox_gain_cmd), "sox -v %.4f %s.%s -e signed-integer -b 16 %s",
+									   vmu->volgain, attach, altformat, fname);
+					} else {
+						res = snprintf(sox_gain_cmd, sizeof(sox_gain_cmd), "sox %s.%s -e signed-integer -b 16 %s",
+									   attach, altformat, fname);
+					}
+				} else {
+					if (vmu->volgain < -.001 || vmu->volgain > .001) {
+						res = snprintf(sox_gain_cmd, sizeof(sox_gain_cmd), "sox -v %.4f %s.%s %s",
+									   vmu->volgain, attach, altformat, fname);
+					} else {
+						res = snprintf(sox_gain_cmd, sizeof(sox_gain_cmd), "sox %s.%s %s",
+									   attach, altformat, fname);
+					}
+				}
+			}
+
 			if (res >= sizeof(sox_gain_cmd)) {
 				ast_log(LOG_ERROR, "Failed to generate sox command, out of buffer space\n");
 				break;
@@ -7212,7 +7182,7 @@ leave_vm_out:
 #ifdef IMAP_STORAGE
 	/* expunge message - use UID Expunge if supported on IMAP server*/
 	ast_debug(3, "*** Checking if we can expunge, expungeonhangup set to %d\n", expungeonhangup);
-	if (expungeonhangup == 1) {
+	if (expungeonhangup == 1 && vms->mailstream != NULL) {
 		ast_mutex_lock(&vms->lock);
 #ifdef HAVE_IMAP_TK2006
 		if (LEVELUIDPLUS (vms->mailstream)) {
@@ -8180,10 +8150,12 @@ static void queue_mwi_event(const char *channel_id, const char *box, int urgent,
 
 		aliases = ao2_find(mailbox_alias_mappings, box, OBJ_SEARCH_KEY | OBJ_MULTIPLE);
 		while ((mapping = ao2_iterator_next(aliases))) {
+			char alias[strlen(mapping->alias) + 1];
+			strcpy(alias, mapping->alias); /* safe */
 			mailbox = NULL;
 			context = NULL;
 			ast_debug(3, "Found alias mapping: %s -> %s\n", mapping->alias, box);
-			separate_mailbox(ast_strdupa(mapping->alias), &mailbox, &context);
+			separate_mailbox(alias, &mailbox, &context);
 			ast_publish_mwi_state_channel(mailbox, context, new + urgent, old, channel_id);
 			ao2_ref(mapping, -1);
 		}
@@ -8403,12 +8375,12 @@ static int forward_message(struct ast_channel *chan, char *context, struct vm_st
 			directory_app = pbx_findapp("Directory");
 			if (directory_app) {
 				char vmcontext[256];
-				char *old_context;
-				char *old_exten;
+				char old_context[strlen(ast_channel_context(chan)) + 1];
+				char old_exten[strlen(ast_channel_exten(chan)) + 1];
 				int old_priority;
 				/* make backup copies */
-				old_context = ast_strdupa(ast_channel_context(chan));
-				old_exten = ast_strdupa(ast_channel_exten(chan));
+				strcpy(old_context, ast_channel_context(chan)); /* safe */
+				strcpy(old_exten, ast_channel_exten(chan)); /* safe */
 				old_priority = ast_channel_priority(chan);
 
 				/* call the Directory, changes the channel */
@@ -9089,7 +9061,6 @@ static int imap_remove_file(char *dir, int msgnum)
 static int imap_delete_old_greeting (char *dir, struct vm_state *vms)
 {
 	char *file, *filename;
-	char *attachment;
 	char arg[11];
 	int i;
 	BODY* body;
@@ -9118,17 +9089,22 @@ static int imap_delete_old_greeting (char *dir, struct vm_state *vms)
 		mail_fetchstructure(vms->mailstream, i + 1, &body);
 		/* We have the body, now we extract the file name of the first attachment. */
 		if (body->nested.part->next && body->nested.part->next->body.parameter->value) {
-			attachment = ast_strdupa(body->nested.part->next->body.parameter->value);
+			char *attachment = body->nested.part->next->body.parameter->value;
+			char copy[strlen(attachment) + 1];
+
+			strcpy(copy, attachment); /* safe */
+			attachment = copy;
+
+			filename = strsep(&attachment, ".");
+			if (!strcmp(filename, file)) {
+				snprintf(arg, sizeof(arg), "%d", i + 1);
+				mail_setflag(vms->mailstream, arg, "\\DELETED");
+			}
 		} else {
 			ast_log(AST_LOG_ERROR, "There is no file attached to this IMAP message.\n");
 			ast_mutex_unlock(&vms->lock);
 			return -1;
 		}
-		filename = strsep(&attachment, ".");
-		if (!strcmp(filename, file)) {
-			snprintf(arg, sizeof(arg), "%d", i + 1);
-			mail_setflag(vms->mailstream, arg, "\\DELETED");
-		}
 	}
 	mail_expunge(vms->mailstream);
 
@@ -13744,26 +13720,30 @@ static void load_zonemessages(struct ast_config *cfg)
 
 	var = ast_variable_browse(cfg, "zonemessages");
 	while (var) {
-		struct vm_zone *z;
-		char *msg_format, *tzone;
+		if (var->value) {
+			struct vm_zone *z;
+			char *msg_format, *tzone;
+			char storage[strlen(var->value) + 1];
 
-		z = ast_malloc(sizeof(*z));
-		if (!z) {
-			return;
-		}
+			z = ast_malloc(sizeof(*z));
+			if (!z) {
+				return;
+			}
 
-		msg_format = ast_strdupa(var->value);
-		tzone = strsep(&msg_format, "|,");
-		if (msg_format) {
-			ast_copy_string(z->name, var->name, sizeof(z->name));
-			ast_copy_string(z->timezone, tzone, sizeof(z->timezone));
-			ast_copy_string(z->msg_format, msg_format, sizeof(z->msg_format));
-			AST_LIST_LOCK(&zones);
-			AST_LIST_INSERT_HEAD(&zones, z, list);
-			AST_LIST_UNLOCK(&zones);
-		} else {
-			ast_log(AST_LOG_WARNING, "Invalid timezone definition at line %d\n", var->lineno);
-			ast_free(z);
+			strcpy(storage, var->value); /* safe */
+			msg_format = storage;
+			tzone = strsep(&msg_format, "|,");
+			if (msg_format) {
+				ast_copy_string(z->name, var->name, sizeof(z->name));
+				ast_copy_string(z->timezone, tzone, sizeof(z->timezone));
+				ast_copy_string(z->msg_format, msg_format, sizeof(z->msg_format));
+				AST_LIST_LOCK(&zones);
+				AST_LIST_INSERT_HEAD(&zones, z, list);
+				AST_LIST_UNLOCK(&zones);
+			} else {
+				ast_log(AST_LOG_WARNING, "Invalid timezone definition at line %d\n", var->lineno);
+				ast_free(z);
+			}
 		}
 		var = var->next;
 	}
@@ -15187,6 +15167,11 @@ static int unload_module(void)
 #endif
 	ao2_ref(inprocess_container, -1);
 
+	ao2_container_unregister("voicemail_alias_mailbox_mappings");
+	ao2_cleanup(alias_mailbox_mappings);
+	ao2_container_unregister("voicemail_mailbox_alias_mappings");
+	ao2_cleanup(mailbox_alias_mappings);
+
 	if (poll_thread != AST_PTHREADT_NULL)
 		stop_poll_thread();
 
@@ -15258,6 +15243,7 @@ static int load_module(void)
 	if (!mailbox_alias_mappings) {
 		ast_log(LOG_ERROR, "Unable to create mailbox_alias_mappings container\n");
 		ao2_cleanup(inprocess_container);
+		ao2_container_unregister("voicemail_alias_mailbox_mappings");
 		ao2_cleanup(alias_mailbox_mappings);
 		return AST_MODULE_LOAD_DECLINE;
 	}
@@ -15265,6 +15251,7 @@ static int load_module(void)
 	if (res) {
 		ast_log(LOG_ERROR, "Unable to register mailbox_alias_mappings container\n");
 		ao2_cleanup(inprocess_container);
+		ao2_container_unregister("voicemail_alias_mailbox_mappings");
 		ao2_cleanup(alias_mailbox_mappings);
 		ao2_cleanup(mailbox_alias_mappings);
 		return AST_MODULE_LOAD_DECLINE;
@@ -15273,9 +15260,6 @@ static int load_module(void)
 	/* compute the location of the voicemail spool directory */
 	snprintf(VM_SPOOL_DIR, sizeof(VM_SPOOL_DIR), "%s/voicemail/", ast_config_AST_SPOOL_DIR);
 
-	/* Now that we have a spool directory, clean up old lock files */
-	cleanup_orphaned_lock_files(VM_SPOOL_DIR);
-
 	if (!(mwi_subscription_tps = ast_taskprocessor_get("app_voicemail", 0))) {
 		ast_log(AST_LOG_WARNING, "failed to reference mwi subscription taskprocessor.  MWI will not work\n");
 	}

asterisk-17.1.0-summary.html

@@ -0,0 +1,372 @@
+<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN"http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd"><html xmlns="http://www.w3.org/1999/xhtml"><title>Release Summary - asterisk-17.1.0</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-17.1.0</h3><h3 align="center">Date: 2019-12-23</h3><h3 align="center">&lt;asteriskteam@digium.com&gt;</h3><hr><h2 align="center">Table of Contents</h2><ol>
+<li><a href="#summary">Summary</a></li>
+<li><a href="#contributors">Contributors</a></li>
+<li><a href="#closed_issues">Closed Issues</a></li>
+<li><a href="#commits">Other Changes</a></li>
+<li><a href="#diffstat">Diffstat</a></li>
+</ol><hr><a name="summary"><h2 align="center">Summary</h2></a><center><a href="#top">[Back to Top]</a></center><p>This release is a point release of an existing major version. The changes included were made to address problems that have been identified in this release series, or are minor, backwards compatible new features or improvements. Users should be able to safely upgrade to this version if this release series is already in use. Users considering upgrading from a previous version are strongly encouraged to review the UPGRADE.txt document as well as the CHANGES document for information about upgrading to this release series.</p><p>The data in this summary reflects changes that have been made since the previous release, asterisk-17.0.0.</p><hr><a name="contributors"><h2 align="center">Contributors</h2></a><center><a href="#top">[Back to Top]</a></center><p>This table lists the people who have submitted code, those that have tested patches, as well as those that reported issues on the issue tracker that were resolved in this release. For coders, the number is how many of their patches (of any size) were committed into this release. For testers, the number is the number of times their name was listed as assisting with testing a patch. Finally, for reporters, the number is the number of issues that they reported that were affected by commits that went into this release.</p><table width="100%" border="0">
+<tr><th width="33%">Coders</th><th width="33%">Testers</th><th width="33%">Reporters</th></tr>
+<tr valign="top"><td width="33%">23 George Joseph <gjoseph@digium.com><br/>23 Sean Bright <sean.bright@gmail.com><br/>16 Joshua Colp <jcolp@digium.com><br/>13 Kevin Harwell <kharwell@digium.com><br/>6 Corey Farrell <git@cfware.com><br/>5 Asterisk Development Team <asteriskteam@digium.com><br/>5 Alexei Gradinari <alex2grad@gmail.com><br/>5 Ben Ford <bford@digium.com><br/>4 Frederic LE FOLL <frederic.lefoll@c-s.fr><br/>3 Igor Goncharovsky <igor.goncharovsky@gmail.com><br/>2 Salah Ahmed <txrubel@gmail.com><br/>2 Joshua C. Colp <jcolp@sangoma.com><br/>2 Torrey Searle <torrey@voxbone.com><br/>2 lvl <digium@lvlconsultancy.nl><br/>1 Thomas Arimont (license 5525)<br/>1 Pascal Cadotte Michaud <pcm@wazo.io><br/>1 Martin Tomec <tomec.martin@gmail.com><br/>1 Walter Doekes <walter+asterisk@wjd.nu><br/>1 Stas Kobzar <stas@modulis.ca><br/>1 Jonathan Rose <jrose@digium.com><br/>1 Michael Goryainov<br/>1 Chris-Savinovich <csavinovich@digium.com><br/>1 Michael Cargile <mikec@vicidial.com><br/>1 Chris Savinovich <csavinovich@digium.com><br/>1 sungtae kim <pchero21@gmail.com><br/>1 Florian Floimair <f.floimair@commend.com><br/>1 cmaj <chris@penguinpbx.com><br/>1 Christoph Moench-Tegeder <cmt@burggraben.net><br/>1 Dan Cropp <dan@amtelco.com><br/>1 Guido Falsi <madpilot@FreeBSD.org><br/></td><td width="33%">1 tests/test_utils.c.<br/></td><td width="33%">5 Kevin Harwell <kharwell@digium.com><br/>5 Joshua C. Colp <jcolp@digium.com><br/>4 Frederic LE FOLL <frederic.lefoll@c-s.fr><br/>3 Salah Ahmed <txrubel@gmail.com><br/>3 Ross Beer <ross.beer@voicehost.co.uk><br/>2 Alexei Gradinari <alex2grad@gmail.com><br/>2 Torrey Searle <tsearle@gmail.com><br/>2 Ross Beer<br/>2 Joshua Elson <joshelson@gmail.com><br/>2 George Joseph <gjoseph@digium.com><br/>2 Ruddy G <plugworld@micnes.com><br/>1 Michael Cargile <mikec@vicidial.com><br/>1 Walter Doekes <walter+asterisk@wjd.nu><br/>1 Ted G<br/>1 Martin Tomec <tomec.martin@gmail.com><br/>1 Chris Savinovich <csavinovich@digium.com><br/>1 Byron Clark <bclark@getjive.com><br/>1 Niklas Larsson <niklas@tese.se><br/>1 Jonas Swiatek <jonas@telzio.com><br/>1 Yoooooo Ha <n1906374c@e.ntu.edu.sg><br/>1 Michael <ringo@vianet.ca><br/>1 Eliel Sardañons <eliels@gmail.com><br/>1 Guido Falsi <madpilot@freebsd.org><br/>1 Gregory Massel <greg@csurf.co.za><br/>1 Dan Cropp<br/>1 Jeremiah Gadd <jeremygadd@gmail.com><br/>1 Bernhard Schmidt<br/>1 Stas Kobzar <stas@modulis.ca><br/>1 nappsoft <infos@nappsoft.ch><br/>1 Pascal Cadotte Michaud <pascal.cadotte@gmail.com><br/>1 Kilburn <kilburna@gmail.com><br/>1 Ian Jones <tech@iljones.net><br/>1 Michael Goryainov <gms4nlt@gmail.com><br/>1 Bernhard Schmidt <berni@birkenwald.de><br/>1 Ted G <tgwaste@gmail.com><br/>1 Marian Piater <marian.piater@voipsun.cz><br/>1 Aheliotech <phones@aheliotech.com><br/>1 Dan Cropp <dan@amtelco.com><br/>1 Mark <mark@wrapped.cx><br/>1 Alexander Traud <pabstraud@compuserve.com><br/>1 Speed Dial Dave <speed_dial_dave@gmx.com><br/>1 Timothy Vanderaerden <timothy.vanderaerden@optimise-group.be><br/>1 Niklas Larsson<br/>1 Andrey  V. T. <avt1203@gmail.com><br/>1 Christoph Moench-Tegeder <cmt@burggraben.net><br/>1 Florian Floimair <f.floimair@commend.com><br/>1 Jonathan Harris <lardconcepts@gmail.com><br/>1 Daniel <depeee@gmail.com><br/>1 Sam Banks <sam.banks.nz@gmail.com><br/>1 Eliel Sardañons<br/>1 Alexander Traud<br/>1 Cyril Ramière <cyril.ramiere@ino.global><br/>1 Jørgen H <asterisk.org@hovland.cx><br/>1 cmaj <chris@penguinpbx.com><br/>1 Juan Martin <jmartin79@yandex.com><br/>1 lvl <digium@lvlconsultancy.nl><br/></td></tr>
+</table><hr><a name="closed_issues"><h2 align="center">Closed Issues</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a list of all issues from the issue tracker that were closed by changes that went into this release.</p><h3>Security</h3><h4>Category: Channels/chan_sip/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28589">ASTERISK-28589</a>: chan_sip: Depending on configuration an INVITE can alter Addr of a peer<br/>Reported by: Andrey  V. T.<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=665a94cb7622dd5f9591df5d68962e41266ee47b">[665a94cb76]</a> Ben Ford -- chan_sip.c: Prevent address change on unauthenticated SIP request.</li>
+</ul><br><h4>Category: Core/ManagerInterface</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28580">ASTERISK-28580</a>: Bypass SYSTEM write permission in manager action allows system commands execution<br/>Reported by: Eliel Sardañons<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6b1ba589671a0e6e9a52b89af6e5398f040ef99f">[6b1ba58967]</a> George Joseph -- manager.c:  Prevent the Originate action from running the Originate app</li>
+</ul><br><h4>Category: Resources/res_pjsip_t38</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28495">ASTERISK-28495</a>: res_pjsip_t38: 200 OK with SDP answer with declined stream causes crash<br/>Reported by: Alexei Gradinari<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9d4f1e8ebe5dc1152c59aee60955fc29ae3b3191">[9d4f1e8ebe]</a> Alexei Gradinari -- AST-2019-004 - res_pjsip_t38.c: Add NULL checks before using session media</li>
+</ul><br><h3>Improvement</h3><h4>Category: Applications/app_voicemail</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28567">ASTERISK-28567</a>: Problem with ASTERISK-20207: Asterisk should clear out any .lock files in the voice mail directory on startup.<br/>Reported by: Michael<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b90399498787b898da89d21564251e6aa84dda98">[b903994987]</a> Sean Bright -- Revert "app_voicemail: Cleanup stale lock files on module load"</li>
+</ul><br><h4>Category: Applications/app_voicemail/ODBC</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-22192">ASTERISK-22192</a>: [patch] Allow voicemail forwards with ODBC backend when format differs from attachfmt column<br/>Reported by: cmaj<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=aa0973f868ac463bc31b760f364afb2ab691b030">[aa0973f868]</a> cmaj -- app_voicemail.c: Support multiple file formats for forwarded messages.</li>
+</ul><br><h4>Category: Core/CodecInterface</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28512">ASTERISK-28512</a>: Add pass-through support for H.265 (HEVC) codec<br/>Reported by: Florian Floimair<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d7a3e4f5cf6dadc0c8fa9b0cdc0c2d13b45d33ab">[d7a3e4f5cf]</a> Florian Floimair -- core: Add H.265/HEVC passthrough support</li>
+</ul><br><h4>Category: Documentation</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28586">ASTERISK-28586</a>: Typo in README-SERIOUSLY.bestpractices.md<br/>Reported by: Sam Banks<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4bc1c170cd09ef890133e438e7030ac9b6eeba85">[4bc1c170cd]</a> Sean Bright -- README-SERIOUSLY.bestpractices.md: Speling correetions.</li>
+</ul><br><h4>Category: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28542">ASTERISK-28542</a>: [patch] add the ability for asterisk to generate on-hold re-invites<br/>Reported by: Torrey Searle<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=55b760d7625f3c78c8f4de5587a2051b7a16dfa4">[55b760d762]</a> Torrey Searle -- channel/chan_pjsip: add dialplan function for music on hold</li>
+</ul><br><h4>Category: Resources/res_pjsip_outbound_registration</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28602">ASTERISK-28602</a>: res_pjsip_outbound_registration: Maximum retries reached<br/>Reported by: Daniel<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=eea2d499f4cc0982ff8817640c803dbd57a008ed">[eea2d499f4]</a> Joshua Colp -- res_pjsip_outbound_registration: Extend documentation for "max_retries".</li>
+</ul><br><h3>Bug</h3><h4>Category: .Release/Targets</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28488">ASTERISK-28488</a>: pjsip mwi: n+1 sip notify's sent on re-register<br/>Reported by: Chris Savinovich<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a36fb473fe3cda2fd56a44f4e09234a167ef44d2">[a36fb473fe]</a> Kevin Harwell -- res_pjsip_mwi: add better handling of solicited vs unsolicited subscriptions</li>
+</ul><br><h4>Category: Applications/app_amd</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28608">ASTERISK-28608</a>: app_amd: Use time calculation to calculate timeout<br/>Reported by: Michael Cargile<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e23b2856d09a7843e3d8dda6f1f3b72de89a48e6">[e23b2856d0]</a> Michael Cargile -- app_amd: Fixed timeout issue</li>
+</ul><br><h4>Category: Applications/app_chanisavail</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28527">ASTERISK-28527</a>: ChanIsAvail() creates a CDR if unanswered=yes is set in cdr.conf<br/>Reported by: Frederic LE FOLL<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=50997de8878f0ad08147540395dab3bc4ffb0d2a">[50997de887]</a> Frederic LE FOLL -- ChanIsAvail() generates a CDR when unanswered=yes in cdr.conf.</li>
+</ul><br><h4>Category: Applications/app_meetme</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28604">ASTERISK-28604</a>: app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0<br/>Reported by: George Joseph<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3d29b06e37de0480ba4c0e3fb61ff690d67572aa">[3d29b06e37]</a> Joshua C. Colp -- configure: Add check for MySQL client bool and my_bool type usage.</li>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=36b28c98dd8fd8906b9b6eefee2489820f1a1295">[36b28c98dd]</a> George Joseph -- Build:  Fix compile issues with seldom used modules</li>
+</ul><br><h4>Category: Applications/app_queue</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28644">ASTERISK-28644</a>: Stale comment in app_queue about ring_entry exception<br/>Reported by: Walter Doekes<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e1eb5e8dc269ca90c4503d8c8da058b3f3822eb9">[e1eb5e8dc2]</a> Walter Doekes -- app_queue: Fix old confusing comment about when the members are called</li>
+</ul><br><h4>Category: Applications/app_voicemail/IMAP</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28505">ASTERISK-28505</a>: app_voicemail/IMAP: segfault in leave_voicemail because not checking mailstream<br/>Reported by: Alexei Gradinari<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=052ab9d96604ba5f4e6666788bf93d357607fb95">[052ab9d966]</a> Alexei Gradinari -- app_voicemail/IMAP: check mailstream not NULL in leave_voicemail</li>
+</ul><br><h4>Category: Bridges/bridge_native_rtp</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28637">ASTERISK-28637</a>: chan_sip+native_bridge_rtp: directmedia compatibility check failure when negociated ptime is not default ptime.<br/>Reported by: Frederic LE FOLL<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3e73893e53dddfc43f47dc6a451bc9e2a5387345">[3e73893e53]</a> Frederic LE FOLL -- chan_sip+native_bridge_rtp: no directmedia for ptime other than default ptime.</li>
+</ul><br><h4>Category: Bridges/bridge_softmix</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28618">ASTERISK-28618</a>: bridge_softmix: hold not cleared when joining a softmix bridge<br/>Reported by: Kevin Harwell<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8b4610acfe476f111a52234e1fd2cab03b7d58f9">[8b4610acfe]</a> Kevin Harwell -- bridge_softmix: clear hold when joining a softmix bridge</li>
+</ul><br><h4>Category: CDR/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28566">ASTERISK-28566</a>: CDR backend unload problem during active call(s)<br/>Reported by: Marian Piater<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1dc3451a3464bdc52d03344535dd7ac7751777a2">[1dc3451a34]</a> Sean Bright -- cdr_mysql: Don't clean up on unload unless we can unregister from CDRs</li>
+</ul><br><h4>Category: CDR/cdr_pgsql</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28571">ASTERISK-28571</a>: cdr_pgsql: accesses obsolete (and finally removed) column<br/>Reported by: Christoph Moench-Tegeder<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=79cc8ae3b809629df8f4739dfd537aa98d0c43ec">[79cc8ae3b8]</a> Christoph Moench-Tegeder -- cdr_pgsql cel_pgsql res_config_pgsql: compatibility with PostgreSQL 12</li>
+</ul><br><h4>Category: Channels/chan_dahdi</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28615">ASTERISK-28615</a>: chan_dahdi: PRI span status may stay "Down, Active" after a short alarm<br/>Reported by: Frederic LE FOLL<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d3dd4c5459846b0bff4864f8089418a475c603db">[d3dd4c5459]</a> Frederic LE FOLL -- chan_dahdi: PRI span status may stay "Down, Active" after a short alarm</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28536">ASTERISK-28536</a>: Asterisk release candidates fail to build on FreeBSD<br/>Reported by: Guido Falsi<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5ff2f7a0169d30952f1ba9c5ff48027e6c950f03">[5ff2f7a016]</a> Guido Falsi -- chan_dahdi: Fix build with clang/llvm</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28525">ASTERISK-28525</a>: chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up<br/>Reported by: Frederic LE FOLL<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9e67c925027ec59154406f82cea4cb3934bdf2a7">[9e67c92502]</a> Frederic LE FOLL -- chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up</li>
+</ul><br><h4>Category: Channels/chan_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28578">ASTERISK-28578</a>: race condition on pjsip channelstats command<br/>Reported by: Salah Ahmed<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=40acd7d1987e38d010430e013dd3dc0d54f8d4b9">[40acd7d198]</a> Salah Ahmed -- Crash during "pjsip show channelstats" execution</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28561">ASTERISK-28561</a>: Asterisk Deadlocks<br/>Reported by: Aheliotech<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ae761c74738c04c11b72b539cc7e852bef6da260">[ae761c7473]</a> Joshua Colp -- pbx: deadlock when outgoing dialed channel hangs up too quickly</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28086">ASTERISK-28086</a>: chan_pjsip: Crash when initiating PlayDTMF over AMI<br/>Reported by: Jeremiah Gadd<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=71f86e78b6e5694ef90baab9683f4f519d9540c1">[71f86e78b6]</a> lvl -- chan_pjsip: Prevent segfault when running PlayDTMF on hungup channel</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28538">ASTERISK-28538</a>: chan_pjsip: Deadlock on fax detection<br/>Reported by: Joshua C. Colp<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4d1baa3ae86b7adb6988ea73aa93f9966b714a4a">[4d1baa3ae8]</a> Joshua Colp -- chan_pjsip: Relock correct channel during "fax" redirect.</li>
+</ul><br><h4>Category: Channels/chan_sip/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28637">ASTERISK-28637</a>: chan_sip+native_bridge_rtp: directmedia compatibility check failure when negociated ptime is not default ptime.<br/>Reported by: Frederic LE FOLL<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3e73893e53dddfc43f47dc6a451bc9e2a5387345">[3e73893e53]</a> Frederic LE FOLL -- chan_sip+native_bridge_rtp: no directmedia for ptime other than default ptime.</li>
+</ul><br><h4>Category: Channels/chan_unistim</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25592">ASTERISK-25592</a>: chan_unistim: Clang Warning: variable sized type not at end of a struct<br/>Reported by: Alexander Traud<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=056ddf76cee4f11dbb6529f9581c93301ebc3507">[056ddf76ce]</a> Igor Goncharovsky -- chan_unistim: Fix clang warning: variable sized type not at end of a struct</li>
+</ul><br><h4>Category: Codecs/codec_resample</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28511">ASTERISK-28511</a>: codec_resample: Bad sound quality when up sampling from SLIN16 to SLIN32<br/>Reported by: Ruddy G<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=75d6418d8eada0e59c55f7fcc5b680b31caf4303">[75d6418d8e]</a> Sean Bright -- codec_resample: Ensure OUTSIDE_SPEEX is defined when necessary</li>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=34ab9964f57721fc4d9d54bc8cd7bab96bd52602">[34ab9964f5]</a> Sean Bright -- codec_resample: Upgrade speex_resample to fix up-sampling bug</li>
+</ul><br><h4>Category: Core/BuildSystem</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28487">ASTERISK-28487</a>: compile menuselect on gentoo<br/>Reported by: Kilburn<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a5f05eed70694e16f332e5433756ec27abd34e5c">[a5f05eed70]</a> Sean Bright -- menuselect: Fix curses build on Gentoo Linux</li>
+</ul><br><h4>Category: Core/Channels</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28499">ASTERISK-28499</a>: translate: Crash when frame does not have a "src" field set<br/>Reported by: Gregory Massel<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=61c01df560f4cd193dd9bf2b0aaf7cc30b873a6a">[61c01df560]</a> Joshua Colp -- AST-2019-005 - translate: Don't assume all frames will have a src.</li>
+</ul><br><h4>Category: Core/Configuration</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-23756">ASTERISK-23756</a>: setvar directive when used in template and a child of said template, results in duplicate variable names<br/>Reported by: Michael Goryainov<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2fa296e7d41526f000841c083fcbd4d38a7fd776">[2fa296e7d4]</a> Michael Goryainov -- channels: Allow updating variable value</li>
+</ul><br><h4>Category: Core/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28498">ASTERISK-28498</a>: cel / cdr: Event times may be incorrect<br/>Reported by: Joshua C. Colp<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=108b1abbd9d2ee0148b4833f581d9612ab46256c">[108b1abbd9]</a> Joshua Colp -- cdr / cel: Use event time at event creation instead of processing.</li>
+</ul><br><h4>Category: Core/RTP</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28480">ASTERISK-28480</a>: json integer overflow in ssrc and timestamp<br/>Reported by: Salah Ahmed<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a305f2fdcb6c665c7bdfe779a370f09e1be35794">[a305f2fdcb]</a> Kevin Harwell -- various modules: json integer overflow</li>
+</ul><br><h4>Category: Core/Stasis</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28553">ASTERISK-28553</a>: stasis.c: Crash during unload<br/>Reported by: Kevin Harwell<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=57fa6045711d2533de5ea08ec4d30b4ccfc885c2">[57fa604571]</a> Joshua Colp -- stasis: Pass bumped topic_all reference to proxy_dtor.</li>
+</ul><br><h4>Category: Core/UDPTL</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28483">ASTERISK-28483</a>: packet lost on UDPTL wrap around<br/>Reported by: Torrey Searle<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=44af3e9018108b01f219fc40a7fdb400a0d477d1">[44af3e9018]</a> Torrey Searle -- main/udptl.c: correctly handle udptl sequence wrap around</li>
+</ul><br><h4>Category: Functions/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28626">ASTERISK-28626</a>: Missing arguments in PJSIP_CONTACT function documentation<br/>Reported by: Pascal Cadotte Michaud<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=450173a0ae3aace0bbdcbfbcd336c5069747b0d1">[450173a0ae]</a> Pascal Cadotte Michaud -- PJSIP_CONTACT: add missing argument documentation</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26481">ASTERISK-26481</a>: FILE function grabs garbage along with read data when target line has no newline<br/>Reported by: Jonathan Harris<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=92bb381d5d1878b9268e5da23a2e85cba3cdb715">[92bb381d5d]</a> Sean Bright -- func_env: Prevent FILE() from reading garbage at end-of-file</li>
+</ul><br><h4>Category: General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28609">ASTERISK-28609</a>: Memory Leak in res_rtp_asterisk.c<br/>Reported by: Ted G<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=27a4a3c761613904a6105aec6ff211e206cbbf67">[27a4a3c761]</a> George Joseph -- res_rtp_asterisk:  Add frame list cleanups to ast_rtp_read</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28590">ASTERISK-28590</a>: utils.c throws repeated warnings; "pthread_attr_setstacksize: Invalid argument"<br/>Reported by: Speed Dial Dave<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b3c56c7fa516944852ad2792b00410fc10448beb">[b3c56c7fa5]</a> Sean Bright -- utils.h: Set lower bound for thread stack size to PTHREAD_STACK_MIN</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28523">ASTERISK-28523</a>: Asterisk 16.5.0 Memory leak<br/>Reported by: Cyril Ramière<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f821e8107180e7c49c00c16ae8f998e12b51e79a">[f821e81071]</a> Kevin Harwell -- res_sorcery_memory_cache: stale item update leak</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28472">ASTERISK-28472</a>: Asterisk occasionally passes a NULL as srtp->session to srtp_protect/unprotect causing SEGV<br/>Reported by: Jonas Swiatek<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5daa9bbaee476fcf11223d174d90a987d96fb1a5">[5daa9bbaee]</a> Kevin Harwell -- srtp: Fix possible race condition, and add NULL checks</li>
+</ul><br><h4>Category: PBX/pbx_config</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28534">ASTERISK-28534</a>: Segmentation fault when there is no priority for an extension<br/>Reported by: Timothy Vanderaerden<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8d0edf2b3714bd1096902d15d240c2a43c270bd5">[8d0edf2b37]</a> Sean Bright -- pbx: Prevent Realtime switch crash on invalid priority</li>
+</ul><br><h4>Category: Resources/res_ari</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28585">ASTERISK-28585</a>: ari/resource_events: Crash in event session cleanup<br/>Reported by: Kevin Harwell<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e37d5461095e93ad274dd9893d21750b8d2cacc1">[e37d546109]</a> Joshua Colp -- res_ari_events: Add module reference when a WebSocket is open.</li>
+</ul><br><h4>Category: Resources/res_calendar_exchange</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28572">ASTERISK-28572</a>: Memory leaks in res_calendar_exchange and res_calendar_icalendar<br/>Reported by: Yoooooo Ha<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b3792e1288d481718875fde9da35b9332301d766">[b3792e1288]</a> Sean Bright -- res_calendar: Resolve memory leak on calendar destruction</li>
+</ul><br><h4>Category: Resources/res_calendar_icalendar</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28572">ASTERISK-28572</a>: Memory leaks in res_calendar_exchange and res_calendar_icalendar<br/>Reported by: Yoooooo Ha<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b3792e1288d481718875fde9da35b9332301d766">[b3792e1288]</a> Sean Bright -- res_calendar: Resolve memory leak on calendar destruction</li>
+</ul><br><h4>Category: Resources/res_parking</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28631">ASTERISK-28631</a>: res_parking: Doesn't park when parkee and parker are the same<br/>Reported by: Ross Beer<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=41d58a4ce2ea34345c6c9645238eb80ae39e15ac">[41d58a4ce2]</a> Joshua Colp -- parking: Fall back to parker channel name even if it matches parkee.</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28616">ASTERISK-28616</a>: parking: Deadlock when multi call parking<br/>Reported by: Joshua C. Colp<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=de433cdcaf64070164300a51ab8ab0d0c596c71e">[de433cdcaf]</a> Joshua Colp -- parking: Fix case where we can't get the parker.</li>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d638d9c6c6b9ffe8feb0ee35466a6d884a5d04e4">[d638d9c6c6]</a> Joshua Colp -- parking: Use channel snapshot instead of channel.</li>
+</ul><br><h4>Category: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28641">ASTERISK-28641</a>: res_pjsip Segfaults when realtime configuration to an AOR points to a not existent AOR<br/>Reported by: Ross Beer<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4e057eb9d2c3572373c95f5b272ac8950efcbfec">[4e057eb9d2]</a> Sean Bright -- res_pjsip_registrar.c: Prevent potential double free if AOR is not found</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28544">ASTERISK-28544</a>: Wrong contact representation in ipv6 mode<br/>Reported by: Jørgen H<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6527eb8213e185e3b5e3a194764d688f4d73a088">[6527eb8213]</a> Sean Bright -- res_pjsip_transport_websocket: Don't put brackets around local_name if IPv6</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28521">ASTERISK-28521</a>: pjsip: Memory Leak<br/>Reported by: Mark<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7c0435f854d34f2723fa744ee599845b25bcc5f8">[7c0435f854]</a> George Joseph -- pjproject_bundled:  Revert pjproject 2.9 commits causing leaks</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28228">ASTERISK-28228</a>: res_pjsip: pjsip show contacts prints double entries<br/>Reported by: Ian Jones<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=20459d4cacc94335b7e39ed696f336c521f872ba">[20459d4cac]</a> Joshua Colp -- res_pjsip: Fix multiple of the same contact in "pjsip show contacts".</li>
+</ul><br><h4>Category: Resources/res_pjsip_mwi</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28575">ASTERISK-28575</a>: MWI Send Notify Crash on 16.6<br/>Reported by: Joshua Elson<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=45c0d991857ec1ee61d12bc26a106cc8d5cc84a3">[45c0d99185]</a> Kevin Harwell -- res_pjsip_mwi: potential double unref, and potential unwanted double link</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28552">ASTERISK-28552</a>: res_pjsip_mwi: Frack during unload on unsolicited_mwi container<br/>Reported by: Kevin Harwell<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=996fc40e2b489c7df13567d2255d8238c0cad1bd">[996fc40e2b]</a> Kevin Harwell -- res_pjsip_mwi: use an ao2_global object for mwi containers</li>
+</ul><br><h4>Category: Resources/res_pjsip_outbound_registration</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28624">ASTERISK-28624</a>: res_pjsip_outbound_registration: add SRV failover<br/>Reported by: Kevin Harwell<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8c99930375bb73072a20e30893fecd1c382f5671">[8c99930375]</a> Kevin Harwell -- res_pjsip_outbound_registration: add support for SRV failover</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28521">ASTERISK-28521</a>: pjsip: Memory Leak<br/>Reported by: Mark<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7c0435f854d34f2723fa744ee599845b25bcc5f8">[7c0435f854]</a> George Joseph -- pjproject_bundled:  Revert pjproject 2.9 commits causing leaks</li>
+</ul><br><h4>Category: Resources/res_pjsip_path</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28463">ASTERISK-28463</a>: res_pjsip_path: Crash when invalid contact is configured<br/>Reported by: Juan Martin<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=51cf060c6ccca4d545ab4723f412318388b185f1">[51cf060c6c]</a> Sean Bright -- res_pjsip_registrar: Validate Contact URI before adding to responses</li>
+</ul><br><h4>Category: Resources/res_pjsip_sdp_rtp</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28659">ASTERISK-28659</a>: res_pjsip_sdp_rtp: Bundle includes non-existent media stream if codecs create additional streams and offer does not have them<br/>Reported by: nappsoft<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5949f9a86a14892ca66d34207f780ed59bff59e5">[5949f9a86a]</a> Joshua C. Colp -- res_pjsip_session: Set stream state on created streams for incoming SDP.</li>
+</ul><br><h4>Category: Resources/res_pjsip_session</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28659">ASTERISK-28659</a>: res_pjsip_sdp_rtp: Bundle includes non-existent media stream if codecs create additional streams and offer does not have them<br/>Reported by: nappsoft<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5949f9a86a14892ca66d34207f780ed59bff59e5">[5949f9a86a]</a> Joshua C. Colp -- res_pjsip_session: Set stream state on created streams for incoming SDP.</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28445">ASTERISK-28445</a>: res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on hangup when TEST_FRAMEWORK enabled<br/>Reported by: Bernhard Schmidt<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4d56adf8fbbbaa6c05c547eb482f1d154ec006d4">[4d56adf8fb]</a> Sean Bright -- res_pjsip_session.c: Prevent use-after-free with TEST_FRAMEWORK enabled</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28086">ASTERISK-28086</a>: chan_pjsip: Crash when initiating PlayDTMF over AMI<br/>Reported by: Jeremiah Gadd<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=71f86e78b6e5694ef90baab9683f4f519d9540c1">[71f86e78b6]</a> lvl -- chan_pjsip: Prevent segfault when running PlayDTMF on hungup channel</li>
+</ul><br><h4>Category: Resources/res_pjsip_t38</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28621">ASTERISK-28621</a>: Enforce T.38 error correction mode at 200 ok received  <br/>Reported by: Salah Ahmed<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4ac0299bfbdfc851040ab2e8136e0078378ea19a">[4ac0299bfb]</a> Salah Ahmed -- res_pjsip_t38: T.38 error correction mode selection at 200 ok received</li>
+</ul><br><h4>Category: Resources/res_rtp_asterisk</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28576">ASTERISK-28576</a>: res_rtp_asterisk: ICE Completion Crash when sent packet length doesn't match<br/>Reported by: Joshua Elson<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0c486e7edf9afae2e649aba47be2464e9a25db38">[0c486e7edf]</a> Joshua Colp -- res_rtp_asterisk: Always return provided DTLS packet length.</li>
+</ul><br><h4>Category: Resources/res_stasis</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28423">ASTERISK-28423</a>: ARI causes STASIS Deadlock<br/>Reported by: Ross Beer<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7202624b3b46bf013a08f26480a60fa2c515a80f">[7202624b3b]</a> George Joseph -- stasis: Don't hold app_registry and session locks unnecessarily</li>
+</ul><br><h4>Category: pjproject/pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28574">ASTERISK-28574</a>: pjproject fails to build on 16.6.0, works on 16.5<br/>Reported by: Niklas Larsson<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2652bda3a0bb0dd2b444a4e224ff3dcea2eb6e93">[2652bda3a0]</a> George Joseph -- pjproject_bundled:  Replace earlier reverts with official fixes.</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28509">ASTERISK-28509</a>: PJSIP cnonce generated on Linux contains 36 characters, NEC only supports up to 32 characters<br/>Reported by: Dan Cropp<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a1d38e19a267e06042c5c98d6204bd14a0a0c4e4">[a1d38e19a2]</a> Dan Cropp -- pjproject: Configurable setting for cnonce to include hyphens or not</li>
+</ul><br><h3>New Feature</h3><h4>Category: Applications/app_senddtmf</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28614">ASTERISK-28614</a>: app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead of only "sending"<br/>Reported by: lvl<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6345a002284e68ae5637088678a4491963d28845">[6345a00228]</a> lvl -- app_senddtmf: Add receive mode to AMI Action PlayDTMF</li>
+</ul><br><h4>Category: Core/Jitterbuffer</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28533">ASTERISK-28533</a>: func_jitterbuffer: Add support for video synchronization<br/>Reported by: Joshua C. Colp<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=926053d7bdf6245d9081550eadd5c61ed2aeb784">[926053d7bd]</a> Joshua Colp -- func_jitterbuffer: Add audio/video sync support.</li>
+</ul><br><h4>Category: Functions/func_curl</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28613">ASTERISK-28613</a>: func_curl: CURLOPT cannot set Content-Type header<br/>Reported by: Martin Tomec<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d579ec9cdf85ab31c1b7d9c8391d41b4ec513fc9">[d579ec9cdf]</a> Martin Tomec -- func_curl.c: Support custom http headers</li>
+</ul><br><h4>Category: Resources/res_musiconhold</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-17808">ASTERISK-17808</a>: [patch] Unregister a realtime moh class<br/>Reported by: Byron Clark<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9e26136ee6a2d09dfe26528d109eedd12a7948ef">[9e26136ee6]</a> sungtae kim -- res_musiconhold: Added unregister realtime moh class</li>
+</ul><br><h4>Category: pjproject/pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28489">ASTERISK-28489</a>: Channel variable SIPFROMDOMAIN for chan_pjsip to setup From header URI domain<br/>Reported by: Stas Kobzar<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3a246c2a69cd7f160d80c31058e99e9f99de6746">[3a246c2a69]</a> Stas Kobzar -- res_pjsip: Channel variable SIPFROMDOMAIN</li>
+</ul><br><hr><a name="commits"><h2 align="center">Commits Not Associated with an Issue</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a list of all changes that went into this release that did not reference a JIRA issue.</p><table width="100%" border="1">
+<tr><th>Revision</th><th>Author</th><th>Summary</th></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=de2b8ef23aa78a0b2c8e733e70edd8f733d25bed">de2b8ef23a</a></td><td>Asterisk Development Team</td><td>Update for 17.1.0-rc2</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a76d3103fb4aa420d9cb7af8cfc5b384d30d6658">a76d3103fb</a></td><td>George Joseph</td><td>Revert "chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up"</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c8563727d5343e18213fb499579a1f45d0566d4e">c8563727d5</a></td><td>Asterisk Development Team</td><td>Update for 17.1.0-rc1</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=08a6e8c55315deedeae5e7c1d0528286f1c8a153">08a6e8c553</a></td><td>Asterisk Development Team</td><td>Update CHANGES and UPGRADE.txt for 17.1.0</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6a89784b7820c301e8b438d140c836db84d3f2b9">6a89784b78</a></td><td>Joshua Colp</td><td>Revert "PJSIP_CONTACT: add missing argument documentation"</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f26e5bacc008fc5335cb260f75ae73ee3fcbb67a">f26e5bacc0</a></td><td>Sean Bright</td><td>res_pjsip_registrar.c: Prevent possible buffer overflow with domain aliases</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=88150323a235efeb051dc0cae9ef615dd34e86d1">88150323a2</a></td><td>Thomas Arimont</td><td>channel.c: Resolve issue with receiving SIP INFO packets for DTMF</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5b15a1c6393b380ad86daa40a08491d880f94e68">5b15a1c639</a></td><td>George Joseph</td><td>CI: Turn off shallow cloning altogether</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cc59e21409b22c109560b92a6073bffc30006f7c">cc59e21409</a></td><td>Sean Bright</td><td>media_cache.c: Various CLI improvements</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2a92e6b576334ac926d7d498cfab6081d16e702e">2a92e6b576</a></td><td>George Joseph</td><td>CI: Fix missing script block in jenkinsfiles</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f0d1ce50afd25a1269e680b90c8bb612bd543565">f0d1ce50af</a></td><td>George Joseph</td><td>CI: Fix missing script block in jenkinsfiles</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=46dceab33f5b54ee0f222646bb92654c07efe01f">46dceab33f</a></td><td>George Joseph</td><td>CI: Increase clone depth and do better cleanup</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=76ef36fafcb9a0250e036eb8172681b506a81ee0">76ef36fafc</a></td><td>Sean Bright</td><td>res_pjsip_registrar: Fix uninitlized variable warning</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=649733612d84fbc8566822513dfe9b6fc88c778c">649733612d</a></td><td>Alexei Gradinari</td><td>serializer: set high/low alert levels on whole pool</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8bc6fa0fbd82f4a57e492f0f92b2acd6454a47b7">8bc6fa0fbd</a></td><td>Kevin Harwell</td><td>various files - fix some alerts raised by lgtm code analysis</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ea3daa94c8c04e2037c5fd801bcd703a8209ac94">ea3daa94c8</a></td><td>Kevin Harwell</td><td>res_pjsip_session: initialize pending's topology to endpoint's</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2d665091a36882070dae8ac733aa551f260488ad">2d665091a3</a></td><td>George Joseph</td><td>ExternalMedia:  Change return object from ExternalMedia to Channel</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b8ae799ca9da487e87406edc932a575a600ac735">b8ae799ca9</a></td><td>Joshua Colp</td><td>res_rtp_asterisk: Remove a log message that slipped in.</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ba688e6891e069761ee996baab507e00e4da8278">ba688e6891</a></td><td>Joshua Colp</td><td>test_res_rtp: Enable FIR and REMB nominal tests.</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c84135d2a3527b383280a6732dcee5d30a5a5486">c84135d2a3</a></td><td>Chris Savinovich</td><td>test_taskprocessor.c: Fix test failure on Ubuntu</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=37ec88c4c8bd71d9a50209966274bbf8559562fa">37ec88c4c8</a></td><td>Kevin Harwell</td><td>serializer: move/add asterisk serializer pool functionality</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=299ba78b0955f17bfac5703b891d0d19466f99c1">299ba78b09</a></td><td>Kevin Harwell</td><td>res_pjsip/res_pjsip_mwi: use centralized serializer pools</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=25fbe7979396d83da3bef91353894e0862bb7897">25fbe79793</a></td><td>Corey Farrell</td><td>stasis_state: Create internal stasis_state_proxy object.</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4b47d4774dd75fda72cb2a8ebc39624b2e575a3c">4b47d4774d</a></td><td>Alexei Gradinari</td><td>res_pjsip_pubsub: add endpoint to some warning</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d223419bcddaf0a695ee90fdb70d4f8fd721090e">d223419bcd</a></td><td>Jonathan Rose</td><td>basic-pbx: Bring forward queue configuration from 13</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8269fcbf0343b7515df2073cd8b847ce4b24aa6b">8269fcbf03</a></td><td>Ben Ford</td><td>taskprocessor.c: Added "like" support to 'core show taskprocessors'</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=37139e16a554e99b5cc8457681cf04e20c585fb4">37139e16a5</a></td><td>Asterisk Development Team</td><td>Update CHANGES and UPGRADE.txt for 17.0.0-rc2</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7550a82fe0cfbcf1309bb410293c8086a13a8bd2">7550a82fe0</a></td><td>Sean Bright</td><td>res_musiconhold: Add new 'playlist' mode</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=175a7ccac76529b8f238f2ad61b8cd4cba0c0d10">175a7ccac7</a></td><td>Kevin Harwell</td><td>res_pjsip_pubsub: change warning to debug</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5bedd4a9b45bb2d7ded09ad6d0b3561e2d091711">5bedd4a9b4</a></td><td>Corey Farrell</td><td>core: Fix ABI mismatch of ao2_global_obj.</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f181dfc0657cfe8e2558eed68665a9ce8f0d4a54">f181dfc065</a></td><td>Ben Ford</td><td>taskprocessor.c: Add CLI commands to reset taskprocessor stats.</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ec712cfab24552406a7e104bd8ab02201f78c233">ec712cfab2</a></td><td>Corey Farrell</td><td>core: Add AO2_ALLOC_OPT_NO_REF_DEBUG option.</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b1c5753b78344d2fc6b4d209116dea1f50d76bcd">b1c5753b78</a></td><td>George Joseph</td><td>astmm.c:  Display backtrace with memory show allocations</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=af0ccbc45e62efc92bccabcd7e19593ae7b4955b">af0ccbc45e</a></td><td>Corey Farrell</td><td>stasis: refcounter.py can incorrectly report skewed objects.</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fd7fc3d7a5fcabb9dbf925fdcdf51b4fe50ece82">fd7fc3d7a5</a></td><td>Corey Farrell</td><td>stasis: Fix leaks</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=78f329dad9a84312909cf517945ae1fe71b801d9">78f329dad9</a></td><td>Corey Farrell</td><td>app_voicemail: Fix module unload leak.</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1d960195c2b28d6c78794f699b07dc6f54460d2f">1d960195c2</a></td><td>Ben Ford</td><td>res_rtp_asterisk.c: Send RTCP as compound packets.</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5dad4f4ea1bce2e823ed72e075b51d637673670f">5dad4f4ea1</a></td><td>Ben Ford</td><td>res_rtp: Add unit tests for RTCP stats.</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=44068cc6bf802742ce8a0283351781bbcd9d7690">44068cc6bf</a></td><td>George Joseph</td><td>ARI: External Media</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c060ef7dee4bdd56504478f3ab74ae56646e2b6f">c060ef7dee</a></td><td>George Joseph</td><td>chan_sip:  Update links referenced in deprecation notice</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7d99710fa683fbceec1ff12f7d52ecd5549f08e0">7d99710fa6</a></td><td>Chris-Savinovich</td><td>test_utils.c: Skip test adsi_loaded_test if module not loaded.</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=95b3c9839da7a4cc01feeeab2b65350bc8e0ce40">95b3c9839d</a></td><td>Igor Goncharovsky</td><td>chan_unistim: Fix code, causing all incoming DTMF sent back to asterisk</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e7f1f7eefd13c535ed90b30fa6122dbe43f076a4">e7f1f7eefd</a></td><td>Igor Goncharovsky</td><td>chan_unistim: Fix RTP port byte order for big-endian arch</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a0b0c7540c2d9623ccbe73390f95fd3f2573eb3e">a0b0c7540c</a></td><td>Alexei Gradinari</td><td>Fix misname 'res_external_mwi' to 'res_mwi_external' in comments.</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=581ffdf6a98c53ec41d640161420bdb07f4d3d9b">581ffdf6a9</a></td><td>George Joseph</td><td>chan_rtp:  Accept hostname as well as ip address as destination</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b3d9f6ef6cf7541676631909f3852c6b1b5030fb">b3d9f6ef6c</a></td><td>George Joseph</td><td>dns_core:  Create new API ast_dns_resolve_ipv6_and_ipv4</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f6ab42df2641ad8e32849d7bb608bd35f4fa313e">f6ab42df26</a></td><td>George Joseph</td><td>res_ari.c:  Prefer exact handler match over wildcard</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e123d1ccc581d319d5efbe647920b738f1af6505">e123d1ccc5</a></td><td>Sean Bright</td><td>audiohook.c: Substitute silence for unavailable audio frames</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5489ad5574331c909075539dc9ccaef4acebb477">5489ad5574</a></td><td>George Joseph</td><td>CI: Escape backslashes in printenv/sort/tr</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=aa44e723c56f65b363955f5c945b9be97ed545cc">aa44e723c5</a></td><td>George Joseph</td><td>CI:  Add "throttle" label and "skip_gate" capability</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=865214079447f987e3a3d1b1a31cea2524547614">8652140794</a></td><td>George Joseph</td><td>CI:  Make node labels job-specific</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=87f8ca2812b45e2c1eb8f5897b35575d2ded4e23">87f8ca2812</a></td><td>Sean Bright</td><td>app_voicemail: Remove extra menuselect build options</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1976db6ffd912dfb9ac0bc42748ae9c687171c7a">1976db6ffd</a></td><td>Sean Bright</td><td>res_musiconhold: Use a vector instead of custom array allocation</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c748187bbfc693cce2ffbdb80cdf8f0310d0ad76">c748187bbf</a></td><td>Sean Bright</td><td>manager: Send fewer packets</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f1e0c9d1b819ccda353412e6cf8e29ff273f0145">f1e0c9d1b8</a></td><td>Asterisk Development Team</td><td>Update CHANGES and UPGRADE.txt for 17.0.0</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4f2ad5a55d5d44550e21a817c35c46a6baa7f916">4f2ad5a55d</a></td><td>George Joseph</td><td>doc:  Add "master-only" flag back to the CHANGES and UPGRADE files</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a66ccb90ac7175b33dea0590aa9482e8afb6078b">a66ccb90ac</a></td><td>Sean Bright</td><td>res_musiconhold: Use ast_pipe_nonblock() wrapper</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e821a637a84c7ee2e6549c44c2f1ca6637daae18">e821a637a8</a></td><td>George Joseph</td><td>loader.c:  Fix possible SEGV when a module fails to register</td></tr>
+</table><hr><a name="diffstat"><h2 align="center">Diffstat Results</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a summary of the changes to the source code that went into this release that was generated using the diffstat utility.</p><pre>asterisk-17.0.0-summary.html                                                | 1265 ----
+asterisk-17.0.0-summary.txt                                                 | 2973 ----------
+b/.version                                                                  |    2
+b/CHANGES                                                                   |   94
+b/ChangeLog                                                                 | 1876 +++++-
+b/README-SERIOUSLY.bestpractices.md                                         |    4
+b/UPGRADE.txt                                                               |   10
+b/addons/cdr_mysql.c                                                        |   18
+b/addons/chan_ooh323.c                                                      |    7
+b/apps/app_amd.c                                                            |   24
+b/apps/app_cdr.c                                                            |    8
+b/apps/app_chanisavail.c                                                    |    3
+b/apps/app_dictate.c                                                        |    4
+b/apps/app_followme.c                                                       |   10
+b/apps/app_meetme.c                                                         |    2
+b/apps/app_minivm.c                                                         |    3
+b/apps/app_mixmonitor.c                                                     |   13
+b/apps/app_playback.c                                                       |    9
+b/apps/app_queue.c                                                          |    6
+b/apps/app_readexten.c                                                      |    3
+b/apps/app_senddtmf.c                                                       |   13
+b/apps/app_voicemail.c                                                      |  217
+b/asterisk-17.1.0-rc2-summary.html                                          |   28
+b/asterisk-17.1.0-rc2-summary.txt                                           |  133
+b/bridges/bridge_native_rtp.c                                               |    5
+b/bridges/bridge_softmix.c                                                  |    7
+b/cdr/cdr_pgsql.c                                                           |    2
+b/cel/cel_pgsql.c                                                           |    2
+b/channels/chan_dahdi.c                                                     |   35
+b/channels/chan_dahdi.h                                                     |   18
+b/channels/chan_iax2.c                                                      |   16
+b/channels/chan_motif.c                                                     |    9
+b/channels/chan_pjsip.c                                                     |   24
+b/channels/chan_rtp.c                                                       |   19
+b/channels/chan_sip.c                                                       |   39
+b/channels/chan_unistim.c                                                   |  174
+b/channels/pjsip/cli_commands.c                                             |   13
+b/channels/pjsip/dialplan_functions.c                                       |   65
+b/channels/pjsip/include/dialplan_functions.h                               |   25
+b/channels/sig_pri.c                                                        |    2
+b/codecs/Makefile                                                           |    3
+b/codecs/ex_alaw.h                                                          |    5
+b/codecs/ex_g722.h                                                          |    5
+b/codecs/ex_ulaw.h                                                          |    5
+b/codecs/speex/arch.h                                                       |   13
+b/codecs/speex/fixed_generic.h                                              |    4
+b/codecs/speex/resample.c                                                   |  332 -
+b/codecs/speex/speex_resampler.h                                            |    4
+b/configs/basic-pbx/extensions.conf                                         |   14
+b/configs/basic-pbx/modules.conf                                            |    1
+b/configs/basic-pbx/queues.conf                                             |   19
+b/configs/samples/extconfig.conf.sample                                     |    1
+b/configs/samples/musiconhold.conf.sample                                   |   23
+b/configure                                                                 |   82
+b/configure.ac                                                              |   20
+b/contrib/ast-db-manage/config/versions/fbb7766f17bc_add_playlist_to_moh.py |   54
+b/contrib/realtime/mysql/mysql_config.sql                                   |   15
+b/contrib/realtime/postgresql/postgresql_config.sql                         |   21
+b/formats/format_g726.c                                                     |   16
+b/formats/msgsm.h                                                           |    4
+b/funcs/func_curl.c                                                         |   37
+b/funcs/func_env.c                                                          |    5
+b/funcs/func_jitterbuffer.c                                                 |   19
+b/include/asterisk/abstract_jb.h                                            |    4
+b/include/asterisk/ari.h                                                    |    2
+b/include/asterisk/astobj2.h                                                |    5
+b/include/asterisk/audiohook.h                                              |    2
+b/include/asterisk/autoconfig.h.in                                          |    6
+b/include/asterisk/calendar.h                                               |    4
+b/include/asterisk/channel_internal.h                                       |    5
+b/include/asterisk/config.h                                                 |   18
+b/include/asterisk/config_options.h                                         |    2
+b/include/asterisk/dns_core.h                                               |   22
+b/include/asterisk/dns_internal.h                                           |    5
+b/include/asterisk/format_cache.h                                           |    5
+b/include/asterisk/max_forwards.h                                           |    1
+b/include/asterisk/mixmonitor.h                                             |    5
+b/include/asterisk/parking.h                                                |    5
+b/include/asterisk/res_pjsip.h                                              |    5
+b/include/asterisk/res_pjsip_presence_xml.h                                 |    5
+b/include/asterisk/res_pjsip_session.h                                      |    2
+b/include/asterisk/rtp_engine.h                                             |  111
+b/include/asterisk/serializer.h                                             |   85
+b/include/asterisk/slin.h                                                   |    5
+b/include/asterisk/taskprocessor.h                                          |    9
+b/include/asterisk/utils.h                                                  |    9
+b/main/abstract_jb.c                                                        |  178
+b/main/app.c                                                                |    9
+b/main/asterisk.c                                                           |   15
+b/main/astmm.c                                                              |   23
+b/main/astobj2.c                                                            |   88
+b/main/astobj2_container.c                                                  |   24
+b/main/astobj2_global.c                                                     |   97
+b/main/astobj2_hash.c                                                       |   21
+b/main/astobj2_rbtree.c                                                     |   13
+b/main/audiohook.c                                                          |   11
+b/main/channel.c                                                            |  105
+b/main/codec_builtin.c                                                      |    8
+b/main/config.c                                                             |   16
+b/main/dns_core.c                                                           |   72
+b/main/event.c                                                              |   17
+b/main/file.c                                                               |   37
+b/main/format_cache.c                                                       |    8
+b/main/indications.c                                                        |    6
+b/main/manager.c                                                            |   92
+b/main/media_cache.c                                                        |   47
+b/main/pbx.c                                                                |    5
+b/main/pbx_variables.c                                                      |   23
+b/main/rtp_engine.c                                                         |  137
+b/main/serializer.c                                                         |  189
+b/main/stasis.c                                                             |   22
+b/main/stasis_cache.c                                                       |   10
+b/main/stasis_state.c                                                       |  298 -
+b/main/taskprocessor.c                                                      |  219
+b/res/ari/config.c                                                          |   10
+b/res/ari/resource_channels.c                                               |  158
+b/res/ari/resource_channels.h                                               |   42
+b/res/ari/resource_events.c                                                 |   10
+b/res/parking/parking_bridge.c                                              |   36
+b/res/parking/parking_bridge_features.c                                     |    2
+b/res/parking/res_parking.h                                                 |    5
+b/res/res_ari_channels.c                                                    |  135
+b/res/res_ari_events.c                                                      |    2
+b/res/res_calendar_ews.c                                                    |    1
+b/res/res_calendar_exchange.c                                               |    1
+b/res/res_calendar_icalendar.c                                              |    1
+b/res/res_config_curl.c                                                     |    5
+b/res/res_config_pgsql.c                                                    |    2
+b/res/res_musiconhold.c                                                     |  294
+b/res/res_phoneprov.c                                                       |    6
+b/res/res_pjsip.c                                                           |   83
+b/res/res_pjsip/config_system.c                                             |    2
+b/res/res_pjsip/config_transport.c                                          |   17
+b/res/res_pjsip/pjsip_configuration.c                                       |    4
+b/res/res_pjsip/pjsip_resolver.c                                            |    4
+b/res/res_pjsip_endpoint_identifier_ip.c                                    |   18
+b/res/res_pjsip_mwi.c                                                       |  339 -
+b/res/res_pjsip_outbound_registration.c                                     |   17
+b/res/res_pjsip_pubsub.c                                                    |   20
+b/res/res_pjsip_registrar.c                                                 |   55
+b/res/res_pjsip_session.c                                                   |   35
+b/res/res_pjsip_t38.c                                                       |   40
+b/res/res_pjsip_transport_websocket.c                                       |    2
+b/res/res_resolver_unbound.c                                                |    6
+b/res/res_rtp_asterisk.c                                                    |  297
+b/res/res_stasis.c                                                          |   43
+b/res/stasis/command.c                                                      |    2
+b/res/stasis/control.c                                                      |    2
+b/rest-api-templates/res_ari_resource.c.mustache                            |    2
+b/rest-api/api-docs/channels.json                                           |  125
+b/tests/CI/gates.jenkinsfile                                                |   12
+b/tests/CI/periodics-daily.jenkinsfile                                      |   11
+b/tests/CI/ref_debug.jenkinsfile                                            |    9
+b/tests/CI/unittests.jenkinsfile                                            |    9
+b/tests/test_data_buffer.c                                                  |    2
+b/tests/test_res_rtp.c                                                      |  516 +
+b/tests/test_taskprocessor.c                                                |    2
+157 files changed, 6536 insertions(+), 5842 deletions(-)</pre><br></html>

asterisk-17.1.0-summary.txt

@@ -0,0 +1,962 @@
+                                Release Summary
+
+                                asterisk-17.1.0
+
+                                Date: 2019-12-23
+
+                           <asteriskteam@digium.com>
+
+     ----------------------------------------------------------------------
+
+                               Table of Contents
+
+    1. Summary
+    2. Contributors
+    3. Closed Issues
+    4. Other Changes
+    5. Diffstat
+
+     ----------------------------------------------------------------------
+
+                                    Summary
+
+                                 [Back to Top]
+
+   This release is a point release of an existing major version. The changes
+   included were made to address problems that have been identified in this
+   release series, or are minor, backwards compatible new features or
+   improvements. Users should be able to safely upgrade to this version if
+   this release series is already in use. Users considering upgrading from a
+   previous version are strongly encouraged to review the UPGRADE.txt
+   document as well as the CHANGES document for information about upgrading
+   to this release series.
+
+   The data in this summary reflects changes that have been made since the
+   previous release, asterisk-17.0.0.
+
+     ----------------------------------------------------------------------
+
+                                  Contributors
+
+                                 [Back to Top]
+
+   This table lists the people who have submitted code, those that have
+   tested patches, as well as those that reported issues on the issue tracker
+   that were resolved in this release. For coders, the number is how many of
+   their patches (of any size) were committed into this release. For testers,
+   the number is the number of times their name was listed as assisting with
+   testing a patch. Finally, for reporters, the number is the number of
+   issues that they reported that were affected by commits that went into
+   this release.
+
+   Coders                     Testers               Reporters                 
+   23 George Joseph           1 tests/test_utils.c. 5 Kevin Harwell           
+   23 Sean Bright                                   5 Joshua C. Colp          
+   16 Joshua Colp                                   4 Frederic LE FOLL        
+   13 Kevin Harwell                                 3 Salah Ahmed             
+   6 Corey Farrell                                  3 Ross Beer               
+   5 Asterisk Development                           2 Alexei Gradinari        
+   Team                                             2 Torrey Searle           
+   5 Alexei Gradinari                               2 Ross Beer               
+   5 Ben Ford                                       2 Joshua Elson            
+   4 Frederic LE FOLL                               2 George Joseph           
+   3 Igor Goncharovsky                              2 Ruddy G                 
+   2 Salah Ahmed                                    1 Michael Cargile         
+   2 Joshua C. Colp                                 1 Walter Doekes           
+   2 Torrey Searle                                  1 Ted G                   
+   2 lvl                                            1 Martin Tomec            
+   1 Thomas Arimont (license                        1 Chris Savinovich        
+   5525)                                            1 Byron Clark             
+   1 Pascal Cadotte Michaud                         1 Niklas Larsson          
+   1 Martin Tomec                                   1 Jonas Swiatek           
+   1 Walter Doekes                                  1 Yoooooo Ha              
+   1 Stas Kobzar                                    1 Michael                 
+   1 Jonathan Rose                                  1 Eliel Sardañons        
+   1 Michael Goryainov                              1 Guido Falsi             
+   1 Chris-Savinovich                               1 Gregory Massel          
+   1 Michael Cargile                                1 Dan Cropp               
+   1 Chris Savinovich                               1 Jeremiah Gadd           
+   1 sungtae kim                                    1 Bernhard Schmidt        
+   1 Florian Floimair                               1 Stas Kobzar             
+   1 cmaj                                           1 nappsoft                
+   1 Christoph Moench-Tegeder                       1 Pascal Cadotte Michaud  
+   1 Dan Cropp                                      1 Kilburn                 
+   1 Guido Falsi                                    1 Ian Jones               
+                                                    1 Michael Goryainov       
+                                                    1 Bernhard Schmidt        
+                                                    1 Ted G                   
+                                                    1 Marian Piater           
+                                                    1 Aheliotech              
+                                                    1 Dan Cropp               
+                                                    1 Mark                    
+                                                    1 Alexander Traud         
+                                                    1 Speed Dial Dave         
+                                                    1 Timothy Vanderaerden    
+                                                    1 Niklas Larsson          
+                                                    1 Andrey V. T.            
+                                                    1 Christoph               
+                                                    Moench-Tegeder            
+                                                    1 Florian Floimair        
+                                                    1 Jonathan Harris         
+                                                    1 Daniel                  
+                                                    1 Sam Banks               
+                                                    1 Eliel Sardañons        
+                                                    1 Alexander Traud         
+                                                    1 Cyril Ramière          
+                                                    1 Jørgen H               
+                                                    1 cmaj                    
+                                                    1 Juan Martin             
+                                                    1 lvl                     
+
+     ----------------------------------------------------------------------
+
+                                 Closed Issues
+
+                                 [Back to Top]
+
+   This is a list of all issues from the issue tracker that were closed by
+   changes that went into this release.
+
+  Security
+
+    Category: Channels/chan_sip/General
+
+   ASTERISK-28589: chan_sip: Depending on configuration an INVITE can alter
+   Addr of a peer
+   Reported by: Andrey V. T.
+     * [665a94cb76] Ben Ford -- chan_sip.c: Prevent address change on
+       unauthenticated SIP request.
+
+    Category: Core/ManagerInterface
+
+   ASTERISK-28580: Bypass SYSTEM write permission in manager action allows
+   system commands execution
+   Reported by: Eliel Sardañons
+     * [6b1ba58967] George Joseph -- manager.c: Prevent the Originate action
+       from running the Originate app
+
+    Category: Resources/res_pjsip_t38
+
+   ASTERISK-28495: res_pjsip_t38: 200 OK with SDP answer with declined stream
+   causes crash
+   Reported by: Alexei Gradinari
+     * [9d4f1e8ebe] Alexei Gradinari -- AST-2019-004 - res_pjsip_t38.c: Add
+       NULL checks before using session media
+
+  Improvement
+
+    Category: Applications/app_voicemail
+
+   ASTERISK-28567: Problem with ASTERISK-20207: Asterisk should clear out any
+   .lock files in the voice mail directory on startup.
+   Reported by: Michael
+     * [b903994987] Sean Bright -- Revert "app_voicemail: Cleanup stale lock
+       files on module load"
+
+    Category: Applications/app_voicemail/ODBC
+
+   ASTERISK-22192: [patch] Allow voicemail forwards with ODBC backend when
+   format differs from attachfmt column
+   Reported by: cmaj
+     * [aa0973f868] cmaj -- app_voicemail.c: Support multiple file formats
+       for forwarded messages.
+
+    Category: Core/CodecInterface
+
+   ASTERISK-28512: Add pass-through support for H.265 (HEVC) codec
+   Reported by: Florian Floimair
+     * [d7a3e4f5cf] Florian Floimair -- core: Add H.265/HEVC passthrough
+       support
+
+    Category: Documentation
+
+   ASTERISK-28586: Typo in README-SERIOUSLY.bestpractices.md
+   Reported by: Sam Banks
+     * [4bc1c170cd] Sean Bright -- README-SERIOUSLY.bestpractices.md: Speling
+       correetions.
+
+    Category: Resources/res_pjsip
+
+   ASTERISK-28542: [patch] add the ability for asterisk to generate on-hold
+   re-invites
+   Reported by: Torrey Searle
+     * [55b760d762] Torrey Searle -- channel/chan_pjsip: add dialplan
+       function for music on hold
+
+    Category: Resources/res_pjsip_outbound_registration
+
+   ASTERISK-28602: res_pjsip_outbound_registration: Maximum retries reached
+   Reported by: Daniel
+     * [eea2d499f4] Joshua Colp -- res_pjsip_outbound_registration: Extend
+       documentation for "max_retries".
+
+  Bug
+
+    Category: .Release/Targets
+
+   ASTERISK-28488: pjsip mwi: n+1 sip notify's sent on re-register
+   Reported by: Chris Savinovich
+     * [a36fb473fe] Kevin Harwell -- res_pjsip_mwi: add better handling of
+       solicited vs unsolicited subscriptions
+
+    Category: Applications/app_amd
+
+   ASTERISK-28608: app_amd: Use time calculation to calculate timeout
+   Reported by: Michael Cargile
+     * [e23b2856d0] Michael Cargile -- app_amd: Fixed timeout issue
+
+    Category: Applications/app_chanisavail
+
+   ASTERISK-28527: ChanIsAvail() creates a CDR if unanswered=yes is set in
+   cdr.conf
+   Reported by: Frederic LE FOLL
+     * [50997de887] Frederic LE FOLL -- ChanIsAvail() generates a CDR when
+       unanswered=yes in cdr.conf.
+
+    Category: Applications/app_meetme
+
+   ASTERISK-28604: app_meetme, chan_ooh323 and cdr_mysql don't build on
+   17.0.0
+   Reported by: George Joseph
+     * [3d29b06e37] Joshua C. Colp -- configure: Add check for MySQL client
+       bool and my_bool type usage.
+     * [36b28c98dd] George Joseph -- Build: Fix compile issues with seldom
+       used modules
+
+    Category: Applications/app_queue
+
+   ASTERISK-28644: Stale comment in app_queue about ring_entry exception
+   Reported by: Walter Doekes
+     * [e1eb5e8dc2] Walter Doekes -- app_queue: Fix old confusing comment
+       about when the members are called
+
+    Category: Applications/app_voicemail/IMAP
+
+   ASTERISK-28505: app_voicemail/IMAP: segfault in leave_voicemail because
+   not checking mailstream
+   Reported by: Alexei Gradinari
+     * [052ab9d966] Alexei Gradinari -- app_voicemail/IMAP: check mailstream
+       not NULL in leave_voicemail
+
+    Category: Bridges/bridge_native_rtp
+
+   ASTERISK-28637: chan_sip+native_bridge_rtp: directmedia compatibility
+   check failure when negociated ptime is not default ptime.
+   Reported by: Frederic LE FOLL
+     * [3e73893e53] Frederic LE FOLL -- chan_sip+native_bridge_rtp: no
+       directmedia for ptime other than default ptime.
+
+    Category: Bridges/bridge_softmix
+
+   ASTERISK-28618: bridge_softmix: hold not cleared when joining a softmix
+   bridge
+   Reported by: Kevin Harwell
+     * [8b4610acfe] Kevin Harwell -- bridge_softmix: clear hold when joining
+       a softmix bridge
+
+    Category: CDR/General
+
+   ASTERISK-28566: CDR backend unload problem during active call(s)
+   Reported by: Marian Piater
+     * [1dc3451a34] Sean Bright -- cdr_mysql: Don't clean up on unload unless
+       we can unregister from CDRs
+
+    Category: CDR/cdr_pgsql
+
+   ASTERISK-28571: cdr_pgsql: accesses obsolete (and finally removed) column
+   Reported by: Christoph Moench-Tegeder
+     * [79cc8ae3b8] Christoph Moench-Tegeder -- cdr_pgsql cel_pgsql
+       res_config_pgsql: compatibility with PostgreSQL 12
+
+    Category: Channels/chan_dahdi
+
+   ASTERISK-28615: chan_dahdi: PRI span status may stay "Down, Active" after
+   a short alarm
+   Reported by: Frederic LE FOLL
+     * [d3dd4c5459] Frederic LE FOLL -- chan_dahdi: PRI span status may stay
+       "Down, Active" after a short alarm
+   ASTERISK-28536: Asterisk release candidates fail to build on FreeBSD
+   Reported by: Guido Falsi
+     * [5ff2f7a016] Guido Falsi -- chan_dahdi: Fix build with clang/llvm
+   ASTERISK-28525: chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
+   hangs up
+   Reported by: Frederic LE FOLL
+     * [9e67c92502] Frederic LE FOLL -- chan_dahdi: set CHANNEL(hangupsource)
+       when a PRI channel hangs up
+
+    Category: Channels/chan_pjsip
+
+   ASTERISK-28578: race condition on pjsip channelstats command
+   Reported by: Salah Ahmed
+     * [40acd7d198] Salah Ahmed -- Crash during "pjsip show channelstats"
+       execution
+   ASTERISK-28561: Asterisk Deadlocks
+   Reported by: Aheliotech
+     * [ae761c7473] Joshua Colp -- pbx: deadlock when outgoing dialed channel
+       hangs up too quickly
+   ASTERISK-28086: chan_pjsip: Crash when initiating PlayDTMF over AMI
+   Reported by: Jeremiah Gadd
+     * [71f86e78b6] lvl -- chan_pjsip: Prevent segfault when running PlayDTMF
+       on hungup channel
+   ASTERISK-28538: chan_pjsip: Deadlock on fax detection
+   Reported by: Joshua C. Colp
+     * [4d1baa3ae8] Joshua Colp -- chan_pjsip: Relock correct channel during
+       "fax" redirect.
+
+    Category: Channels/chan_sip/General
+
+   ASTERISK-28637: chan_sip+native_bridge_rtp: directmedia compatibility
+   check failure when negociated ptime is not default ptime.
+   Reported by: Frederic LE FOLL
+     * [3e73893e53] Frederic LE FOLL -- chan_sip+native_bridge_rtp: no
+       directmedia for ptime other than default ptime.
+
+    Category: Channels/chan_unistim
+
+   ASTERISK-25592: chan_unistim: Clang Warning: variable sized type not at
+   end of a struct
+   Reported by: Alexander Traud
+     * [056ddf76ce] Igor Goncharovsky -- chan_unistim: Fix clang warning:
+       variable sized type not at end of a struct
+
+    Category: Codecs/codec_resample
+
+   ASTERISK-28511: codec_resample: Bad sound quality when up sampling from
+   SLIN16 to SLIN32
+   Reported by: Ruddy G
+     * [75d6418d8e] Sean Bright -- codec_resample: Ensure OUTSIDE_SPEEX is
+       defined when necessary
+     * [34ab9964f5] Sean Bright -- codec_resample: Upgrade speex_resample to
+       fix up-sampling bug
+
+    Category: Core/BuildSystem
+
+   ASTERISK-28487: compile menuselect on gentoo
+   Reported by: Kilburn
+     * [a5f05eed70] Sean Bright -- menuselect: Fix curses build on Gentoo
+       Linux
+
+    Category: Core/Channels
+
+   ASTERISK-28499: translate: Crash when frame does not have a "src" field
+   set
+   Reported by: Gregory Massel
+     * [61c01df560] Joshua Colp -- AST-2019-005 - translate: Don't assume all
+       frames will have a src.
+
+    Category: Core/Configuration
+
+   ASTERISK-23756: setvar directive when used in template and a child of said
+   template, results in duplicate variable names
+   Reported by: Michael Goryainov
+     * [2fa296e7d4] Michael Goryainov -- channels: Allow updating variable
+       value
+
+    Category: Core/General
+
+   ASTERISK-28498: cel / cdr: Event times may be incorrect
+   Reported by: Joshua C. Colp
+     * [108b1abbd9] Joshua Colp -- cdr / cel: Use event time at event
+       creation instead of processing.
+
+    Category: Core/RTP
+
+   ASTERISK-28480: json integer overflow in ssrc and timestamp
+   Reported by: Salah Ahmed
+     * [a305f2fdcb] Kevin Harwell -- various modules: json integer overflow
+
+    Category: Core/Stasis
+
+   ASTERISK-28553: stasis.c: Crash during unload
+   Reported by: Kevin Harwell
+     * [57fa604571] Joshua Colp -- stasis: Pass bumped topic_all reference to
+       proxy_dtor.
+
+    Category: Core/UDPTL
+
+   ASTERISK-28483: packet lost on UDPTL wrap around
+   Reported by: Torrey Searle
+     * [44af3e9018] Torrey Searle -- main/udptl.c: correctly handle udptl
+       sequence wrap around
+
+    Category: Functions/General
+
+   ASTERISK-28626: Missing arguments in PJSIP_CONTACT function documentation
+   Reported by: Pascal Cadotte Michaud
+     * [450173a0ae] Pascal Cadotte Michaud -- PJSIP_CONTACT: add missing
+       argument documentation
+   ASTERISK-26481: FILE function grabs garbage along with read data when
+   target line has no newline
+   Reported by: Jonathan Harris
+     * [92bb381d5d] Sean Bright -- func_env: Prevent FILE() from reading
+       garbage at end-of-file
+
+    Category: General
+
+   ASTERISK-28609: Memory Leak in res_rtp_asterisk.c
+   Reported by: Ted G
+     * [27a4a3c761] George Joseph -- res_rtp_asterisk: Add frame list
+       cleanups to ast_rtp_read
+   ASTERISK-28590: utils.c throws repeated warnings;
+   "pthread_attr_setstacksize: Invalid argument"
+   Reported by: Speed Dial Dave
+     * [b3c56c7fa5] Sean Bright -- utils.h: Set lower bound for thread stack
+       size to PTHREAD_STACK_MIN
+   ASTERISK-28523: Asterisk 16.5.0 Memory leak
+   Reported by: Cyril Ramière
+     * [f821e81071] Kevin Harwell -- res_sorcery_memory_cache: stale item
+       update leak
+   ASTERISK-28472: Asterisk occasionally passes a NULL as srtp->session to
+   srtp_protect/unprotect causing SEGV
+   Reported by: Jonas Swiatek
+     * [5daa9bbaee] Kevin Harwell -- srtp: Fix possible race condition, and
+       add NULL checks
+
+    Category: PBX/pbx_config
+
+   ASTERISK-28534: Segmentation fault when there is no priority for an
+   extension
+   Reported by: Timothy Vanderaerden
+     * [8d0edf2b37] Sean Bright -- pbx: Prevent Realtime switch crash on
+       invalid priority
+
+    Category: Resources/res_ari
+
+   ASTERISK-28585: ari/resource_events: Crash in event session cleanup
+   Reported by: Kevin Harwell
+     * [e37d546109] Joshua Colp -- res_ari_events: Add module reference when
+       a WebSocket is open.
+
+    Category: Resources/res_calendar_exchange
+
+   ASTERISK-28572: Memory leaks in res_calendar_exchange and
+   res_calendar_icalendar
+   Reported by: Yoooooo Ha
+     * [b3792e1288] Sean Bright -- res_calendar: Resolve memory leak on
+       calendar destruction
+
+    Category: Resources/res_calendar_icalendar
+
+   ASTERISK-28572: Memory leaks in res_calendar_exchange and
+   res_calendar_icalendar
+   Reported by: Yoooooo Ha
+     * [b3792e1288] Sean Bright -- res_calendar: Resolve memory leak on
+       calendar destruction
+
+    Category: Resources/res_parking
+
+   ASTERISK-28631: res_parking: Doesn't park when parkee and parker are the
+   same
+   Reported by: Ross Beer
+     * [41d58a4ce2] Joshua Colp -- parking: Fall back to parker channel name
+       even if it matches parkee.
+   ASTERISK-28616: parking: Deadlock when multi call parking
+   Reported by: Joshua C. Colp
+     * [de433cdcaf] Joshua Colp -- parking: Fix case where we can't get the
+       parker.
+     * [d638d9c6c6] Joshua Colp -- parking: Use channel snapshot instead of
+       channel.
+
+    Category: Resources/res_pjsip
+
+   ASTERISK-28641: res_pjsip Segfaults when realtime configuration to an AOR
+   points to a not existent AOR
+   Reported by: Ross Beer
+     * [4e057eb9d2] Sean Bright -- res_pjsip_registrar.c: Prevent potential
+       double free if AOR is not found
+   ASTERISK-28544: Wrong contact representation in ipv6 mode
+   Reported by: Jørgen H
+     * [6527eb8213] Sean Bright -- res_pjsip_transport_websocket: Don't put
+       brackets around local_name if IPv6
+   ASTERISK-28521: pjsip: Memory Leak
+   Reported by: Mark
+     * [7c0435f854] George Joseph -- pjproject_bundled: Revert pjproject 2.9
+       commits causing leaks
+   ASTERISK-28228: res_pjsip: pjsip show contacts prints double entries
+   Reported by: Ian Jones
+     * [20459d4cac] Joshua Colp -- res_pjsip: Fix multiple of the same
+       contact in "pjsip show contacts".
+
+    Category: Resources/res_pjsip_mwi
+
+   ASTERISK-28575: MWI Send Notify Crash on 16.6
+   Reported by: Joshua Elson
+     * [45c0d99185] Kevin Harwell -- res_pjsip_mwi: potential double unref,
+       and potential unwanted double link
+   ASTERISK-28552: res_pjsip_mwi: Frack during unload on unsolicited_mwi
+   container
+   Reported by: Kevin Harwell
+     * [996fc40e2b] Kevin Harwell -- res_pjsip_mwi: use an ao2_global object
+       for mwi containers
+
+    Category: Resources/res_pjsip_outbound_registration
+
+   ASTERISK-28624: res_pjsip_outbound_registration: add SRV failover
+   Reported by: Kevin Harwell
+     * [8c99930375] Kevin Harwell -- res_pjsip_outbound_registration: add
+       support for SRV failover
+   ASTERISK-28521: pjsip: Memory Leak
+   Reported by: Mark
+     * [7c0435f854] George Joseph -- pjproject_bundled: Revert pjproject 2.9
+       commits causing leaks
+
+    Category: Resources/res_pjsip_path
+
+   ASTERISK-28463: res_pjsip_path: Crash when invalid contact is configured
+   Reported by: Juan Martin
+     * [51cf060c6c] Sean Bright -- res_pjsip_registrar: Validate Contact URI
+       before adding to responses
+
+    Category: Resources/res_pjsip_sdp_rtp
+
+   ASTERISK-28659: res_pjsip_sdp_rtp: Bundle includes non-existent media
+   stream if codecs create additional streams and offer does not have them
+   Reported by: nappsoft
+     * [5949f9a86a] Joshua C. Colp -- res_pjsip_session: Set stream state on
+       created streams for incoming SDP.
+
+    Category: Resources/res_pjsip_session
+
+   ASTERISK-28659: res_pjsip_sdp_rtp: Bundle includes non-existent media
+   stream if codecs create additional streams and offer does not have them
+   Reported by: nappsoft
+     * [5949f9a86a] Joshua C. Colp -- res_pjsip_session: Set stream state on
+       created streams for incoming SDP.
+   ASTERISK-28445: res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
+   hangup when TEST_FRAMEWORK enabled
+   Reported by: Bernhard Schmidt
+     * [4d56adf8fb] Sean Bright -- res_pjsip_session.c: Prevent
+       use-after-free with TEST_FRAMEWORK enabled
+   ASTERISK-28086: chan_pjsip: Crash when initiating PlayDTMF over AMI
+   Reported by: Jeremiah Gadd
+     * [71f86e78b6] lvl -- chan_pjsip: Prevent segfault when running PlayDTMF
+       on hungup channel
+
+    Category: Resources/res_pjsip_t38
+
+   ASTERISK-28621: Enforce T.38 error correction mode at 200 ok received
+   Reported by: Salah Ahmed
+     * [4ac0299bfb] Salah Ahmed -- res_pjsip_t38: T.38 error correction mode
+       selection at 200 ok received
+
+    Category: Resources/res_rtp_asterisk
+
+   ASTERISK-28576: res_rtp_asterisk: ICE Completion Crash when sent packet
+   length doesn't match
+   Reported by: Joshua Elson
+     * [0c486e7edf] Joshua Colp -- res_rtp_asterisk: Always return provided
+       DTLS packet length.
+
+    Category: Resources/res_stasis
+
+   ASTERISK-28423: ARI causes STASIS Deadlock
+   Reported by: Ross Beer
+     * [7202624b3b] George Joseph -- stasis: Don't hold app_registry and
+       session locks unnecessarily
+
+    Category: pjproject/pjsip
+
+   ASTERISK-28574: pjproject fails to build on 16.6.0, works on 16.5
+   Reported by: Niklas Larsson
+     * [2652bda3a0] George Joseph -- pjproject_bundled: Replace earlier
+       reverts with official fixes.
+   ASTERISK-28509: PJSIP cnonce generated on Linux contains 36 characters,
+   NEC only supports up to 32 characters
+   Reported by: Dan Cropp
+     * [a1d38e19a2] Dan Cropp -- pjproject: Configurable setting for cnonce
+       to include hyphens or not
+
+  New Feature
+
+    Category: Applications/app_senddtmf
+
+   ASTERISK-28614: app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
+   of only "sending"
+   Reported by: lvl
+     * [6345a00228] lvl -- app_senddtmf: Add receive mode to AMI Action
+       PlayDTMF
+
+    Category: Core/Jitterbuffer
+
+   ASTERISK-28533: func_jitterbuffer: Add support for video synchronization
+   Reported by: Joshua C. Colp
+     * [926053d7bd] Joshua Colp -- func_jitterbuffer: Add audio/video sync
+       support.
+
+    Category: Functions/func_curl
+
+   ASTERISK-28613: func_curl: CURLOPT cannot set Content-Type header
+   Reported by: Martin Tomec
+     * [d579ec9cdf] Martin Tomec -- func_curl.c: Support custom http headers
+
+    Category: Resources/res_musiconhold
+
+   ASTERISK-17808: [patch] Unregister a realtime moh class
+   Reported by: Byron Clark
+     * [9e26136ee6] sungtae kim -- res_musiconhold: Added unregister realtime
+       moh class
+
+    Category: pjproject/pjsip
+
+   ASTERISK-28489: Channel variable SIPFROMDOMAIN for chan_pjsip to setup
+   From header URI domain
+   Reported by: Stas Kobzar
+     * [3a246c2a69] Stas Kobzar -- res_pjsip: Channel variable SIPFROMDOMAIN
+
+     ----------------------------------------------------------------------
+
+                      Commits Not Associated with an Issue
+
+                                 [Back to Top]
+
+   This is a list of all changes that went into this release that did not
+   reference a JIRA issue.
+
+   +------------------------------------------------------------------------+
+   | Revision   | Author               | Summary                            |
+   |------------+----------------------+------------------------------------|
+   | de2b8ef23a | Asterisk Development | Update for 17.1.0-rc2              |
+   |            | Team                 |                                    |
+   |------------+----------------------+------------------------------------|
+   |            |                      | Revert "chan_dahdi: set            |
+   | a76d3103fb | George Joseph        | CHANNEL(hangupsource) when a PRI   |
+   |            |                      | channel hangs up"                  |
+   |------------+----------------------+------------------------------------|
+   | c8563727d5 | Asterisk Development | Update for 17.1.0-rc1              |
+   |            | Team                 |                                    |
+   |------------+----------------------+------------------------------------|
+   | 08a6e8c553 | Asterisk Development | Update CHANGES and UPGRADE.txt for |
+   |            | Team                 | 17.1.0                             |
+   |------------+----------------------+------------------------------------|
+   | 6a89784b78 | Joshua Colp          | Revert "PJSIP_CONTACT: add missing |
+   |            |                      | argument documentation"            |
+   |------------+----------------------+------------------------------------|
+   |            |                      | res_pjsip_registrar.c: Prevent     |
+   | f26e5bacc0 | Sean Bright          | possible buffer overflow with      |
+   |            |                      | domain aliases                     |
+   |------------+----------------------+------------------------------------|
+   |            |                      | channel.c: Resolve issue with      |
+   | 88150323a2 | Thomas Arimont       | receiving SIP INFO packets for     |
+   |            |                      | DTMF                               |
+   |------------+----------------------+------------------------------------|
+   | 5b15a1c639 | George Joseph        | CI: Turn off shallow cloning       |
+   |            |                      | altogether                         |
+   |------------+----------------------+------------------------------------|
+   | cc59e21409 | Sean Bright          | media_cache.c: Various CLI         |
+   |            |                      | improvements                       |
+   |------------+----------------------+------------------------------------|
+   | 2a92e6b576 | George Joseph        | CI: Fix missing script block in    |
+   |            |                      | jenkinsfiles                       |
+   |------------+----------------------+------------------------------------|
+   | f0d1ce50af | George Joseph        | CI: Fix missing script block in    |
+   |            |                      | jenkinsfiles                       |
+   |------------+----------------------+------------------------------------|
+   | 46dceab33f | George Joseph        | CI: Increase clone depth and do    |
+   |            |                      | better cleanup                     |
+   |------------+----------------------+------------------------------------|
+   | 76ef36fafc | Sean Bright          | res_pjsip_registrar: Fix           |
+   |            |                      | uninitlized variable warning       |
+   |------------+----------------------+------------------------------------|
+   | 649733612d | Alexei Gradinari     | serializer: set high/low alert     |
+   |            |                      | levels on whole pool               |
+   |------------+----------------------+------------------------------------|
+   | 8bc6fa0fbd | Kevin Harwell        | various files - fix some alerts    |
+   |            |                      | raised by lgtm code analysis       |
+   |------------+----------------------+------------------------------------|
+   | ea3daa94c8 | Kevin Harwell        | res_pjsip_session: initialize      |
+   |            |                      | pending's topology to endpoint's   |
+   |------------+----------------------+------------------------------------|
+   |            |                      | ExternalMedia: Change return       |
+   | 2d665091a3 | George Joseph        | object from ExternalMedia to       |
+   |            |                      | Channel                            |
+   |------------+----------------------+------------------------------------|
+   | b8ae799ca9 | Joshua Colp          | res_rtp_asterisk: Remove a log     |
+   |            |                      | message that slipped in.           |
+   |------------+----------------------+------------------------------------|
+   | ba688e6891 | Joshua Colp          | test_res_rtp: Enable FIR and REMB  |
+   |            |                      | nominal tests.                     |
+   |------------+----------------------+------------------------------------|
+   | c84135d2a3 | Chris Savinovich     | test_taskprocessor.c: Fix test     |
+   |            |                      | failure on Ubuntu                  |
+   |------------+----------------------+------------------------------------|
+   | 37ec88c4c8 | Kevin Harwell        | serializer: move/add asterisk      |
+   |            |                      | serializer pool functionality      |
+   |------------+----------------------+------------------------------------|
+   | 299ba78b09 | Kevin Harwell        | res_pjsip/res_pjsip_mwi: use       |
+   |            |                      | centralized serializer pools       |
+   |------------+----------------------+------------------------------------|
+   | 25fbe79793 | Corey Farrell        | stasis_state: Create internal      |
+   |            |                      | stasis_state_proxy object.         |
+   |------------+----------------------+------------------------------------|
+   | 4b47d4774d | Alexei Gradinari     | res_pjsip_pubsub: add endpoint to  |
+   |            |                      | some warning                       |
+   |------------+----------------------+------------------------------------|
+   | d223419bcd | Jonathan Rose        | basic-pbx: Bring forward queue     |
+   |            |                      | configuration from 13              |
+   |------------+----------------------+------------------------------------|
+   |            |                      | taskprocessor.c: Added "like"      |
+   | 8269fcbf03 | Ben Ford             | support to 'core show              |
+   |            |                      | taskprocessors'                    |
+   |------------+----------------------+------------------------------------|
+   | 37139e16a5 | Asterisk Development | Update CHANGES and UPGRADE.txt for |
+   |            | Team                 | 17.0.0-rc2                         |
+   |------------+----------------------+------------------------------------|
+   | 7550a82fe0 | Sean Bright          | res_musiconhold: Add new           |
+   |            |                      | 'playlist' mode                    |
+   |------------+----------------------+------------------------------------|
+   | 175a7ccac7 | Kevin Harwell        | res_pjsip_pubsub: change warning   |
+   |            |                      | to debug                           |
+   |------------+----------------------+------------------------------------|
+   | 5bedd4a9b4 | Corey Farrell        | core: Fix ABI mismatch of          |
+   |            |                      | ao2_global_obj.                    |
+   |------------+----------------------+------------------------------------|
+   | f181dfc065 | Ben Ford             | taskprocessor.c: Add CLI commands  |
+   |            |                      | to reset taskprocessor stats.      |
+   |------------+----------------------+------------------------------------|
+   | ec712cfab2 | Corey Farrell        | core: Add                          |
+   |            |                      | AO2_ALLOC_OPT_NO_REF_DEBUG option. |
+   |------------+----------------------+------------------------------------|
+   | b1c5753b78 | George Joseph        | astmm.c: Display backtrace with    |
+   |            |                      | memory show allocations            |
+   |------------+----------------------+------------------------------------|
+   | af0ccbc45e | Corey Farrell        | stasis: refcounter.py can          |
+   |            |                      | incorrectly report skewed objects. |
+   |------------+----------------------+------------------------------------|
+   | fd7fc3d7a5 | Corey Farrell        | stasis: Fix leaks                  |
+   |------------+----------------------+------------------------------------|
+   | 78f329dad9 | Corey Farrell        | app_voicemail: Fix module unload   |
+   |            |                      | leak.                              |
+   |------------+----------------------+------------------------------------|
+   | 1d960195c2 | Ben Ford             | res_rtp_asterisk.c: Send RTCP as   |
+   |            |                      | compound packets.                  |
+   |------------+----------------------+------------------------------------|
+   | 5dad4f4ea1 | Ben Ford             | res_rtp: Add unit tests for RTCP   |
+   |            |                      | stats.                             |
+   |------------+----------------------+------------------------------------|
+   | 44068cc6bf | George Joseph        | ARI: External Media                |
+   |------------+----------------------+------------------------------------|
+   | c060ef7dee | George Joseph        | chan_sip: Update links referenced  |
+   |            |                      | in deprecation notice              |
+   |------------+----------------------+------------------------------------|
+   |            |                      | test_utils.c: Skip test            |
+   | 7d99710fa6 | Chris-Savinovich     | adsi_loaded_test if module not     |
+   |            |                      | loaded.                            |
+   |------------+----------------------+------------------------------------|
+   |            |                      | chan_unistim: Fix code, causing    |
+   | 95b3c9839d | Igor Goncharovsky    | all incoming DTMF sent back to     |
+   |            |                      | asterisk                           |
+   |------------+----------------------+------------------------------------|
+   | e7f1f7eefd | Igor Goncharovsky    | chan_unistim: Fix RTP port byte    |
+   |            |                      | order for big-endian arch          |
+   |------------+----------------------+------------------------------------|
+   | a0b0c7540c | Alexei Gradinari     | Fix misname 'res_external_mwi' to  |
+   |            |                      | 'res_mwi_external' in comments.    |
+   |------------+----------------------+------------------------------------|
+   | 581ffdf6a9 | George Joseph        | chan_rtp: Accept hostname as well  |
+   |            |                      | as ip address as destination       |
+   |------------+----------------------+------------------------------------|
+   | b3d9f6ef6c | George Joseph        | dns_core: Create new API           |
+   |            |                      | ast_dns_resolve_ipv6_and_ipv4      |
+   |------------+----------------------+------------------------------------|
+   | f6ab42df26 | George Joseph        | res_ari.c: Prefer exact handler    |
+   |            |                      | match over wildcard                |
+   |------------+----------------------+------------------------------------|
+   | e123d1ccc5 | Sean Bright          | audiohook.c: Substitute silence    |
+   |            |                      | for unavailable audio frames       |
+   |------------+----------------------+------------------------------------|
+   | 5489ad5574 | George Joseph        | CI: Escape backslashes in          |
+   |            |                      | printenv/sort/tr                   |
+   |------------+----------------------+------------------------------------|
+   | aa44e723c5 | George Joseph        | CI: Add "throttle" label and       |
+   |            |                      | "skip_gate" capability             |
+   |------------+----------------------+------------------------------------|
+   | 8652140794 | George Joseph        | CI: Make node labels job-specific  |
+   |------------+----------------------+------------------------------------|
+   | 87f8ca2812 | Sean Bright          | app_voicemail: Remove extra        |
+   |            |                      | menuselect build options           |
+   |------------+----------------------+------------------------------------|
+   | 1976db6ffd | Sean Bright          | res_musiconhold: Use a vector      |
+   |            |                      | instead of custom array allocation |
+   |------------+----------------------+------------------------------------|
+   | c748187bbf | Sean Bright          | manager: Send fewer packets        |
+   |------------+----------------------+------------------------------------|
+   | f1e0c9d1b8 | Asterisk Development | Update CHANGES and UPGRADE.txt for |
+   |            | Team                 | 17.0.0                             |
+   |------------+----------------------+------------------------------------|
+   | 4f2ad5a55d | George Joseph        | doc: Add "master-only" flag back   |
+   |            |                      | to the CHANGES and UPGRADE files   |
+   |------------+----------------------+------------------------------------|
+   | a66ccb90ac | Sean Bright          | res_musiconhold: Use               |
+   |            |                      | ast_pipe_nonblock() wrapper        |
+   |------------+----------------------+------------------------------------|
+   | e821a637a8 | George Joseph        | loader.c: Fix possible SEGV when a |
+   |            |                      | module fails to register           |
+   +------------------------------------------------------------------------+
+
+     ----------------------------------------------------------------------
+
+                                Diffstat Results
+
+                                 [Back to Top]
+
+   This is a summary of the changes to the source code that went into this
+   release that was generated using the diffstat utility.
+
+ asterisk-17.0.0-summary.html                                                | 1265 ----
+ asterisk-17.0.0-summary.txt                                                 | 2973 ----------
+ b/.version                                                                  |    2
+ b/CHANGES                                                                   |   94
+ b/ChangeLog                                                                 | 1876 +++++-
+ b/README-SERIOUSLY.bestpractices.md                                         |    4
+ b/UPGRADE.txt                                                               |   10
+ b/addons/cdr_mysql.c                                                        |   18
+ b/addons/chan_ooh323.c                                                      |    7
+ b/apps/app_amd.c                                                            |   24
+ b/apps/app_cdr.c                                                            |    8
+ b/apps/app_chanisavail.c                                                    |    3
+ b/apps/app_dictate.c                                                        |    4
+ b/apps/app_followme.c                                                       |   10
+ b/apps/app_meetme.c                                                         |    2
+ b/apps/app_minivm.c                                                         |    3
+ b/apps/app_mixmonitor.c                                                     |   13
+ b/apps/app_playback.c                                                       |    9
+ b/apps/app_queue.c                                                          |    6
+ b/apps/app_readexten.c                                                      |    3
+ b/apps/app_senddtmf.c                                                       |   13
+ b/apps/app_voicemail.c                                                      |  217
+ b/asterisk-17.1.0-rc2-summary.html                                          |   28
+ b/asterisk-17.1.0-rc2-summary.txt                                           |  133
+ b/bridges/bridge_native_rtp.c                                               |    5
+ b/bridges/bridge_softmix.c                                                  |    7
+ b/cdr/cdr_pgsql.c                                                           |    2
+ b/cel/cel_pgsql.c                                                           |    2
+ b/channels/chan_dahdi.c                                                     |   35
+ b/channels/chan_dahdi.h                                                     |   18
+ b/channels/chan_iax2.c                                                      |   16
+ b/channels/chan_motif.c                                                     |    9
+ b/channels/chan_pjsip.c                                                     |   24
+ b/channels/chan_rtp.c                                                       |   19
+ b/channels/chan_sip.c                                                       |   39
+ b/channels/chan_unistim.c                                                   |  174
+ b/channels/pjsip/cli_commands.c                                             |   13
+ b/channels/pjsip/dialplan_functions.c                                       |   65
+ b/channels/pjsip/include/dialplan_functions.h                               |   25
+ b/channels/sig_pri.c                                                        |    2
+ b/codecs/Makefile                                                           |    3
+ b/codecs/ex_alaw.h                                                          |    5
+ b/codecs/ex_g722.h                                                          |    5
+ b/codecs/ex_ulaw.h                                                          |    5
+ b/codecs/speex/arch.h                                                       |   13
+ b/codecs/speex/fixed_generic.h                                              |    4
+ b/codecs/speex/resample.c                                                   |  332 -
+ b/codecs/speex/speex_resampler.h                                            |    4
+ b/configs/basic-pbx/extensions.conf                                         |   14
+ b/configs/basic-pbx/modules.conf                                            |    1
+ b/configs/basic-pbx/queues.conf                                             |   19
+ b/configs/samples/extconfig.conf.sample                                     |    1
+ b/configs/samples/musiconhold.conf.sample                                   |   23
+ b/configure                                                                 |   82
+ b/configure.ac                                                              |   20
+ b/contrib/ast-db-manage/config/versions/fbb7766f17bc_add_playlist_to_moh.py |   54
+ b/contrib/realtime/mysql/mysql_config.sql                                   |   15
+ b/contrib/realtime/postgresql/postgresql_config.sql                         |   21
+ b/formats/format_g726.c                                                     |   16
+ b/formats/msgsm.h                                                           |    4
+ b/funcs/func_curl.c                                                         |   37
+ b/funcs/func_env.c                                                          |    5
+ b/funcs/func_jitterbuffer.c                                                 |   19
+ b/include/asterisk/abstract_jb.h                                            |    4
+ b/include/asterisk/ari.h                                                    |    2
+ b/include/asterisk/astobj2.h                                                |    5
+ b/include/asterisk/audiohook.h                                              |    2
+ b/include/asterisk/autoconfig.h.in                                          |    6
+ b/include/asterisk/calendar.h                                               |    4
+ b/include/asterisk/channel_internal.h                                       |    5
+ b/include/asterisk/config.h                                                 |   18
+ b/include/asterisk/config_options.h                                         |    2
+ b/include/asterisk/dns_core.h                                               |   22
+ b/include/asterisk/dns_internal.h                                           |    5
+ b/include/asterisk/format_cache.h                                           |    5
+ b/include/asterisk/max_forwards.h                                           |    1
+ b/include/asterisk/mixmonitor.h                                             |    5
+ b/include/asterisk/parking.h                                                |    5
+ b/include/asterisk/res_pjsip.h                                              |    5
+ b/include/asterisk/res_pjsip_presence_xml.h                                 |    5
+ b/include/asterisk/res_pjsip_session.h                                      |    2
+ b/include/asterisk/rtp_engine.h                                             |  111
+ b/include/asterisk/serializer.h                                             |   85
+ b/include/asterisk/slin.h                                                   |    5
+ b/include/asterisk/taskprocessor.h                                          |    9
+ b/include/asterisk/utils.h                                                  |    9
+ b/main/abstract_jb.c                                                        |  178
+ b/main/app.c                                                                |    9
+ b/main/asterisk.c                                                           |   15
+ b/main/astmm.c                                                              |   23
+ b/main/astobj2.c                                                            |   88
+ b/main/astobj2_container.c                                                  |   24
+ b/main/astobj2_global.c                                                     |   97
+ b/main/astobj2_hash.c                                                       |   21
+ b/main/astobj2_rbtree.c                                                     |   13
+ b/main/audiohook.c                                                          |   11
+ b/main/channel.c                                                            |  105
+ b/main/codec_builtin.c                                                      |    8
+ b/main/config.c                                                             |   16
+ b/main/dns_core.c                                                           |   72
+ b/main/event.c                                                              |   17
+ b/main/file.c                                                               |   37
+ b/main/format_cache.c                                                       |    8
+ b/main/indications.c                                                        |    6
+ b/main/manager.c                                                            |   92
+ b/main/media_cache.c                                                        |   47
+ b/main/pbx.c                                                                |    5
+ b/main/pbx_variables.c                                                      |   23
+ b/main/rtp_engine.c                                                         |  137
+ b/main/serializer.c                                                         |  189
+ b/main/stasis.c                                                             |   22
+ b/main/stasis_cache.c                                                       |   10
+ b/main/stasis_state.c                                                       |  298 -
+ b/main/taskprocessor.c                                                      |  219
+ b/res/ari/config.c                                                          |   10
+ b/res/ari/resource_channels.c                                               |  158
+ b/res/ari/resource_channels.h                                               |   42
+ b/res/ari/resource_events.c                                                 |   10
+ b/res/parking/parking_bridge.c                                              |   36
+ b/res/parking/parking_bridge_features.c                                     |    2
+ b/res/parking/res_parking.h                                                 |    5
+ b/res/res_ari_channels.c                                                    |  135
+ b/res/res_ari_events.c                                                      |    2
+ b/res/res_calendar_ews.c                                                    |    1
+ b/res/res_calendar_exchange.c                                               |    1
+ b/res/res_calendar_icalendar.c                                              |    1
+ b/res/res_config_curl.c                                                     |    5
+ b/res/res_config_pgsql.c                                                    |    2
+ b/res/res_musiconhold.c                                                     |  294
+ b/res/res_phoneprov.c                                                       |    6
+ b/res/res_pjsip.c                                                           |   83
+ b/res/res_pjsip/config_system.c                                             |    2
+ b/res/res_pjsip/config_transport.c                                          |   17
+ b/res/res_pjsip/pjsip_configuration.c                                       |    4
+ b/res/res_pjsip/pjsip_resolver.c                                            |    4
+ b/res/res_pjsip_endpoint_identifier_ip.c                                    |   18
+ b/res/res_pjsip_mwi.c                                                       |  339 -
+ b/res/res_pjsip_outbound_registration.c                                     |   17
+ b/res/res_pjsip_pubsub.c                                                    |   20
+ b/res/res_pjsip_registrar.c                                                 |   55
+ b/res/res_pjsip_session.c                                                   |   35
+ b/res/res_pjsip_t38.c                                                       |   40
+ b/res/res_pjsip_transport_websocket.c                                       |    2
+ b/res/res_resolver_unbound.c                                                |    6
+ b/res/res_rtp_asterisk.c                                                    |  297
+ b/res/res_stasis.c                                                          |   43
+ b/res/stasis/command.c                                                      |    2
+ b/res/stasis/control.c                                                      |    2
+ b/rest-api-templates/res_ari_resource.c.mustache                            |    2
+ b/rest-api/api-docs/channels.json                                           |  125
+ b/tests/CI/gates.jenkinsfile                                                |   12
+ b/tests/CI/periodics-daily.jenkinsfile                                      |   11
+ b/tests/CI/ref_debug.jenkinsfile                                            |    9
+ b/tests/CI/unittests.jenkinsfile                                            |    9
+ b/tests/test_data_buffer.c                                                  |    2
+ b/tests/test_res_rtp.c                                                      |  516 +
+ b/tests/test_taskprocessor.c                                                |    2
+ 157 files changed, 6536 insertions(+), 5842 deletions(-)

bridges/bridge_native_rtp.c

@@ -713,6 +713,8 @@ static int native_rtp_bridge_compatible_check(struct ast_bridge *bridge, struct
 				framing_inst0, framing_inst1);
 			return 0;
 		}
+		ast_debug(3, "Symmetric ptimes on the two call legs (%u). May be able to native bridge in RTP\n",
+			framing_inst0);
 	}
 
 	read_ptime0 = ast_format_cap_get_format_framing(cap0, ast_channel_rawreadformat(bc0->chan));
@@ -726,6 +728,9 @@ static int native_rtp_bridge_compatible_check(struct ast_bridge *bridge, struct
 			read_ptime0, write_ptime1, read_ptime1, write_ptime0);
 		return 0;
 	}
+	ast_debug(3, "Bridge '%s': Packetization comparison success between RTP streams (read_ptime0:%d == write_ptime1:%d and read_ptime1:%d == write_ptime0:%d).\n",
+		bridge->uniqueid,
+		read_ptime0, write_ptime1, read_ptime1, write_ptime0);
 
 	return 1;
 }

bridges/bridge_softmix.c

@@ -726,6 +726,13 @@ static int softmix_bridge_join(struct ast_bridge *bridge, struct ast_bridge_chan
 		sfu_topologies_on_join(bridge, bridge_channel);
 	}
 
+	/* Complete any active hold before entering, or transitioning to softmix. */
+	if (ast_channel_hold_state(bridge_channel->chan) == AST_CONTROL_HOLD) {
+		ast_debug(1, "Channel %s simulating UNHOLD for bridge softmix join.\n",
+			ast_channel_name(bridge_channel->chan));
+		ast_indicate(bridge_channel->chan, AST_CONTROL_UNHOLD);
+	}
+
 	softmix_poke_thread(softmix_data);
 	return 0;
 }

cdr/cdr_pgsql.c

@@ -707,7 +707,7 @@ static int config_module(int reload)
 			schemaname = ast_alloca(strlen(tmp_schemaname) * 2 + 1);
 			PQescapeStringConn(conn, schemaname, tmp_schemaname, strlen(tmp_schemaname), NULL);
 
-			snprintf(sqlcmd, sizeof(sqlcmd), "SELECT a.attname, t.typname, a.attlen, a.attnotnull, d.adsrc, a.atttypmod FROM (((pg_catalog.pg_class c INNER JOIN pg_catalog.pg_namespace n ON n.oid = c.relnamespace AND c.relname = '%s' AND n.nspname = %s%s%s) INNER JOIN pg_catalog.pg_attribute a ON (NOT a.attisdropped) AND a.attnum > 0 AND a.attrelid = c.oid) INNER JOIN pg_catalog.pg_type t ON t.oid = a.atttypid) LEFT OUTER JOIN pg_attrdef d ON a.atthasdef AND d.adrelid = a.attrelid AND d.adnum = a.attnum ORDER BY n.nspname, c.relname, attnum",
+			snprintf(sqlcmd, sizeof(sqlcmd), "SELECT a.attname, t.typname, a.attlen, a.attnotnull, pg_catalog.pg_get_expr(d.adbin, d.adrelid) adsrc, a.atttypmod FROM (((pg_catalog.pg_class c INNER JOIN pg_catalog.pg_namespace n ON n.oid = c.relnamespace AND c.relname = '%s' AND n.nspname = %s%s%s) INNER JOIN pg_catalog.pg_attribute a ON (NOT a.attisdropped) AND a.attnum > 0 AND a.attrelid = c.oid) INNER JOIN pg_catalog.pg_type t ON t.oid = a.atttypid) LEFT OUTER JOIN pg_attrdef d ON a.atthasdef AND d.adrelid = a.attrelid AND d.adnum = a.attnum ORDER BY n.nspname, c.relname, attnum",
 				tablename,
 				ast_strlen_zero(schemaname) ? "" : "'", ast_strlen_zero(schemaname) ? "current_schema()" : schemaname, ast_strlen_zero(schemaname) ? "" : "'");
 		} else {

cel/cel_pgsql.c

@@ -597,7 +597,7 @@ static int process_my_load_module(struct ast_config *cfg)
 			PQescapeStringConn(conn, schemaname, schema, lenschema, NULL);
 
 			snprintf(sqlcmd, sizeof(sqlcmd),
-			"SELECT a.attname, t.typname, a.attlen, a.attnotnull, d.adsrc, a.atttypmod "
+			"SELECT a.attname, t.typname, a.attlen, a.attnotnull, pg_catalog.pg_get_expr(d.adbin, d.adrelid) adsrc, a.atttypmod "
 			"FROM (((pg_catalog.pg_class c INNER JOIN pg_catalog.pg_namespace n ON n.oid = c.relnamespace "
 			         "AND c.relname = '%s' AND n.nspname = %s%s%s) "
 			       "INNER JOIN pg_catalog.pg_attribute a ON ("

channels/chan_dahdi.c

@@ -4382,7 +4382,7 @@ static char *alarm2str(int alm)
 static const char *event2str(int event)
 {
 	static char buf[256];
-	if ((event < (ARRAY_LEN(events))) && (event > -1))
+	if ((event > -1) && (event < (ARRAY_LEN(events))) )
 		return events[event];
 	sprintf(buf, "Event %d", event); /* safe */
 	return buf;
@@ -12770,8 +12770,10 @@ static struct dahdi_pvt *mkintf(int channel, const struct dahdi_chan_conf *conf,
 			struct ast_variable *v, *tmpvar;
 			for (v = conf->chan.vars ; v ; v = v->next) {
 				if ((tmpvar = ast_variable_new(v->name, v->value, v->file))) {
-					tmpvar->next = tmp->vars;
-					tmp->vars = tmpvar;
+					if (ast_variable_list_replace(&tmp->vars, tmpvar)) {
+						tmpvar->next = tmp->vars;
+						tmp->vars = tmpvar;
+					}
 				}
 			}
 		}
@@ -15158,10 +15160,12 @@ static void mfcr2_show_links_of(struct ast_cli_args *a, struct r2links *list_hea
 			int channo;
 			int prev_channo;
 			x++;
-			switch (mfcr2->r2master) {
-			case 0L: thread_status = "zero"; break;
-			case AST_PTHREADT_NULL: thread_status = "none"; break;
-			default: thread_status = "created"; break;
+			if (mfcr2->r2master == 0L) {
+				thread_status = "zero";
+			} else if (mfcr2->r2master == AST_PTHREADT_NULL) {
+				thread_status = "none";
+			} else {
+				thread_status = "created";
 			}
 			snprintf(index, sizeof(index), "%d", mfcr2->index);
 			snprintf(live_chans_str, sizeof(live_chans_str), "%d", mfcr2->live_chans);
@@ -15170,7 +15174,7 @@ static void mfcr2_show_links_of(struct ast_cli_args *a, struct r2links *list_hea
 			inside_range = 0;
 			len = 0;
 			/* Prepare nice string in channel_list[] */
-			for (i = 0; i < mfcr2->numchans; i++) {
+			for (i = 0; i < mfcr2->numchans && len < sizeof(channel_list) - 1; i++) {
 				struct dahdi_pvt *p = mfcr2->pvts[i];
 				if (!p) {
 					continue;
@@ -18128,13 +18132,17 @@ static int process_dahdi(struct dahdi_chan_conf *confp, const char *cat, struct
 		} else if (!strcasecmp(v->name, "namedpickupgroup")) {
 			confp->chan.named_pickupgroups = ast_get_namedgroups(v->value);
 		} else if (!strcasecmp(v->name, "setvar")) {
-			char *varname = ast_strdupa(v->value), *varval = NULL;
-			struct ast_variable *tmpvar;
-			if (varname && (varval = strchr(varname, '='))) {
-				*varval++ = '\0';
-				if ((tmpvar = ast_variable_new(varname, varval, ""))) {
-					tmpvar->next = confp->chan.vars;
-					confp->chan.vars = tmpvar;
+			if (v->value) {
+				char *varval = NULL;
+				struct ast_variable *tmpvar;
+				char varname[strlen(v->value) + 1];
+				strcpy(varname, v->value); /* safe */
+				if ((varval = strchr(varname, '='))) {
+					*varval++ = '\0';
+					if ((tmpvar = ast_variable_new(varname, varval, ""))) {
+						tmpvar->next = confp->chan.vars;
+						confp->chan.vars = tmpvar;
+					}
 				}
 			}
 		} else if (!strcasecmp(v->name, "immediate")) {
@@ -19078,9 +19086,12 @@ static int process_dahdi(struct dahdi_chan_conf *confp, const char *cat, struct
 			} else if (!strcasecmp(v->name, "mfcr2_logging")) {
 				openr2_log_level_t tmplevel;
 				char *clevel;
-				char *logval = ast_strdupa(v->value);
+				char *logval;
+				char copy[strlen(v->value) + 1];
+				strcpy(copy, v->value); /* safe */
+				logval = copy;
 				while (logval) {
- 					clevel = strsep(&logval,",");
+					clevel = strsep(&logval,",");
 					if (-1 == (tmplevel = openr2_log_get_level(clevel))) {
 						ast_log(LOG_WARNING, "Ignoring invalid logging level: '%s' at line %d.\n", clevel, v->lineno);
 						continue;

channels/chan_dahdi.h

@@ -692,15 +692,15 @@ struct dahdi_pvt {
 	openr2_calling_party_category_t mfcr2_category;
 	int mfcr2_dnis_index;
 	int mfcr2_ani_index;
-	int mfcr2call:1;
-	int mfcr2_answer_pending:1;
-	int mfcr2_charge_calls:1;
-	int mfcr2_allow_collect_calls:1;
-	int mfcr2_forced_release:1;
-	int mfcr2_dnis_matched:1;
-	int mfcr2_call_accepted:1;
-	int mfcr2_accept_on_offer:1;
-	int mfcr2_progress_sent:1;
+	unsigned int mfcr2call:1;
+	unsigned int mfcr2_answer_pending:1;
+	unsigned int mfcr2_charge_calls:1;
+	unsigned int mfcr2_allow_collect_calls:1;
+	unsigned int mfcr2_forced_release:1;
+	unsigned int mfcr2_dnis_matched:1;
+	unsigned int mfcr2_call_accepted:1;
+	unsigned int mfcr2_accept_on_offer:1;
+	unsigned int mfcr2_progress_sent:1;
 #endif	/* defined(HAVE_OPENR2) */
 	/*! \brief DTMF digit in progress.  0 when no digit in progress. */
 	char begindigit;

channels/chan_iax2.c

@@ -7916,9 +7916,11 @@ static int check_access(int callno, struct ast_sockaddr *addr, struct iax_ies *i
 		/* We found our match (use the first) */
 		/* copy vars */
 		for (v = user->vars ; v ; v = v->next) {
-			if((tmpvar = ast_variable_new(v->name, v->value, v->file))) {
-				tmpvar->next = iaxs[callno]->vars;
-				iaxs[callno]->vars = tmpvar;
+			if ((tmpvar = ast_variable_new(v->name, v->value, v->file))) {
+				if (ast_variable_list_replace(&iaxs[callno]->vars, tmpvar)) {
+					tmpvar->next = iaxs[callno]->vars;
+					iaxs[callno]->vars = tmpvar;
+				}
 			}
 		}
 		/* If a max AUTHREQ restriction is in place, activate it */
@@ -11111,18 +11113,18 @@ static int socket_process_helper(struct iax2_thread *thread)
 						if (iaxs[fr->callno]->pingtime <= peer->maxms) {
 							ast_log(LOG_NOTICE, "Peer '%s' is now REACHABLE! Time: %u\n", peer->name, iaxs[fr->callno]->pingtime);
 							ast_endpoint_set_state(peer->endpoint, AST_ENDPOINT_ONLINE);
-							blob = ast_json_pack("{s: s, s: i}",
+							blob = ast_json_pack("{s: s, s: I}",
 								"peer_status", "Reachable",
-								"time", iaxs[fr->callno]->pingtime);
+								"time", (ast_json_int_t)iaxs[fr->callno]->pingtime);
 							ast_devstate_changed(AST_DEVICE_NOT_INUSE, AST_DEVSTATE_CACHABLE, "IAX2/%s", peer->name); /* Activate notification */
 						}
 					} else if ((peer->historicms > 0) && (peer->historicms <= peer->maxms)) {
 						if (iaxs[fr->callno]->pingtime > peer->maxms) {
 							ast_log(LOG_NOTICE, "Peer '%s' is now TOO LAGGED (%u ms)!\n", peer->name, iaxs[fr->callno]->pingtime);
 							ast_endpoint_set_state(peer->endpoint, AST_ENDPOINT_ONLINE);
-							blob = ast_json_pack("{s: s, s: i}",
+							blob = ast_json_pack("{s: s, s: I}",
 								"peer_status", "Lagged",
-								"time", iaxs[fr->callno]->pingtime);
+								"time", (ast_json_int_t)iaxs[fr->callno]->pingtime);
 							ast_devstate_changed(AST_DEVICE_UNAVAILABLE, AST_DEVSTATE_CACHABLE, "IAX2/%s", peer->name); /* Activate notification */
 						}
 					}
@@ -13179,9 +13181,11 @@ static struct iax2_user *build_user(const char *name, struct ast_variable *v, st
 				if ((varval = strchr(varname, '='))) {
 					*varval = '\0';
 					varval++;
-					if((tmpvar = ast_variable_new(varname, varval, ""))) {
-						tmpvar->next = user->vars;
-						user->vars = tmpvar;
+					if ((tmpvar = ast_variable_new(varname, varval, ""))) {
+						if (ast_variable_list_replace(&user->vars, tmpvar)) {
+							tmpvar->next = user->vars;
+							user->vars = tmpvar;
+						}
 					}
 				}
 			} else if (!strcasecmp(v->name, "allow")) {

channels/chan_motif.c

@@ -2086,15 +2086,16 @@ static int jingle_interpret_description(struct jingle_session *session, iks *des
 
 	/* Iterate the codecs updating the relevant RTP instance as we go */
 	for (codec = iks_child(description); codec; codec = iks_next(codec)) {
-		char *id = iks_find_attrib(codec, "id"), *name = iks_find_attrib(codec, "name");
+		char *id = iks_find_attrib(codec, "id");
+		char *attr_name = iks_find_attrib(codec, "name");
 		char *clockrate = iks_find_attrib(codec, "clockrate");
 		int rtp_id, rtp_clockrate;
 
-		if (!ast_strlen_zero(id) && !ast_strlen_zero(name) && (sscanf(id, "%30d", &rtp_id) == 1)) {
+		if (!ast_strlen_zero(id) && !ast_strlen_zero(attr_name) && (sscanf(id, "%30d", &rtp_id) == 1)) {
 			if (!ast_strlen_zero(clockrate) && (sscanf(clockrate, "%30d", &rtp_clockrate) == 1)) {
-				ast_rtp_codecs_payloads_set_rtpmap_type_rate(&codecs, NULL, rtp_id, media, name, 0, rtp_clockrate);
+				ast_rtp_codecs_payloads_set_rtpmap_type_rate(&codecs, NULL, rtp_id, media, attr_name, 0, rtp_clockrate);
 			} else {
-				ast_rtp_codecs_payloads_set_rtpmap_type(&codecs, NULL, rtp_id, media, name, 0);
+				ast_rtp_codecs_payloads_set_rtpmap_type(&codecs, NULL, rtp_id, media, attr_name, 0);
 			}
 		}
 	}

channels/chan_pjsip.c

@@ -749,7 +749,8 @@ static int chan_pjsip_answer(struct ast_channel *ast)
 }
 
 /*! \brief Internal helper function called when CNG tone is detected */
-static struct ast_frame *chan_pjsip_cng_tone_detected(struct ast_sip_session *session, struct ast_frame *f)
+static struct ast_frame *chan_pjsip_cng_tone_detected(struct ast_channel *ast, struct ast_sip_session *session,
+	struct ast_frame *f)
 {
 	const char *target_context;
 	int exists;
@@ -765,11 +766,11 @@ static struct ast_frame *chan_pjsip_cng_tone_detected(struct ast_sip_session *se
 	}
 
 	/* If already executing in the fax extension don't do anything */
-	if (!strcmp(ast_channel_exten(session->channel), "fax")) {
+	if (!strcmp(ast_channel_exten(ast), "fax")) {
 		return f;
 	}
 
-	target_context = S_OR(ast_channel_macrocontext(session->channel), ast_channel_context(session->channel));
+	target_context = S_OR(ast_channel_macrocontext(ast), ast_channel_context(ast));
 
 	/*
 	 * We need to unlock the channel here because ast_exists_extension has the
@@ -778,25 +779,30 @@ static struct ast_frame *chan_pjsip_cng_tone_detected(struct ast_sip_session *se
 	 *
 	 * ast_async_goto() has its own restriction on not holding the channel lock.
 	 */
-	ast_channel_unlock(session->channel);
+	ast_channel_unlock(ast);
 	ast_frfree(f);
 	f = &ast_null_frame;
-	exists = ast_exists_extension(session->channel, target_context, "fax", 1,
-		S_COR(ast_channel_caller(session->channel)->id.number.valid,
-			ast_channel_caller(session->channel)->id.number.str, NULL));
+	exists = ast_exists_extension(ast, target_context, "fax", 1,
+		S_COR(ast_channel_caller(ast)->id.number.valid,
+			ast_channel_caller(ast)->id.number.str, NULL));
 	if (exists) {
 		ast_verb(2, "Redirecting '%s' to fax extension due to CNG detection\n",
-			ast_channel_name(session->channel));
-		pbx_builtin_setvar_helper(session->channel, "FAXEXTEN", ast_channel_exten(session->channel));
-		if (ast_async_goto(session->channel, target_context, "fax", 1)) {
+			ast_channel_name(ast));
+		pbx_builtin_setvar_helper(ast, "FAXEXTEN", ast_channel_exten(ast));
+		if (ast_async_goto(ast, target_context, "fax", 1)) {
 			ast_log(LOG_ERROR, "Failed to async goto '%s' into fax extension in '%s'\n",
-				ast_channel_name(session->channel), target_context);
+				ast_channel_name(ast), target_context);
 		}
 	} else {
 		ast_log(LOG_NOTICE, "FAX CNG detected on '%s' but no fax extension in '%s'\n",
-			ast_channel_name(session->channel), target_context);
+			ast_channel_name(ast), target_context);
 	}
-	ast_channel_lock(session->channel);
+
+	/* It's possible for a masquerade to have occurred when doing the ast_async_goto resulting in
+	 * the channel on the session having changed. Since we need to return with the original channel
+	 * locked we lock the channel that was passed in and not session->channel.
+	 */
+	ast_channel_lock(ast);
 
 	return f;
 }
@@ -895,7 +901,11 @@ static struct ast_frame *chan_pjsip_read_stream(struct ast_channel *ast)
 			if (f->subclass.integer == 'f') {
 				ast_debug(3, "Channel driver fax CNG detected on %s\n",
 					ast_channel_name(ast));
-				f = chan_pjsip_cng_tone_detected(session, f);
+				f = chan_pjsip_cng_tone_detected(ast, session, f);
+				/* When chan_pjsip_cng_tone_detected returns it is possible for the
+				 * channel pointed to by ast and by session->channel to differ due to a
+				 * masquerade. It's best not to touch things after this.
+				 */
 			} else {
 				ast_debug(3, "* Detected inband DTMF '%c' on '%s'\n", f->subclass.integer,
 					ast_channel_name(ast));
@@ -1648,6 +1658,7 @@ static int chan_pjsip_indicate(struct ast_channel *ast, int condition, const voi
 
 				if (ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), ast_format_vp8) != AST_FORMAT_CMP_NOT_EQUAL ||
 					ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), ast_format_vp9) != AST_FORMAT_CMP_NOT_EQUAL ||
+					ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), ast_format_h265) != AST_FORMAT_CMP_NOT_EQUAL ||
 					(channel->session->endpoint->media.webrtc &&
 					 ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), ast_format_h264) != AST_FORMAT_CMP_NOT_EQUAL)) {
 					/* FIXME Fake RTP write, this will be sent as an RTCP packet. Ideally the
@@ -1727,7 +1738,7 @@ static int chan_pjsip_indicate(struct ast_channel *ast, int condition, const voi
 		device_buf = alloca(device_buf_size);
 		ast_channel_get_device_name(ast, device_buf, device_buf_size);
 		ast_devstate_changed_literal(AST_DEVICE_ONHOLD, 1, device_buf);
-		if (!channel->session->endpoint->moh_passthrough) {
+		if (!channel->session->moh_passthrough) {
 			ast_moh_start(ast, data, NULL);
 		} else {
 			if (ast_sip_push_task(channel->session->serializer, remote_send_hold, ao2_bump(channel->session))) {
@@ -1743,7 +1754,7 @@ static int chan_pjsip_indicate(struct ast_channel *ast, int condition, const voi
 		device_buf = alloca(device_buf_size);
 		ast_channel_get_device_name(ast, device_buf, device_buf_size);
 		ast_devstate_changed_literal(AST_DEVICE_UNKNOWN, 1, device_buf);
-		if (!channel->session->endpoint->moh_passthrough) {
+		if (!channel->session->moh_passthrough) {
 			ast_moh_stop(ast);
 		} else {
 			if (ast_sip_push_task(channel->session->serializer, remote_send_unhold, ao2_bump(channel->session))) {
@@ -2250,6 +2261,12 @@ static int chan_pjsip_digit_end(struct ast_channel *ast, char digit, unsigned in
 	struct ast_sip_session_media *media;
 	int res = 0;
 
+	if (!channel || !channel->session) {
+		/* This happens when the channel is hungup while a DTMF digit is playing. See ASTERISK-28086 */
+		ast_debug(3, "Channel %s disappeared while calling digit_end\n", ast_channel_name(ast));
+		return -1;
+	}
+
 	media = channel->session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO];
 
 	switch (channel->session->dtmf) {
@@ -3211,6 +3228,12 @@ static struct ast_custom_function dtmf_mode_function = {
 	.write = pjsip_acf_dtmf_mode_write
 };
 
+static struct ast_custom_function moh_passthrough_function = {
+	.name = "PJSIP_MOH_PASSTHROUGH",
+	.read = pjsip_acf_moh_passthrough_read,
+	.write = pjsip_acf_moh_passthrough_write
+};
+
 static struct ast_custom_function session_refresh_function = {
 	.name = "PJSIP_SEND_SESSION_REFRESH",
 	.write = pjsip_acf_session_refresh_write,
@@ -3263,6 +3286,11 @@ static int load_module(void)
 		goto end;
 	}
 
+	if (ast_custom_function_register(&moh_passthrough_function)) {
+		ast_log(LOG_WARNING, "Unable to register PJSIP_MOH_PASSTHROUGH dialplan function\n");
+		goto end;
+	}
+
 	if (ast_custom_function_register(&session_refresh_function)) {
 		ast_log(LOG_WARNING, "Unable to register PJSIP_SEND_SESSION_REFRESH dialplan function\n");
 		goto end;
@@ -3308,6 +3336,7 @@ end:
 	ast_sip_session_unregister_supplement(&call_pickup_supplement);
 	ast_sip_unregister_service(&refer_callback_module);
 	ast_custom_function_unregister(&dtmf_mode_function);
+	ast_custom_function_unregister(&moh_passthrough_function);
 	ast_custom_function_unregister(&media_offer_function);
 	ast_custom_function_unregister(&chan_pjsip_dial_contacts_function);
 	ast_custom_function_unregister(&chan_pjsip_parse_uri_function);
@@ -3335,6 +3364,7 @@ static int unload_module(void)
 	ast_sip_unregister_service(&refer_callback_module);
 
 	ast_custom_function_unregister(&dtmf_mode_function);
+	ast_custom_function_unregister(&moh_passthrough_function);
 	ast_custom_function_unregister(&media_offer_function);
 	ast_custom_function_unregister(&chan_pjsip_dial_contacts_function);
 	ast_custom_function_unregister(&chan_pjsip_parse_uri_function);

channels/chan_rtp.c

@@ -43,6 +43,7 @@
 #include "asterisk/causes.h"
 #include "asterisk/format_cache.h"
 #include "asterisk/multicast_rtp.h"
+#include "asterisk/dns_core.h"
 
 /* Forward declarations */
 static struct ast_channel *multicast_rtp_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);
@@ -293,9 +294,23 @@ static struct ast_channel *unicast_rtp_request(const char *type, struct ast_form
 		ast_log(LOG_ERROR, "Destination is required for the 'UnicastRTP' channel\n");
 		goto failure;
 	}
+
 	if (!ast_sockaddr_parse(&address, args.destination, PARSE_PORT_REQUIRE)) {
-		ast_log(LOG_ERROR, "Destination '%s' could not be parsed\n", args.destination);
-		goto failure;
+	    int rc;
+	    char *host;
+	    char *port;
+
+	    rc = ast_sockaddr_split_hostport(args.destination, &host, &port, PARSE_PORT_REQUIRE);
+	    if (!rc) {
+	        ast_log(LOG_ERROR, "Unable to parse destination '%s' into host and port\n", args.destination);
+	        goto failure;
+	    }
+
+	    rc = ast_dns_resolve_ipv6_and_ipv4(&address, host, port);
+	    if (rc != 0) {
+	        ast_log(LOG_ERROR, "Unable to resolve host '%s'\n", host);
+	        goto failure;
+	    }
 	}
 
 	if (!ast_strlen_zero(args.options)

channels/chan_sip.c

@@ -8193,8 +8193,11 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
 	/* Use only the preferred audio format, which is stored at the '0' index */
 	fmt = ast_format_cap_get_best_by_type(what, AST_MEDIA_TYPE_AUDIO); /* get the best audio format */
 	if (fmt) {
+		int framing;
+
 		ast_format_cap_remove_by_type(caps, AST_MEDIA_TYPE_AUDIO); /* remove only the other audio formats */
-		ast_format_cap_append(caps, fmt, 0); /* add our best choice back */
+		framing = ast_format_cap_get_format_framing(what, fmt);
+		ast_format_cap_append(caps, fmt, framing); /* add our best choice back */
 	} else {
 		/* If we don't have an audio format, try to get something */
 		fmt = ast_format_cap_get_format(caps, 0);
@@ -19245,18 +19248,6 @@ static enum check_auth_result check_peer_ok(struct sip_pvt *p, char *of,
 		bogus_peer = NULL;
 	}
 
-	/*  build_peer, called through sip_find_peer, is not able to check the
-	 *  sip_pvt->natdetected flag in order to determine if the peer is behind
-	 *  NAT or not when SIP_PAGE3_NAT_AUTO_RPORT or SIP_PAGE3_NAT_AUTO_COMEDIA
-	 *  are set on the peer.  So we check for that here and set the peer's
-	 *  address accordingly.
-	 */
-	set_peer_nat(p, peer);
-
-	if (p->natdetected && ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_RPORT)) {
-		ast_sockaddr_copy(&peer->addr, &p->recv);
-	}
-
 	if (!ast_apply_acl(peer->acl, addr, "SIP Peer ACL: ")) {
 		ast_debug(2, "Found peer '%s' for '%s', but fails host access\n", peer->name, of);
 		sip_unref_peer(peer, "sip_unref_peer: check_peer_ok: from sip_find_peer call, early return of AUTH_ACL_FAILED");
@@ -19325,6 +19316,21 @@ static enum check_auth_result check_peer_ok(struct sip_pvt *p, char *of,
 		ast_string_field_set(p, peermd5secret, NULL);
 	}
 	if (!(res = check_auth(p, req, peer->name, p->peersecret, p->peermd5secret, sipmethod, uri2, reliable))) {
+
+		/* build_peer, called through sip_find_peer, is not able to check the
+		 * sip_pvt->natdetected flag in order to determine if the peer is behind
+		 * NAT or not when SIP_PAGE3_NAT_AUTO_RPORT or SIP_PAGE3_NAT_AUTO_COMEDIA
+		 * are set on the peer. So we check for that here and set the peer's
+		 * address accordingly. The address should ONLY be set once we are sure
+		 * authentication was a success. If, for example, an INVITE was sent that
+		 * matched the peer name but failed the authentication check, the address
+		 * would be updated, which is bad.
+		 */
+		set_peer_nat(p, peer);
+		if (p->natdetected && ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_RPORT)) {
+			ast_sockaddr_copy(&peer->addr, &p->recv);
+		}
+
 		/* If we have a call limit, set flag */
 		if (peer->call_limit)
 			ast_set_flag(&p->flags[0], SIP_CALL_LIMIT);
@@ -19424,6 +19430,7 @@ static enum check_auth_result check_peer_ok(struct sip_pvt *p, char *of,
 		}
 	}
 	sip_unref_peer(peer, "check_peer_ok: sip_unref_peer: tossing temp ptr to peer from sip_find_peer");
+
 	return res;
 }
 
@@ -31366,8 +31373,10 @@ static struct ast_variable *add_var(const char *buf, struct ast_variable *list)
 	if ((varval = strchr(varname, '='))) {
 		*varval++ = '\0';
 		if ((tmpvar = ast_variable_new(varname, varval, ""))) {
-			tmpvar->next = list;
-			list = tmpvar;
+			if (ast_variable_list_replace(&list, tmpvar)) {
+				tmpvar->next = list;
+				list = tmpvar;
+			}
 		}
 	}
 	return list;
@@ -35349,8 +35358,8 @@ static void deprecation_notice(void)
 {
 	ast_log(LOG_WARNING, "chan_sip has no official maintainer and is deprecated.  Migration to\n");
 	ast_log(LOG_WARNING, "chan_pjsip is recommended.  See guides at the Asterisk Wiki:\n");
-	ast_log(LOG_WARNING, "https://wiki.asterisk.org/wiki/x/tAHOAQ\n");
-	ast_log(LOG_WARNING, "https://wiki.asterisk.org/wiki/x/hYCLAQ\n");
+	ast_log(LOG_WARNING, "https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip\n");
+	ast_log(LOG_WARNING, "https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip\n");
 }
 
 /*! \brief Event callback which indicates we're fully booted */

channels/chan_unistim.c

@@ -62,6 +62,7 @@
 #include "asterisk/module.h"
 #include "asterisk/pbx.h"
 #include "asterisk/rtp_engine.h"
+#include "asterisk/unaligned.h"
 #include "asterisk/netsock2.h"
 #include "asterisk/acl.h"
 #include "asterisk/callerid.h"
@@ -575,7 +576,7 @@ static const unsigned char packet_send_stream_based_tone_on[] =
 	{ 0x16, 0x06, 0x1b, 0x00, 0x00, 0x05 };
 static const unsigned char packet_send_stream_based_tone_single_freq[] =
 	{ 0x16, 0x06, 0x1d, 0x00, 0x01, 0xb8 };
-static const unsigned char packet_send_stream_based_tone_dial_freq[] =
+static const unsigned char packet_send_stream_based_tone_dual_freq[] =
 	{ 0x16, 0x08, 0x1d, 0x00, 0x01, 0xb8, 0x01, 0x5e };
 static const unsigned char packet_send_select_output[] =
 	{ 0x16, 0x06, 0x32, 0xc0, 0x01, 0x00 };
@@ -1002,27 +1003,36 @@ static int get_to_address(int fd, struct sockaddr_in *toAddr)
 {
 #ifdef HAVE_PKTINFO
 	int err;
-	struct msghdr msg;
-	struct {
-		struct cmsghdr cm;
-		int len;
-		struct in_addr address;
-	} ip_msg;
-
-	/* Zero out the structures before we use them */
-	/* This sets several key values to NULL */
-	memset(&msg, 0, sizeof(msg));
-	memset(&ip_msg, 0, sizeof(ip_msg));
-
-	/* Initialize the message structure */
-	msg.msg_control = &ip_msg;
-	msg.msg_controllen = sizeof(ip_msg);
+	char cmbuf[0x100];
+	struct cmsghdr *cmsg;
+	struct sockaddr_in peeraddr;
+	struct in_addr addr;
+	struct msghdr mh = {
+		.msg_name = &peeraddr,
+		.msg_namelen = sizeof(peeraddr),
+		.msg_control = cmbuf,
+		.msg_controllen = sizeof(cmbuf),
+	};
+	memset(&addr, 0, sizeof(addr));
 	/* Get info about the incoming packet */
-	err = recvmsg(fd, &msg, MSG_PEEK);
+	err = recvmsg(fd, &mh, MSG_PEEK);
 	if (err == -1) {
 		ast_log(LOG_WARNING, "recvmsg returned an error: %s\n", strerror(errno));
+		return err;
 	}
-	memcpy(&toAddr->sin_addr, &ip_msg.address, sizeof(struct in_addr));
+	for(cmsg = CMSG_FIRSTHDR(&mh);
+		cmsg != NULL;
+		cmsg = CMSG_NXTHDR(&mh, cmsg))
+	{
+		 if (cmsg->cmsg_level == IPPROTO_IP && cmsg->cmsg_type == IP_PKTINFO) {
+			struct in_pktinfo *pkt = (struct in_pktinfo*)CMSG_DATA(cmsg);
+			addr = pkt->ipi_addr;
+			if (unistimdebug) {
+				ast_verb(0, "message received on address %s\n", ast_inet_ntoa(addr));
+			}
+		}
+	}
+	memcpy(&toAddr->sin_addr, &addr, sizeof(struct in_addr));
 	return err;
 #else
 	memcpy(toAddr, &public_ip, sizeof(*toAddr));
@@ -1030,6 +1040,7 @@ static int get_to_address(int fd, struct sockaddr_in *toAddr)
 #endif
 }
 
+
 /* Allocate memory & initialize structures for a new phone */
 /* addr_from : ip address of the phone */
 static struct unistimsession *create_client(const struct sockaddr_in *addr_from)
@@ -1041,7 +1052,10 @@ static struct unistimsession *create_client(const struct sockaddr_in *addr_from)
 		return NULL;
 
 	memcpy(&s->sin, addr_from, sizeof(struct sockaddr_in));
-	get_to_address(unistimsock, &s->sout);
+	if (get_to_address(unistimsock, &s->sout) < 0) {
+		ast_free(s);
+		return NULL;
+	}
 	s->sout.sin_family = AF_INET;
 	if (unistimdebug) {
 		ast_verb(0, "Creating a new entry for the phone from %s received via server ip %s\n",
@@ -1208,19 +1222,16 @@ static void send_tone(struct unistimsession *pte, uint16_t tone1, uint16_t tone2
 	if (!tone2) {
 		memcpy(buffsend + SIZE_HEADER, packet_send_stream_based_tone_single_freq,
 			   sizeof(packet_send_stream_based_tone_single_freq));
-		buffsend[10] = (tone1 & 0xff00) >> 8;
-		buffsend[11] = (tone1 & 0x00ff);
+		put_unaligned_uint16(&buffsend[10], htons(tone1));
 		send_client(SIZE_HEADER + sizeof(packet_send_stream_based_tone_single_freq), buffsend,
 				   pte);
 	} else {
 		tone2 *= 8;
-		memcpy(buffsend + SIZE_HEADER, packet_send_stream_based_tone_dial_freq,
-			   sizeof(packet_send_stream_based_tone_dial_freq));
-		buffsend[10] = (tone1 & 0xff00) >> 8;
-		buffsend[11] = (tone1 & 0x00ff);
-		buffsend[12] = (tone2 & 0xff00) >> 8;
-		buffsend[13] = (tone2 & 0x00ff);
-		send_client(SIZE_HEADER + sizeof(packet_send_stream_based_tone_dial_freq), buffsend,
+		memcpy(buffsend + SIZE_HEADER, packet_send_stream_based_tone_dual_freq,
+			   sizeof(packet_send_stream_based_tone_dual_freq));
+		put_unaligned_uint16(&buffsend[10], htons(tone1));
+		put_unaligned_uint16(&buffsend[12], htons(tone2));
+		send_client(SIZE_HEADER + sizeof(packet_send_stream_based_tone_dual_freq), buffsend,
 				   pte);
 	}
 
@@ -2747,7 +2758,7 @@ static void send_start_rtp(struct unistim_subchannel *sub)
 		   sizeof(packet_send_jitter_buffer_conf));
 	send_client(SIZE_HEADER + sizeof(packet_send_jitter_buffer_conf), buffsend, pte);
 	if (pte->device->rtp_method != 0) {
-		uint16_t rtcpsin_port = htons(us.sin_port) + 1; /* RTCP port is RTP + 1 */
+		uint16_t rtcpsin_port = ntohs(us.sin_port) + 1; /* RTCP port is RTP + 1 */
 
 		if (unistimdebug) {
 			ast_verb(0, "Sending OpenAudioStreamTX using method #%d\n", pte->device->rtp_method);
@@ -2761,20 +2772,14 @@ static void send_start_rtp(struct unistim_subchannel *sub)
 		}
 		if (pte->device->rtp_method != 2) {
 			memcpy(buffsend + 28, &public.sin_addr, sizeof(public.sin_addr));
-			buffsend[20] = (htons(sin.sin_port) & 0xff00) >> 8;
-			buffsend[21] = (htons(sin.sin_port) & 0x00ff);
-			buffsend[23] = (rtcpsin_port & 0x00ff);
-			buffsend[22] = (rtcpsin_port & 0xff00) >> 8;
-			buffsend[25] = (us.sin_port & 0xff00) >> 8;
-			buffsend[24] = (us.sin_port & 0x00ff);
-			buffsend[27] = (rtcpsin_port & 0x00ff);
-			buffsend[26] = (rtcpsin_port & 0xff00) >> 8;
+			put_unaligned_uint16(&buffsend[20], sin.sin_port);
+			put_unaligned_uint16(&buffsend[22], htons(rtcpsin_port));
+			put_unaligned_uint16(&buffsend[24], us.sin_port);
+			put_unaligned_uint16(&buffsend[26], htons(rtcpsin_port));
 		} else {
 			memcpy(buffsend + 23, &public.sin_addr, sizeof(public.sin_addr));
-			buffsend[15] = (htons(sin.sin_port) & 0xff00) >> 8;
-			buffsend[16] = (htons(sin.sin_port) & 0x00ff);
-			buffsend[20] = (us.sin_port & 0xff00) >> 8;
-			buffsend[19] = (us.sin_port & 0x00ff);
+			put_unaligned_uint16(&buffsend[15], sin.sin_port);
+			put_unaligned_uint16(&buffsend[19], us.sin_port);
 		}
 		buffsend[11] = codec; /* rx */
 		buffsend[12] = codec; /* tx */
@@ -2792,20 +2797,14 @@ static void send_start_rtp(struct unistim_subchannel *sub)
 		}
 		if (pte->device->rtp_method != 2) {
 			memcpy(buffsend + 28, &public.sin_addr, sizeof(public.sin_addr));
-			buffsend[20] = (htons(sin.sin_port) & 0xff00) >> 8;
-			buffsend[21] = (htons(sin.sin_port) & 0x00ff);
-			buffsend[23] = (rtcpsin_port & 0x00ff);
-			buffsend[22] = (rtcpsin_port & 0xff00) >> 8;
-			buffsend[25] = (us.sin_port & 0xff00) >> 8;
-			buffsend[24] = (us.sin_port & 0x00ff);
-			buffsend[27] = (rtcpsin_port & 0x00ff);
-			buffsend[26] = (rtcpsin_port & 0xff00) >> 8;
+			put_unaligned_uint16(&buffsend[20], sin.sin_port);
+			put_unaligned_uint16(&buffsend[22], htons(rtcpsin_port));
+			put_unaligned_uint16(&buffsend[24], us.sin_port);
+			put_unaligned_uint16(&buffsend[26], htons(rtcpsin_port));
 		} else {
 			memcpy(buffsend + 23, &public.sin_addr, sizeof(public.sin_addr));
-			buffsend[15] = (htons(sin.sin_port) & 0xff00) >> 8;
-			buffsend[16] = (htons(sin.sin_port) & 0x00ff);
-			buffsend[20] = (us.sin_port & 0xff00) >> 8;
-			buffsend[19] = (us.sin_port & 0x00ff);
+			put_unaligned_uint16(&buffsend[15], sin.sin_port);
+			put_unaligned_uint16(&buffsend[19], us.sin_port);
 		}
 		buffsend[11] = codec; /* rx */
 		buffsend[12] = codec; /* tx */
@@ -2820,11 +2819,9 @@ static void send_start_rtp(struct unistim_subchannel *sub)
 		memcpy(buffsend + SIZE_HEADER, packet_send_call, sizeof(packet_send_call));
 		memcpy(buffsend + 53, &public.sin_addr, sizeof(public.sin_addr));
 		/* Destination port when sending RTP */
-		buffsend[49] = (us.sin_port & 0x00ff);
-		buffsend[50] = (us.sin_port & 0xff00) >> 8;
+		put_unaligned_uint16(&buffsend[49], us.sin_port);
 		/* Destination port when sending RTCP */
-		buffsend[52] = (rtcpsin_port & 0x00ff);
-		buffsend[51] = (rtcpsin_port & 0xff00) >> 8;
+		put_unaligned_uint16(&buffsend[51], htons(rtcpsin_port));
 		/* Codec */
 		buffsend[40] = codec;
 		buffsend[41] = codec;
@@ -2841,10 +2838,8 @@ static void send_start_rtp(struct unistim_subchannel *sub)
 				ast_format_get_name(ast_channel_readformat(sub->owner)));
 		}
 		/* Source port for transmit RTP and Destination port for receiving RTP */
-		buffsend[45] = (htons(sin.sin_port) & 0xff00) >> 8;
-		buffsend[46] = (htons(sin.sin_port) & 0x00ff);
-		buffsend[47] = (rtcpsin_port & 0xff00) >> 8;
-		buffsend[48] = (rtcpsin_port & 0x00ff);
+		put_unaligned_uint16(&buffsend[45], sin.sin_port);
+		put_unaligned_uint16(&buffsend[47], htons(rtcpsin_port));
 		send_client(SIZE_HEADER + sizeof(packet_send_call), buffsend, pte);
 	}
 }
@@ -3331,22 +3326,12 @@ static void handle_call_incoming(struct unistimsession *s)
 	return;
 }
 
-static int unistim_do_senddigit(struct unistimsession *pte, char digit)
+static int send_dtmf_tone(struct unistimsession *pte, char digit)
 {
-	struct ast_frame f = { .frametype = AST_FRAME_DTMF, .subclass.integer = digit, .src = "unistim" };
-	struct unistim_subchannel *sub;
-        int row, col;
+	int row, col;
 
-	sub = get_sub(pte->device, SUB_REAL);
-	if (!sub || !sub->owner || sub->alreadygone) {
-		ast_log(LOG_WARNING, "Unable to find subchannel in dtmf senddigit\n");
-		return -1;
-	}
-
-	/* Send DTMF indication _before_ playing sounds */
-	ast_queue_frame(sub->owner, &f);
 	if (unistimdebug) {
-		ast_verb(0, "Send Digit %c (%i ms)\n", digit, pte->device->dtmfduration);
+		ast_verb(0, "Phone Play Digit %c\n", digit);
 	}
 	if (pte->device->dtmfduration > 0) {
 		row = (digit - '1') % 3;
@@ -3364,6 +3349,28 @@ static int unistim_do_senddigit(struct unistimsession *pte, char digit)
 		} else {
 			send_tone(pte, 500, 2000);
 		}
+	}
+	return 0;
+}
+
+static int unistim_do_senddigit(struct unistimsession *pte, char digit)
+{
+	struct ast_frame f = { .frametype = AST_FRAME_DTMF, .subclass.integer = digit, .src = "unistim" };
+	struct unistim_subchannel *sub;
+
+	sub = get_sub(pte->device, SUB_REAL);
+	if (!sub || !sub->owner || sub->alreadygone) {
+		ast_log(LOG_WARNING, "Unable to find subchannel in dtmf senddigit\n");
+		return -1;
+	}
+
+	/* Send DTMF indication _before_ playing sounds */
+	ast_queue_frame(sub->owner, &f);
+	if (pte->device->dtmfduration > 0) {
+		if (unistimdebug) {
+			ast_verb(0, "Send Digit %c (%i ms)\n", digit, pte->device->dtmfduration);
+		}
+		send_dtmf_tone(pte, digit);
 		usleep(pte->device->dtmfduration * 1000);	 /* XXX Less than perfect, blocking an important thread is not a good idea */
 		send_tone(pte, 0, 0);
 	}
@@ -5506,34 +5513,21 @@ static int unistim_senddigit_begin(struct ast_channel *ast, char digit)
 	if (!pte) {
 		return -1;
 	}
-	return unistim_do_senddigit(pte, digit);
+	return send_dtmf_tone(pte, digit);
 }
 
 static int unistim_senddigit_end(struct ast_channel *ast, char digit, unsigned int duration)
 {
 	struct unistimsession *pte = channel_to_session(ast);
-	struct ast_frame f = { 0, };
-	struct unistim_subchannel *sub;
 
-	sub = get_sub(pte->device, SUB_REAL);
-
-	if (!sub || !sub->owner || sub->alreadygone) {
-		ast_log(LOG_WARNING, "Unable to find subchannel in dtmf senddigit_end\n");
+	if (!pte) {
 		return -1;
 	}
 
 	if (unistimdebug) {
-		ast_verb(0, "Send Digit off %c\n", digit);
-	}
-	if (!pte) {
-		return -1;
+		ast_verb(0, "Send Digit off %c (duration %d)\n", digit, duration);
 	}
 	send_tone(pte, 0, 0);
-	f.frametype = AST_FRAME_DTMF;
-	f.subclass.integer = digit;
-	f.src = "unistim";
-	ast_queue_frame(sub->owner, &f);
-
 	return 0;
 }
 

channels/pjsip/cli_commands.c

@@ -339,14 +339,14 @@ static int cli_channelstats_print_body(void *obj, void *arg, int flags)
 	struct ast_sip_cli_context *context = arg;
 	const struct ast_channel_snapshot *snapshot = obj;
 	struct ast_channel *channel = ast_channel_get_by_name(snapshot->base->name);
-	struct ast_sip_channel_pvt *cpvt = channel ? ast_channel_tech_pvt(channel) : NULL;
+	struct ast_sip_channel_pvt *cpvt = NULL;
 	struct ast_sip_session *session;
 	struct ast_sip_session_media *media;
-	struct ast_rtp_instance *rtp;
 	struct ast_rtp_instance_stats stats;
 	char *print_name = NULL;
 	char *print_time = alloca(32);
 	char codec_in_use[7];
+	int stats_res = -1;
 
 	ast_assert(context->output_buffer != NULL);
 
@@ -357,7 +357,8 @@ static int cli_channelstats_print_body(void *obj, void *arg, int flags)
 
 	ast_channel_lock(channel);
 
-	session = cpvt->session;
+	cpvt = ast_channel_tech_pvt(channel);
+	session = cpvt ? cpvt->session : NULL;
 	if (!session) {
 		ast_str_append(&context->output_buffer, 0, " %s not valid\n", snapshot->base->name);
 		ast_channel_unlock(channel);
@@ -373,14 +374,13 @@ static int cli_channelstats_print_body(void *obj, void *arg, int flags)
 		return 0;
 	}
 
-	rtp = ao2_bump(media->rtp);
-
 	codec_in_use[0] = '\0';
 
 	if (ast_channel_rawreadformat(channel)) {
 		ast_copy_string(codec_in_use, ast_format_get_name(ast_channel_rawreadformat(channel)), sizeof(codec_in_use));
 	}
 
+	stats_res = ast_rtp_instance_get_stats(media->rtp, &stats, AST_RTP_INSTANCE_STAT_ALL);
 	ast_channel_unlock(channel);
 
 	print_name = ast_strdupa(snapshot->base->name);
@@ -389,7 +389,7 @@ static int cli_channelstats_print_body(void *obj, void *arg, int flags)
 
 	ast_format_duration_hh_mm_ss(ast_tvnow().tv_sec - snapshot->base->creationtime.tv_sec, print_time, 32);
 
-	if (ast_rtp_instance_get_stats(rtp, &stats, AST_RTP_INSTANCE_STAT_ALL)) {
+	if (stats_res == -1) {
 		ast_str_append(&context->output_buffer, 0, "%s direct media\n", snapshot->base->name);
 	} else {
 		ast_str_append(&context->output_buffer, 0,
@@ -414,7 +414,6 @@ static int cli_channelstats_print_body(void *obj, void *arg, int flags)
 		);
 	}
 
-	ao2_cleanup(rtp);
 	ao2_cleanup(channel);
 
 	return 0;

channels/pjsip/dialplan_functions.c

@@ -80,6 +80,19 @@
 		<para>This function uses the same DTMF mode naming as the dtmf_mode configuration option</para>
 	</description>
 </function>
+<function name="PJSIP_MOH_PASSTHROUGH" language="en_US">
+	<synopsis>
+		Get or change the on-hold behavior for a SIP call.
+	</synopsis>
+	<syntax>
+	</syntax>
+	<description>
+		<para>When read, returns the current moh passthrough mode</para>
+		<para>When written, sets the current moh passthrough mode</para>
+		<para>If <replaceable>yes</replaceable>, on-hold re-INVITEs are sent. If <replaceable>no</replaceable>, music on hold is generated.</para>
+		<para>This function can be used to override the moh_passthrough configuration option</para>
+	</description>
+</function>
 <function name="PJSIP_SEND_SESSION_REFRESH" language="en_US">
 	<synopsis>
 		W/O: Initiate a session refresh via an UPDATE or re-INVITE on an established media session
@@ -1431,6 +1444,34 @@ int pjsip_acf_dtmf_mode_read(struct ast_channel *chan, const char *cmd, char *da
 	return 0;
 }
 
+int pjsip_acf_moh_passthrough_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
+{
+	struct ast_sip_channel_pvt *channel;
+
+	if (!chan) {
+		ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
+		return -1;
+	}
+
+	if (len < 3) {
+		ast_log(LOG_WARNING, "%s: buffer too small\n", cmd);
+		return -1;
+	}
+
+	ast_channel_lock(chan);
+	if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
+		ast_log(LOG_WARNING, "Cannot call %s on a non-PJSIP channel\n", cmd);
+		ast_channel_unlock(chan);
+		return -1;
+	}
+
+	channel = ast_channel_tech_pvt(chan);
+	strncpy(buf, AST_YESNO(channel->session->moh_passthrough), len);
+
+	ast_channel_unlock(chan);
+	return 0;
+}
+
 struct refresh_data {
 	struct ast_sip_session *session;
 	enum ast_sip_session_refresh_method method;
@@ -1574,6 +1615,30 @@ int pjsip_acf_dtmf_mode_write(struct ast_channel *chan, const char *cmd, char *d
 	return ast_sip_push_task_wait_serializer(channel->session->serializer, dtmf_mode_refresh_cb, &rdata);
 }
 
+int pjsip_acf_moh_passthrough_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
+{
+	struct ast_sip_channel_pvt *channel;
+
+	if (!chan) {
+		ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
+		return -1;
+	}
+
+	ast_channel_lock(chan);
+	if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
+		ast_log(LOG_WARNING, "Cannot call %s on a non-PJSIP channel\n", cmd);
+		ast_channel_unlock(chan);
+		return -1;
+	}
+
+	channel = ast_channel_tech_pvt(chan);
+	channel->session->moh_passthrough = ast_true(value);
+
+	ast_channel_unlock(chan);
+
+	return 0;
+}
+
 static int refresh_write_cb(void *obj)
 {
 	struct refresh_data *data = obj;

channels/pjsip/include/dialplan_functions.h

@@ -72,6 +72,31 @@ int pjsip_acf_dtmf_mode_read(struct ast_channel *chan, const char *cmd, char *da
  */
 int pjsip_acf_dtmf_mode_write(struct ast_channel *chan, const char *cmd, char *data, const char *value);
 
+/*!
+ * \brief PJSIP_MOH_PASSTHROUGH function read callback
+ * \param chan The channel the function is called on
+ * \param cmd The name of the function
+ * \param data Arguments passed to the function
+ * \param buf Out buffer that should be populated with the data
+ * \param len Size of the buffer
+ *
+ * \retval 0 on success
+ * \retval -1 on failure
+ */
+int pjsip_acf_moh_passthrough_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len);
+
+/*!
+ * \brief PJSIP_MOH_PASSTHROUGH function write callback
+ * \param chan The channel the function is called on
+ * \param cmd The name of the function
+ * \param data Arguments passed to the function
+ * \param value Value to be set by the function
+ *
+ * \retval 0 on success
+ * \retval -1 on failure
+ */
+int pjsip_acf_moh_passthrough_write(struct ast_channel *chan, const char *cmd, char *data, const char *value);
+
 /*!
  * \brief PJSIP_MEDIA_OFFER function read callback
  * \param chan The channel the function is called on

channels/sig_pri.c

@@ -2230,7 +2230,7 @@ static void *pri_ss_thread(void *data)
 
 void pri_event_alarm(struct sig_pri_span *pri, int index, int before_start_pri)
 {
-	pri->dchanavail[index] &= ~(DCHAN_NOTINALARM | DCHAN_UP);
+	pri->dchanavail[index] &= ~DCHAN_NOTINALARM;
 	if (!before_start_pri) {
 		pri_find_dchan(pri);
 	}

codecs/Makefile

@@ -90,4 +90,5 @@ SPEEX_RESAMPLE_CFLAGS:=
 endif
 
 $(call MOD_ADD_C,codec_resample,speex/resample.c)
-speex/resample.o: _ASTCFLAGS+=$(SPEEX_RESAMPLE_CFLAGS)
+codec_resample.o: _ASTCFLAGS+=-DOUTSIDE_SPEEX
+speex/resample.o: _ASTCFLAGS+=$(SPEEX_RESAMPLE_CFLAGS) -DOUTSIDE_SPEEX -DEXPORT=

codecs/ex_alaw.h

@@ -7,6 +7,9 @@
  *
  */
 
+#ifndef ASTERISK_EX_ALAW_H
+#define ASTERISK_EX_ALAW_H
+
 static uint8_t ex_alaw[] = {
 	0x00, 0x03, 0x06, 0x09, 0x0c, 0x0f, 0x12, 0x15,
 	0x10, 0x18, 0x1b, 0x1e, 0x21, 0x24, 0x27, 0x2a,
@@ -34,3 +37,5 @@ static struct ast_frame *alaw_sample(void)
 	f.subclass.format = ast_format_alaw;
 	return &f;
 }
+
+#endif /* ASTERISK_EX_ALAW_H */

codecs/ex_g722.h

@@ -7,6 +7,9 @@
  *
  */
 
+#ifndef ASTERISK_EX_G722_H
+#define ASTERISK_EX_G722_H
+
 static uint8_t ex_g722[] = {
 	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
 	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
@@ -46,3 +49,5 @@ static struct ast_frame *g722_sample(void)
 
 	return &f;
 }
+
+#endif /* ASTERISK_EX_G722_H */

codecs/ex_ulaw.h

@@ -7,6 +7,9 @@
  *
  */
 
+#ifndef ASTERISK_EX_ULAW_H
+#define ASTERISK_EX_ULAW_H
+
 static uint8_t ex_ulaw[] = {
 	0x00, 0x03, 0x06, 0x09, 0x0c, 0x0f, 0x12, 0x15,
 	0x10, 0x18, 0x1b, 0x1e, 0x21, 0x24, 0x27, 0x2a,
@@ -36,3 +39,5 @@ static struct ast_frame *ulaw_sample(void)
 
 	return &f;
 }
+
+#endif /* ASTERISK_EX_ULAW_H */

codecs/speex/arch.h

@@ -35,14 +35,6 @@
 #ifndef ARCH_H
 #define ARCH_H
 
-#ifndef SPEEX_VERSION
-#define SPEEX_MAJOR_VERSION 1         /**< Major Speex version. */
-#define SPEEX_MINOR_VERSION 1         /**< Minor Speex version. */
-#define SPEEX_MICRO_VERSION 15        /**< Micro Speex version. */
-#define SPEEX_EXTRA_VERSION ""        /**< Extra Speex version. */
-#define SPEEX_VERSION "speex-1.2beta3"  /**< Speex version string. */
-#endif
-
 #define FIXED_POINT
 
 /* A couple test to catch stupid option combinations */
@@ -77,7 +69,7 @@
 #endif
 
 #ifndef OUTSIDE_SPEEX
-#include "speex/speex_types.h"
+#include "speex/speexdsp_types.h"
 #endif
 
 #define ABS(x) ((x) < 0 ? (-(x)) : (x))      /**< Absolute integer value. */
@@ -91,7 +83,7 @@
 #ifdef FIXED_POINT
 
 typedef spx_int16_t spx_word16_t;
-typedef spx_int32_t   spx_word32_t;
+typedef spx_int32_t spx_word32_t;
 typedef spx_word32_t spx_mem_t;
 typedef spx_word16_t spx_coef_t;
 typedef spx_word16_t spx_lsp_t;
@@ -173,6 +165,7 @@ typedef float spx_word32_t;
 #define VSHR32(a,shift) (a)
 #define SATURATE16(x,a) (x)
 #define SATURATE32(x,a) (x)
+#define SATURATE32PSHR(x,shift,a) (x)
 
 #define PSHR(a,shift)       (a)
 #define SHR(a,shift)       (a)

codecs/speex/fixed_generic.h

@@ -52,6 +52,10 @@
 #define SATURATE16(x,a) (((x)>(a) ? (a) : (x)<-(a) ? -(a) : (x)))
 #define SATURATE32(x,a) (((x)>(a) ? (a) : (x)<-(a) ? -(a) : (x)))
 
+#define SATURATE32PSHR(x,shift,a) (((x)>=(SHL32(a,shift))) ? (a) : \
+                                   (x)<=-(SHL32(a,shift)) ? -(a) : \
+                                   (PSHR32(x, shift)))
+
 #define SHR(a,shift) ((a) >> (shift))
 #define SHL(a,shift) ((spx_word32_t)(a) << (shift))
 #define PSHR(a,shift) (SHR((a)+((EXTEND32(1)<<((shift))>>1)),shift))

codecs/speex/resample.c

@@ -61,18 +61,26 @@
 #include "config.h"
 #endif
 
+#ifdef OUTSIDE_SPEEX
 #include <stdlib.h>
 static void *speex_alloc (int size) {return calloc(size,1);}
 static void *speex_realloc (void *ptr, int size) {return realloc(ptr, size);}
 static void speex_free (void *ptr) {free(ptr);}
 #include "speex_resampler.h"
 #include "arch.h"
+#else /* OUTSIDE_SPEEX */
+
+#include "speex/speex_resampler.h"
+#include "arch.h"
+#include "os_support.h"
+#endif /* OUTSIDE_SPEEX */
 
 #include "stack_alloc.h"
 #include <math.h>
+#include <limits.h>
 
 #ifndef M_PI
-#define M_PI 3.14159263
+#define M_PI 3.14159265358979323846
 #endif
 
 #ifdef FIXED_POINT
@@ -92,6 +100,10 @@ static void speex_free (void *ptr) {free(ptr);}
 #include "resample_sse.h"
 #endif
 
+#ifdef _USE_NEON
+#include "resample_neon.h"
+#endif
+
 /* Numer of elements to allocate on the stack */
 #ifdef VAR_ARRAYS
 #define FIXED_STACK_ALLOC 8192
@@ -133,7 +145,7 @@ struct SpeexResamplerState_ {
    int    out_stride;
 } ;
 
-static double kaiser12_table[68] = {
+static const double kaiser12_table[68] = {
    0.99859849, 1.00000000, 0.99859849, 0.99440475, 0.98745105, 0.97779076,
    0.96549770, 0.95066529, 0.93340547, 0.91384741, 0.89213598, 0.86843014,
    0.84290116, 0.81573067, 0.78710866, 0.75723148, 0.72629970, 0.69451601,
@@ -147,7 +159,7 @@ static double kaiser12_table[68] = {
    0.00105297, 0.00069463, 0.00043489, 0.00025272, 0.00013031, 0.0000527734,
    0.00001000, 0.00000000};
 /*
-static double kaiser12_table[36] = {
+static const double kaiser12_table[36] = {
    0.99440475, 1.00000000, 0.99440475, 0.97779076, 0.95066529, 0.91384741,
    0.86843014, 0.81573067, 0.75723148, 0.69451601, 0.62920216, 0.56287762,
    0.49704014, 0.43304576, 0.37206735, 0.31506490, 0.26276832, 0.21567274,
@@ -155,7 +167,7 @@ static double kaiser12_table[36] = {
    0.03111947, 0.02127838, 0.01402878, 0.00886058, 0.00531256, 0.00298291,
    0.00153438, 0.00069463, 0.00025272, 0.0000527734, 0.00000500, 0.00000000};
 */
-static double kaiser10_table[36] = {
+static const double kaiser10_table[36] = {
    0.99537781, 1.00000000, 0.99537781, 0.98162644, 0.95908712, 0.92831446,
    0.89005583, 0.84522401, 0.79486424, 0.74011713, 0.68217934, 0.62226347,
    0.56155915, 0.50119680, 0.44221549, 0.38553619, 0.33194107, 0.28205962,
@@ -163,7 +175,7 @@ static double kaiser10_table[36] = {
    0.05731132, 0.04193980, 0.02979584, 0.02044510, 0.01345224, 0.00839739,
    0.00488951, 0.00257636, 0.00115101, 0.00035515, 0.00000000, 0.00000000};
 
-static double kaiser8_table[36] = {
+static const double kaiser8_table[36] = {
    0.99635258, 1.00000000, 0.99635258, 0.98548012, 0.96759014, 0.94302200,
    0.91223751, 0.87580811, 0.83439927, 0.78875245, 0.73966538, 0.68797126,
    0.63451750, 0.58014482, 0.52566725, 0.47185369, 0.41941150, 0.36897272,
@@ -171,7 +183,7 @@ static double kaiser8_table[36] = {
    0.10562887, 0.08273982, 0.06335451, 0.04724088, 0.03412321, 0.02369490,
    0.01563093, 0.00959968, 0.00527363, 0.00233883, 0.00050000, 0.00000000};
 
-static double kaiser6_table[36] = {
+static const double kaiser6_table[36] = {
    0.99733006, 1.00000000, 0.99733006, 0.98935595, 0.97618418, 0.95799003,
    0.93501423, 0.90755855, 0.87598009, 0.84068475, 0.80211977, 0.76076565,
    0.71712752, 0.67172623, 0.62508937, 0.57774224, 0.53019925, 0.48295561,
@@ -180,19 +192,19 @@ static double kaiser6_table[36] = {
    0.05031820, 0.03607231, 0.02432151, 0.01487334, 0.00752000, 0.00000000};
 
 struct FuncDef {
-   double *table;
+   const double *table;
    int oversample;
 };
 
-static struct FuncDef _KAISER12 = {kaiser12_table, 64};
+static const struct FuncDef _KAISER12 = {kaiser12_table, 64};
 #define KAISER12 (&_KAISER12)
 /*static struct FuncDef _KAISER12 = {kaiser12_table, 32};
 #define KAISER12 (&_KAISER12)*/
-static struct FuncDef _KAISER10 = {kaiser10_table, 32};
+static const struct FuncDef _KAISER10 = {kaiser10_table, 32};
 #define KAISER10 (&_KAISER10)
-static struct FuncDef _KAISER8 = {kaiser8_table, 32};
+static const struct FuncDef _KAISER8 = {kaiser8_table, 32};
 #define KAISER8 (&_KAISER8)
-static struct FuncDef _KAISER6 = {kaiser6_table, 32};
+static const struct FuncDef _KAISER6 = {kaiser6_table, 32};
 #define KAISER6 (&_KAISER6)
 
 struct QualityMapping {
@@ -200,7 +212,7 @@ struct QualityMapping {
    int oversample;
    float downsample_bandwidth;
    float upsample_bandwidth;
-   struct FuncDef *window_func;
+   const struct FuncDef *window_func;
 };
 
 
@@ -227,7 +239,7 @@ static const struct QualityMapping quality_map[11] = {
    {256, 32, 0.975f, 0.975f, KAISER12}, /* Q10 */ /* 96.6% cutoff (~100 dB stop) 10 */
 };
 /*8,24,40,56,80,104,128,160,200,256,320*/
-static double compute_func(float x, struct FuncDef *func)
+static double compute_func(float x, const struct FuncDef *func)
 {
    float y, frac;
    double interp[4];
@@ -262,7 +274,7 @@ int main(int argc, char **argv)
 
 #ifdef FIXED_POINT
 /* The slow way of computing a sinc for the table. Should improve that some day */
-static spx_word16_t sinc(float cutoff, float x, int N, struct FuncDef *window_func)
+static spx_word16_t sinc(float cutoff, float x, int N, const struct FuncDef *window_func)
 {
    /*fprintf (stderr, "%f ", x);*/
    float xx = x * cutoff;
@@ -275,7 +287,7 @@ static spx_word16_t sinc(float cutoff, float x, int N, struct FuncDef *window_fu
 }
 #else
 /* The slow way of computing a sinc for the table. Should improve that some day */
-static spx_word16_t sinc(float cutoff, float x, int N, struct FuncDef *window_func)
+static spx_word16_t sinc(float cutoff, float x, int N, const struct FuncDef *window_func)
 {
    /*fprintf (stderr, "%f ", x);*/
    float xx = x * cutoff;
@@ -330,28 +342,35 @@ static int resampler_basic_direct_single(SpeexResamplerState *st, spx_uint32_t c
    const int frac_advance = st->frac_advance;
    const spx_uint32_t den_rate = st->den_rate;
    spx_word32_t sum;
-   int j;
 
    while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len))
    {
-      const spx_word16_t *sinc = & sinc_table[samp_frac_num*N];
+      const spx_word16_t *sinct = & sinc_table[samp_frac_num*N];
       const spx_word16_t *iptr = & in[last_sample];
 
 #ifndef OVERRIDE_INNER_PRODUCT_SINGLE
-      float accum[4] = {0,0,0,0};
+      int j;
+      sum = 0;
+      for(j=0;j<N;j++) sum += MULT16_16(sinct[j], iptr[j]);
 
+/*    This code is slower on most DSPs which have only 2 accumulators.
+      Plus this this forces truncation to 32 bits and you lose the HW guard bits.
+      I think we can trust the compiler and let it vectorize and/or unroll itself.
+      spx_word32_t accum[4] = {0,0,0,0};
       for(j=0;j<N;j+=4) {
-        accum[0] += sinc[j]*iptr[j];
-        accum[1] += sinc[j+1]*iptr[j+1];
-        accum[2] += sinc[j+2]*iptr[j+2];
-        accum[3] += sinc[j+3]*iptr[j+3];
+        accum[0] += MULT16_16(sinct[j], iptr[j]);
+        accum[1] += MULT16_16(sinct[j+1], iptr[j+1]);
+        accum[2] += MULT16_16(sinct[j+2], iptr[j+2]);
+        accum[3] += MULT16_16(sinct[j+3], iptr[j+3]);
       }
       sum = accum[0] + accum[1] + accum[2] + accum[3];
+*/
+      sum = SATURATE32PSHR(sum, 15, 32767);
 #else
-      sum = inner_product_single(sinc, iptr, N);
+      sum = inner_product_single(sinct, iptr, N);
 #endif
 
-      out[out_stride * out_sample++] = PSHR32(sum, 15);
+      out[out_stride * out_sample++] = sum;
       last_sample += int_advance;
       samp_frac_num += frac_advance;
       if (samp_frac_num >= den_rate)
@@ -381,25 +400,25 @@ static int resampler_basic_direct_double(SpeexResamplerState *st, spx_uint32_t c
    const int frac_advance = st->frac_advance;
    const spx_uint32_t den_rate = st->den_rate;
    double sum;
-   int j;
 
    while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len))
    {
-      const spx_word16_t *sinc = & sinc_table[samp_frac_num*N];
+      const spx_word16_t *sinct = & sinc_table[samp_frac_num*N];
       const spx_word16_t *iptr = & in[last_sample];
 
 #ifndef OVERRIDE_INNER_PRODUCT_DOUBLE
+      int j;
       double accum[4] = {0,0,0,0};
 
       for(j=0;j<N;j+=4) {
-        accum[0] += sinc[j]*iptr[j];
-        accum[1] += sinc[j+1]*iptr[j+1];
-        accum[2] += sinc[j+2]*iptr[j+2];
-        accum[3] += sinc[j+3]*iptr[j+3];
+        accum[0] += sinct[j]*iptr[j];
+        accum[1] += sinct[j+1]*iptr[j+1];
+        accum[2] += sinct[j+2]*iptr[j+2];
+        accum[3] += sinct[j+3]*iptr[j+3];
       }
       sum = accum[0] + accum[1] + accum[2] + accum[3];
 #else
-      sum = inner_product_double(sinc, iptr, N);
+      sum = inner_product_double(sinct, iptr, N);
 #endif
 
       out[out_stride * out_sample++] = PSHR32(sum, 15);
@@ -428,7 +447,6 @@ static int resampler_basic_interpolate_single(SpeexResamplerState *st, spx_uint3
    const int int_advance = st->int_advance;
    const int frac_advance = st->frac_advance;
    const spx_uint32_t den_rate = st->den_rate;
-   int j;
    spx_word32_t sum;
 
    while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len))
@@ -445,6 +463,7 @@ static int resampler_basic_interpolate_single(SpeexResamplerState *st, spx_uint3
 
 
 #ifndef OVERRIDE_INTERPOLATE_PRODUCT_SINGLE
+      int j;
       spx_word32_t accum[4] = {0,0,0,0};
 
       for(j=0;j<N;j++) {
@@ -456,13 +475,14 @@ static int resampler_basic_interpolate_single(SpeexResamplerState *st, spx_uint3
       }
 
       cubic_coef(frac, interp);
-      sum = MULT16_32_Q15(interp[0],accum[0]) + MULT16_32_Q15(interp[1],accum[1]) + MULT16_32_Q15(interp[2],accum[2]) + MULT16_32_Q15(interp[3],accum[3]);
+      sum = MULT16_32_Q15(interp[0],SHR32(accum[0], 1)) + MULT16_32_Q15(interp[1],SHR32(accum[1], 1)) + MULT16_32_Q15(interp[2],SHR32(accum[2], 1)) + MULT16_32_Q15(interp[3],SHR32(accum[3], 1));
+      sum = SATURATE32PSHR(sum, 15, 32767);
 #else
       cubic_coef(frac, interp);
       sum = interpolate_product_single(iptr, st->sinc_table + st->oversample + 4 - offset - 2, N, st->oversample, interp);
 #endif
 
-      out[out_stride * out_sample++] = PSHR32(sum,15);
+      out[out_stride * out_sample++] = sum;
       last_sample += int_advance;
       samp_frac_num += frac_advance;
       if (samp_frac_num >= den_rate)
@@ -490,7 +510,6 @@ static int resampler_basic_interpolate_double(SpeexResamplerState *st, spx_uint3
    const int int_advance = st->int_advance;
    const int frac_advance = st->frac_advance;
    const spx_uint32_t den_rate = st->den_rate;
-   int j;
    spx_word32_t sum;
 
    while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len))
@@ -507,6 +526,7 @@ static int resampler_basic_interpolate_double(SpeexResamplerState *st, spx_uint3
 
 
 #ifndef OVERRIDE_INTERPOLATE_PRODUCT_DOUBLE
+      int j;
       double accum[4] = {0,0,0,0};
 
       for(j=0;j<N;j++) {
@@ -540,11 +560,47 @@ static int resampler_basic_interpolate_double(SpeexResamplerState *st, spx_uint3
 }
 #endif
 
-static void update_filter(SpeexResamplerState *st)
+/* This resampler is used to produce zero output in situations where memory
+   for the filter could not be allocated.  The expected numbers of input and
+   output samples are still processed so that callers failing to check error
+   codes are not surprised, possibly getting into infinite loops. */
+static int resampler_basic_zero(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_word16_t *in, spx_uint32_t *in_len, spx_word16_t *out, spx_uint32_t *out_len)
 {
-   spx_uint32_t old_length;
+   int out_sample = 0;
+   int last_sample = st->last_sample[channel_index];
+   spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
+   const int out_stride = st->out_stride;
+   const int int_advance = st->int_advance;
+   const int frac_advance = st->frac_advance;
+   const spx_uint32_t den_rate = st->den_rate;
 
-   old_length = st->filt_len;
+   while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len))
+   {
+      out[out_stride * out_sample++] = 0;
+      last_sample += int_advance;
+      samp_frac_num += frac_advance;
+      if (samp_frac_num >= den_rate)
+      {
+         samp_frac_num -= den_rate;
+         last_sample++;
+      }
+   }
+
+   st->last_sample[channel_index] = last_sample;
+   st->samp_frac_num[channel_index] = samp_frac_num;
+   return out_sample;
+}
+
+static int update_filter(SpeexResamplerState *st)
+{
+   spx_uint32_t old_length = st->filt_len;
+   spx_uint32_t old_alloc_size = st->mem_alloc_size;
+   int use_direct;
+   spx_uint32_t min_sinc_table_length;
+   spx_uint32_t min_alloc_size;
+
+   st->int_advance = st->num_rate/st->den_rate;
+   st->frac_advance = st->num_rate%st->den_rate;
    st->oversample = quality_map[st->quality].oversample;
    st->filt_len = quality_map[st->quality].base_length;
 
@@ -554,8 +610,8 @@ static void update_filter(SpeexResamplerState *st)
       st->cutoff = quality_map[st->quality].downsample_bandwidth * st->den_rate / st->num_rate;
       /* FIXME: divide the numerator and denominator by a certain amount if they're too large */
       st->filt_len = st->filt_len*st->num_rate / st->den_rate;
-      /* Round down to make sure we have a multiple of 4 */
-      st->filt_len &= (~0x3);
+      /* Round up to make sure we have a multiple of 8 for SSE */
+      st->filt_len = ((st->filt_len-1)&(~0x7))+8;
       if (2*st->den_rate < st->num_rate)
          st->oversample >>= 1;
       if (4*st->den_rate < st->num_rate)
@@ -572,16 +628,35 @@ static void update_filter(SpeexResamplerState *st)
    }
 
    /* Choose the resampling type that requires the least amount of memory */
-   if (st->den_rate <= st->oversample)
+#ifdef RESAMPLE_FULL_SINC_TABLE
+   use_direct = 1;
+   if (INT_MAX/sizeof(spx_word16_t)/st->den_rate < st->filt_len)
+      goto fail;
+#else
+   use_direct = st->filt_len*st->den_rate <= st->filt_len*st->oversample+8
+                && INT_MAX/sizeof(spx_word16_t)/st->den_rate >= st->filt_len;
+#endif
+   if (use_direct)
+   {
+      min_sinc_table_length = st->filt_len*st->den_rate;
+   } else {
+      if ((INT_MAX/sizeof(spx_word16_t)-8)/st->oversample < st->filt_len)
+         goto fail;
+
+      min_sinc_table_length = st->filt_len*st->oversample+8;
+   }
+   if (st->sinc_table_length < min_sinc_table_length)
+   {
+      spx_word16_t *sinc_table = (spx_word16_t *)speex_realloc(st->sinc_table,min_sinc_table_length*sizeof(spx_word16_t));
+      if (!sinc_table)
+         goto fail;
+
+      st->sinc_table = sinc_table;
+      st->sinc_table_length = min_sinc_table_length;
+   }
+   if (use_direct)
    {
       spx_uint32_t i;
-      if (!st->sinc_table)
-         st->sinc_table = (spx_word16_t *)speex_alloc(st->filt_len*st->den_rate*sizeof(spx_word16_t));
-      else if (st->sinc_table_length < st->filt_len*st->den_rate)
-      {
-         st->sinc_table = (spx_word16_t *)speex_realloc(st->sinc_table,st->filt_len*st->den_rate*sizeof(spx_word16_t));
-         st->sinc_table_length = st->filt_len*st->den_rate;
-      }
       for (i=0;i<st->den_rate;i++)
       {
          spx_int32_t j;
@@ -601,13 +676,6 @@ static void update_filter(SpeexResamplerState *st)
       /*fprintf (stderr, "resampler uses direct sinc table and normalised cutoff %f\n", cutoff);*/
    } else {
       spx_int32_t i;
-      if (!st->sinc_table)
-         st->sinc_table = (spx_word16_t *)speex_alloc((st->filt_len*st->oversample+8)*sizeof(spx_word16_t));
-      else if (st->sinc_table_length < st->filt_len*st->oversample+8)
-      {
-         st->sinc_table = (spx_word16_t *)speex_realloc(st->sinc_table,(st->filt_len*st->oversample+8)*sizeof(spx_word16_t));
-         st->sinc_table_length = st->filt_len*st->oversample+8;
-      }
       for (i=-4;i<(spx_int32_t)(st->oversample*st->filt_len+4);i++)
          st->sinc_table[i+4] = sinc(st->cutoff,(i/(float)st->oversample - st->filt_len/2), st->filt_len, quality_map[st->quality].window_func);
 #ifdef FIXED_POINT
@@ -620,43 +688,39 @@ static void update_filter(SpeexResamplerState *st)
 #endif
       /*fprintf (stderr, "resampler uses interpolated sinc table and normalised cutoff %f\n", cutoff);*/
    }
-   st->int_advance = st->num_rate/st->den_rate;
-   st->frac_advance = st->num_rate%st->den_rate;
-
 
    /* Here's the place where we update the filter memory to take into account
       the change in filter length. It's probably the messiest part of the code
       due to handling of lots of corner cases. */
-   if (!st->mem)
+
+   /* Adding buffer_size to filt_len won't overflow here because filt_len
+      could be multiplied by sizeof(spx_word16_t) above. */
+   min_alloc_size = st->filt_len-1 + st->buffer_size;
+   if (min_alloc_size > st->mem_alloc_size)
+   {
+      spx_word16_t *mem;
+      if (INT_MAX/sizeof(spx_word16_t)/st->nb_channels < min_alloc_size)
+          goto fail;
+      else if (!(mem = (spx_word16_t*)speex_realloc(st->mem, st->nb_channels*min_alloc_size * sizeof(*mem))))
+          goto fail;
+
+      st->mem = mem;
+      st->mem_alloc_size = min_alloc_size;
+   }
+   if (!st->started)
    {
       spx_uint32_t i;
-      st->mem_alloc_size = st->filt_len-1 + st->buffer_size;
-      st->mem = (spx_word16_t*)speex_alloc(st->nb_channels*st->mem_alloc_size * sizeof(spx_word16_t));
-      for (i=0;i<st->nb_channels*st->mem_alloc_size;i++)
-         st->mem[i] = 0;
-      /*speex_warning("init filter");*/
-   } else if (!st->started)
-   {
-      spx_uint32_t i;
-      st->mem_alloc_size = st->filt_len-1 + st->buffer_size;
-      st->mem = (spx_word16_t*)speex_realloc(st->mem, st->nb_channels*st->mem_alloc_size * sizeof(spx_word16_t));
       for (i=0;i<st->nb_channels*st->mem_alloc_size;i++)
          st->mem[i] = 0;
       /*speex_warning("reinit filter");*/
    } else if (st->filt_len > old_length)
    {
-      spx_int32_t i;
+      spx_uint32_t i;
       /* Increase the filter length */
       /*speex_warning("increase filter size");*/
-      int old_alloc_size = st->mem_alloc_size;
-      if ((st->filt_len-1 + st->buffer_size) > st->mem_alloc_size)
+      for (i=st->nb_channels;i--;)
       {
-         st->mem_alloc_size = st->filt_len-1 + st->buffer_size;
-         st->mem = (spx_word16_t*)speex_realloc(st->mem, st->nb_channels*st->mem_alloc_size * sizeof(spx_word16_t));
-      }
-      for (i=st->nb_channels-1;i>=0;i--)
-      {
-         spx_int32_t j;
+         spx_uint32_t j;
          spx_uint32_t olen = old_length;
          /*if (st->magic_samples[i])*/
          {
@@ -664,7 +728,7 @@ static void update_filter(SpeexResamplerState *st)
 
             /* FIXME: This is wrong but for now we need it to avoid going over the array bounds */
             olen = old_length + 2*st->magic_samples[i];
-            for (j=old_length-2+st->magic_samples[i];j>=0;j--)
+            for (j=old_length-1+st->magic_samples[i];j--;)
                st->mem[i*st->mem_alloc_size+j+st->magic_samples[i]] = st->mem[i*old_alloc_size+j];
             for (j=0;j<st->magic_samples[i];j++)
                st->mem[i*st->mem_alloc_size+j] = 0;
@@ -705,18 +769,28 @@ static void update_filter(SpeexResamplerState *st)
          st->magic_samples[i] += old_magic;
       }
    }
+   return RESAMPLER_ERR_SUCCESS;
 
+fail:
+   st->resampler_ptr = resampler_basic_zero;
+   /* st->mem may still contain consumed input samples for the filter.
+      Restore filt_len so that filt_len - 1 still points to the position after
+      the last of these samples. */
+   st->filt_len = old_length;
+   return RESAMPLER_ERR_ALLOC_FAILED;
 }
 
- SpeexResamplerState *speex_resampler_init(spx_uint32_t nb_channels, spx_uint32_t in_rate, spx_uint32_t out_rate, int quality, int *err)
+EXPORT SpeexResamplerState *speex_resampler_init(spx_uint32_t nb_channels, spx_uint32_t in_rate, spx_uint32_t out_rate, int quality, int *err)
 {
    return speex_resampler_init_frac(nb_channels, in_rate, out_rate, in_rate, out_rate, quality, err);
 }
 
- SpeexResamplerState *speex_resampler_init_frac(spx_uint32_t nb_channels, spx_uint32_t ratio_num, spx_uint32_t ratio_den, spx_uint32_t in_rate, spx_uint32_t out_rate, int quality, int *err)
+EXPORT SpeexResamplerState *speex_resampler_init_frac(spx_uint32_t nb_channels, spx_uint32_t ratio_num, spx_uint32_t ratio_den, spx_uint32_t in_rate, spx_uint32_t out_rate, int quality, int *err)
 {
    spx_uint32_t i;
    SpeexResamplerState *st;
+   int filter_err;
+
    if (quality > 10 || quality < 0)
    {
       if (err)
@@ -742,16 +816,12 @@ static void update_filter(SpeexResamplerState *st)
    st->in_stride = 1;
    st->out_stride = 1;
 
-#ifdef FIXED_POINT
    st->buffer_size = 160;
-#else
-   st->buffer_size = 160;
-#endif
 
    /* Per channel data */
-   st->last_sample = (spx_int32_t*)speex_alloc(nb_channels*sizeof(int));
-   st->magic_samples = (spx_uint32_t*)speex_alloc(nb_channels*sizeof(int));
-   st->samp_frac_num = (spx_uint32_t*)speex_alloc(nb_channels*sizeof(int));
+   st->last_sample = (spx_int32_t*)speex_alloc(nb_channels*sizeof(spx_int32_t));
+   st->magic_samples = (spx_uint32_t*)speex_alloc(nb_channels*sizeof(spx_uint32_t));
+   st->samp_frac_num = (spx_uint32_t*)speex_alloc(nb_channels*sizeof(spx_uint32_t));
    for (i=0;i<nb_channels;i++)
    {
       st->last_sample[i] = 0;
@@ -762,17 +832,21 @@ static void update_filter(SpeexResamplerState *st)
    speex_resampler_set_quality(st, quality);
    speex_resampler_set_rate_frac(st, ratio_num, ratio_den, in_rate, out_rate);
 
-
-   update_filter(st);
-
-   st->initialised = 1;
+   filter_err = update_filter(st);
+   if (filter_err == RESAMPLER_ERR_SUCCESS)
+   {
+      st->initialised = 1;
+   } else {
+      speex_resampler_destroy(st);
+      st = NULL;
+   }
    if (err)
-      *err = RESAMPLER_ERR_SUCCESS;
+      *err = filter_err;
 
    return st;
 }
 
- void speex_resampler_destroy(SpeexResamplerState *st)
+EXPORT void speex_resampler_destroy(SpeexResamplerState *st)
 {
    speex_free(st->mem);
    speex_free(st->sinc_table);
@@ -829,9 +903,9 @@ static int speex_resampler_magic(SpeexResamplerState *st, spx_uint32_t channel_i
 }
 
 #ifdef FIXED_POINT
- int speex_resampler_process_int(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_int16_t *in, spx_uint32_t *in_len, spx_int16_t *out, spx_uint32_t *out_len)
+EXPORT int speex_resampler_process_int(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_int16_t *in, spx_uint32_t *in_len, spx_int16_t *out, spx_uint32_t *out_len)
 #else
- int speex_resampler_process_float(SpeexResamplerState *st, spx_uint32_t channel_index, const float *in, spx_uint32_t *in_len, float *out, spx_uint32_t *out_len)
+EXPORT int speex_resampler_process_float(SpeexResamplerState *st, spx_uint32_t channel_index, const float *in, spx_uint32_t *in_len, float *out, spx_uint32_t *out_len)
 #endif
 {
    int j;
@@ -866,13 +940,13 @@ static int speex_resampler_magic(SpeexResamplerState *st, spx_uint32_t channel_i
    }
    *in_len -= ilen;
    *out_len -= olen;
-   return RESAMPLER_ERR_SUCCESS;
+   return st->resampler_ptr == resampler_basic_zero ? RESAMPLER_ERR_ALLOC_FAILED : RESAMPLER_ERR_SUCCESS;
 }
 
 #ifdef FIXED_POINT
- int speex_resampler_process_float(SpeexResamplerState *st, spx_uint32_t channel_index, const float *in, spx_uint32_t *in_len, float *out, spx_uint32_t *out_len)
+EXPORT int speex_resampler_process_float(SpeexResamplerState *st, spx_uint32_t channel_index, const float *in, spx_uint32_t *in_len, float *out, spx_uint32_t *out_len)
 #else
- int speex_resampler_process_int(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_int16_t *in, spx_uint32_t *in_len, spx_int16_t *out, spx_uint32_t *out_len)
+EXPORT int speex_resampler_process_int(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_int16_t *in, spx_uint32_t *in_len, spx_int16_t *out, spx_uint32_t *out_len)
 #endif
 {
    int j;
@@ -940,20 +1014,22 @@ static int speex_resampler_magic(SpeexResamplerState *st, spx_uint32_t channel_i
    *in_len -= ilen;
    *out_len -= olen;
 
-   return RESAMPLER_ERR_SUCCESS;
+   return st->resampler_ptr == resampler_basic_zero ? RESAMPLER_ERR_ALLOC_FAILED : RESAMPLER_ERR_SUCCESS;
 }
 
- int speex_resampler_process_interleaved_float(SpeexResamplerState *st, const float *in, spx_uint32_t *in_len, float *out, spx_uint32_t *out_len)
+EXPORT int speex_resampler_process_interleaved_float(SpeexResamplerState *st, const float *in, spx_uint32_t *in_len, float *out, spx_uint32_t *out_len)
 {
    spx_uint32_t i;
    int istride_save, ostride_save;
-   spx_uint32_t bak_len = *out_len;
+   spx_uint32_t bak_out_len = *out_len;
+   spx_uint32_t bak_in_len = *in_len;
    istride_save = st->in_stride;
    ostride_save = st->out_stride;
    st->in_stride = st->out_stride = st->nb_channels;
    for (i=0;i<st->nb_channels;i++)
    {
-      *out_len = bak_len;
+      *out_len = bak_out_len;
+      *in_len = bak_in_len;
       if (in != NULL)
          speex_resampler_process_float(st, i, in+i, in_len, out+i, out_len);
       else
@@ -961,20 +1037,22 @@ static int speex_resampler_magic(SpeexResamplerState *st, spx_uint32_t channel_i
    }
    st->in_stride = istride_save;
    st->out_stride = ostride_save;
-   return RESAMPLER_ERR_SUCCESS;
+   return st->resampler_ptr == resampler_basic_zero ? RESAMPLER_ERR_ALLOC_FAILED : RESAMPLER_ERR_SUCCESS;
 }
 
- int speex_resampler_process_interleaved_int(SpeexResamplerState *st, const spx_int16_t *in, spx_uint32_t *in_len, spx_int16_t *out, spx_uint32_t *out_len)
+EXPORT int speex_resampler_process_interleaved_int(SpeexResamplerState *st, const spx_int16_t *in, spx_uint32_t *in_len, spx_int16_t *out, spx_uint32_t *out_len)
 {
    spx_uint32_t i;
    int istride_save, ostride_save;
-   spx_uint32_t bak_len = *out_len;
+   spx_uint32_t bak_out_len = *out_len;
+   spx_uint32_t bak_in_len = *in_len;
    istride_save = st->in_stride;
    ostride_save = st->out_stride;
    st->in_stride = st->out_stride = st->nb_channels;
    for (i=0;i<st->nb_channels;i++)
    {
-      *out_len = bak_len;
+      *out_len = bak_out_len;
+      *in_len = bak_in_len;
       if (in != NULL)
          speex_resampler_process_int(st, i, in+i, in_len, out+i, out_len);
       else
@@ -982,21 +1060,21 @@ static int speex_resampler_magic(SpeexResamplerState *st, spx_uint32_t channel_i
    }
    st->in_stride = istride_save;
    st->out_stride = ostride_save;
-   return RESAMPLER_ERR_SUCCESS;
+   return st->resampler_ptr == resampler_basic_zero ? RESAMPLER_ERR_ALLOC_FAILED : RESAMPLER_ERR_SUCCESS;
 }
 
- int speex_resampler_set_rate(SpeexResamplerState *st, spx_uint32_t in_rate, spx_uint32_t out_rate)
+EXPORT int speex_resampler_set_rate(SpeexResamplerState *st, spx_uint32_t in_rate, spx_uint32_t out_rate)
 {
    return speex_resampler_set_rate_frac(st, in_rate, out_rate, in_rate, out_rate);
 }
 
- void speex_resampler_get_rate(SpeexResamplerState *st, spx_uint32_t *in_rate, spx_uint32_t *out_rate)
+EXPORT void speex_resampler_get_rate(SpeexResamplerState *st, spx_uint32_t *in_rate, spx_uint32_t *out_rate)
 {
    *in_rate = st->in_rate;
    *out_rate = st->out_rate;
 }
 
- int speex_resampler_set_rate_frac(SpeexResamplerState *st, spx_uint32_t ratio_num, spx_uint32_t ratio_den, spx_uint32_t in_rate, spx_uint32_t out_rate)
+EXPORT int speex_resampler_set_rate_frac(SpeexResamplerState *st, spx_uint32_t ratio_num, spx_uint32_t ratio_den, spx_uint32_t in_rate, spx_uint32_t out_rate)
 {
    spx_uint32_t fact;
    spx_uint32_t old_den;
@@ -1031,17 +1109,17 @@ static int speex_resampler_magic(SpeexResamplerState *st, spx_uint32_t channel_i
    }
 
    if (st->initialised)
-      update_filter(st);
+      return update_filter(st);
    return RESAMPLER_ERR_SUCCESS;
 }
 
- void speex_resampler_get_ratio(SpeexResamplerState *st, spx_uint32_t *ratio_num, spx_uint32_t *ratio_den)
+EXPORT void speex_resampler_get_ratio(SpeexResamplerState *st, spx_uint32_t *ratio_num, spx_uint32_t *ratio_den)
 {
    *ratio_num = st->num_rate;
    *ratio_den = st->den_rate;
 }
 
- int speex_resampler_set_quality(SpeexResamplerState *st, int quality)
+EXPORT int speex_resampler_set_quality(SpeexResamplerState *st, int quality)
 {
    if (quality > 10 || quality < 0)
       return RESAMPLER_ERR_INVALID_ARG;
@@ -1049,46 +1127,46 @@ static int speex_resampler_magic(SpeexResamplerState *st, spx_uint32_t channel_i
       return RESAMPLER_ERR_SUCCESS;
    st->quality = quality;
    if (st->initialised)
-      update_filter(st);
+      return update_filter(st);
    return RESAMPLER_ERR_SUCCESS;
 }
 
- void speex_resampler_get_quality(SpeexResamplerState *st, int *quality)
+EXPORT void speex_resampler_get_quality(SpeexResamplerState *st, int *quality)
 {
    *quality = st->quality;
 }
 
- void speex_resampler_set_input_stride(SpeexResamplerState *st, spx_uint32_t stride)
+EXPORT void speex_resampler_set_input_stride(SpeexResamplerState *st, spx_uint32_t stride)
 {
    st->in_stride = stride;
 }
 
- void speex_resampler_get_input_stride(SpeexResamplerState *st, spx_uint32_t *stride)
+EXPORT void speex_resampler_get_input_stride(SpeexResamplerState *st, spx_uint32_t *stride)
 {
    *stride = st->in_stride;
 }
 
- void speex_resampler_set_output_stride(SpeexResamplerState *st, spx_uint32_t stride)
+EXPORT void speex_resampler_set_output_stride(SpeexResamplerState *st, spx_uint32_t stride)
 {
    st->out_stride = stride;
 }
 
- void speex_resampler_get_output_stride(SpeexResamplerState *st, spx_uint32_t *stride)
+EXPORT void speex_resampler_get_output_stride(SpeexResamplerState *st, spx_uint32_t *stride)
 {
    *stride = st->out_stride;
 }
 
- int speex_resampler_get_input_latency(SpeexResamplerState *st)
+EXPORT int speex_resampler_get_input_latency(SpeexResamplerState *st)
 {
   return st->filt_len / 2;
 }
 
- int speex_resampler_get_output_latency(SpeexResamplerState *st)
+EXPORT int speex_resampler_get_output_latency(SpeexResamplerState *st)
 {
   return ((st->filt_len / 2) * st->den_rate + (st->num_rate >> 1)) / st->num_rate;
 }
 
- int speex_resampler_skip_zeros(SpeexResamplerState *st)
+EXPORT int speex_resampler_skip_zeros(SpeexResamplerState *st)
 {
    spx_uint32_t i;
    for (i=0;i<st->nb_channels;i++)
@@ -1096,15 +1174,21 @@ static int speex_resampler_magic(SpeexResamplerState *st, spx_uint32_t channel_i
    return RESAMPLER_ERR_SUCCESS;
 }
 
- int speex_resampler_reset_mem(SpeexResamplerState *st)
+EXPORT int speex_resampler_reset_mem(SpeexResamplerState *st)
 {
    spx_uint32_t i;
+   for (i=0;i<st->nb_channels;i++)
+   {
+      st->last_sample[i] = 0;
+      st->magic_samples[i] = 0;
+      st->samp_frac_num[i] = 0;
+   }
    for (i=0;i<st->nb_channels*(st->filt_len-1);i++)
       st->mem[i] = 0;
    return RESAMPLER_ERR_SUCCESS;
 }
 
- const char *speex_resampler_strerror(int err)
+EXPORT const char *speex_resampler_strerror(int err)
 {
    switch (err)
    {

codecs/speex/speex_resampler.h

@@ -39,8 +39,6 @@
 #ifndef SPEEX_RESAMPLER_H
 #define SPEEX_RESAMPLER_H
 
-#define OUTSIDE_SPEEX
-
 #ifdef OUTSIDE_SPEEX
 
 /********* WARNING: MENTAL SANITY ENDS HERE *************/
@@ -48,7 +46,7 @@
 /* If the resampler is defined outside of Speex, we change the symbol names so that
    there won't be any clash if linking with Speex later on. */
 
-#define RANDOM_PREFIX ast
+#define RANDOM_PREFIX speex
 #ifndef RANDOM_PREFIX
 #error "Please define RANDOM_PREFIX (above) to something specific to your project to prevent symbol name clashes"
 #endif

configs/basic-pbx/extensions.conf

@@ -36,6 +36,20 @@ exten = 2565551234,1,Verbose(1, "User ${CALLERID(num)} dialed into remote voicem
  same = n,VoiceMailMain(example)
  same = n,Hangup()
 
+; Extension to queue for sales.
+; The queue has a 300 second timeout.
+exten = 2565551200,1,Verbose(1, "User ${CALLERID(num)} dialed the sales queue.")
+ same = n,Answer()
+ same = n,Queue(sales,,,,300)
+ same = n,Goto(operator,1)
+
+; Extension to queue for a customer advocate.
+; The queue has a 1200 second timeout.
+exten = 2565551250,1,Verbose(1, "User ${CALLERID(num)} dialed the customer advocate queue.")
+ same = n,Answer()
+ same = n,Queue(customer_advocate,,,,1200)
+ same = n,Goto(operator,1)
+
 [Dialing-Errors]
 ; Handle any extensions dialed internally that don't otherwise exist.
 ; Comment out or remove this extension if you would rather have the calls

configs/basic-pbx/modules.conf

@@ -14,6 +14,7 @@ load = app_verbose.so
 load = app_voicemail.so
 load = app_directory.so
 load = app_confbridge.so
+load = app_queue.so
 
 ; Bridging
 

configs/basic-pbx/queues.conf

@@ -0,0 +1,19 @@
+[general]
+monitor-type = MixMonitor
+
+;========================Sales Queue ==
+; Calls all sales persons in a ring-all fashion
+[sales]
+strategy=ringall
+member => PJSIP/1109 ; Terry Jules - Director of Sales
+member => PJSIP/1105 ; Garnet Claude - Sales Associate
+member => PJSIP/1112 ; Franny Ocean - Sales Associate
+
+;===================== Customer Advocate Queue ==
+; Calls all customer advocates in a ring-all fashion
+[customer_advocate]
+strategy=ringall
+member => PJSIP/1101 ; Maria Berny - Director of Customer Experience
+member => PJSIP/1115 ; Dusty Williams - Customer Advocate
+member => PJSIP/1102 ; Tommy Briar - Customer Advocate
+

configs/samples/extconfig.conf.sample

@@ -95,6 +95,7 @@
 ;queue_rules => odbc,asterisk
 ;acls => odbc,asterisk
 ;musiconhold => mysql,general
+;musiconhold_entry => mysql,general
 ;queue_log => mysql,general
 ;
 ;

configs/samples/musiconhold.conf.sample

@@ -13,6 +13,7 @@
 ; valid mode options:
 ; files		-- read files from a directory in any Asterisk supported
 ;		   media format
+; playlist	-- provide a fixed list of filenames or URLs to play
 ; quietmp3 	-- default
 ; mp3 		-- loud
 ; mp3nb		-- unbuffered
@@ -44,6 +45,22 @@
 ; this, res_musiconhold will skip the files it is not able to
 ; understand when it loads.
 ;
+; =========
+; Playlist (native) music on hold
+; =========
+;
+; This mode is similar to 'files' mode in that it plays through a list
+; of files, but instead of scanning a directory the files are
+; explicitly configured using one or more 'entry' options.
+;
+; Each entry must be one of:
+;
+;   * An absolute path to the file to be played, without an extension.
+;   * A URL
+;
+; The entries are played in the order in which they appear in the
+; configuration. The 'sort' option is not used for this mode.
+;
 
 [default]
 mode=files
@@ -71,6 +88,12 @@ directory=moh
 ;directory=moh
 ;sort=alpha     ; Sort the files in alphabetical order.
 
+;[sales-queue-hold]
+;mode=playlist
+;entry=/var/lib/asterisk/sounds/en/yourcallisimportant
+;entry=http://example.local/sales-queue-hold-music.ulaw
+;entry=/var/lib/asterisk/moh/macroform-robot_dity
+
 ; =========
 ; Other (non-native) playback methods
 ; =========

configure

@@ -1350,6 +1350,7 @@ infodir
 docdir
 oldincludedir
 includedir
+runstatedir
 localstatedir
 sharedstatedir
 sysconfdir
@@ -1538,6 +1539,7 @@ datadir='${datarootdir}'
 sysconfdir='${prefix}/etc'
 sharedstatedir='${prefix}/com'
 localstatedir='${prefix}/var'
+runstatedir='${localstatedir}/run'
 includedir='${prefix}/include'
 oldincludedir='/usr/include'
 docdir='${datarootdir}/doc/${PACKAGE_TARNAME}'
@@ -1790,6 +1792,15 @@ do
   | -silent | --silent | --silen | --sile | --sil)
     silent=yes ;;
 
+  -runstatedir | --runstatedir | --runstatedi | --runstated \
+  | --runstate | --runstat | --runsta | --runst | --runs \
+  | --run | --ru | --r)
+    ac_prev=runstatedir ;;
+  -runstatedir=* | --runstatedir=* | --runstatedi=* | --runstated=* \
+  | --runstate=* | --runstat=* | --runsta=* | --runst=* | --runs=* \
+  | --run=* | --ru=* | --r=*)
+    runstatedir=$ac_optarg ;;
+
   -sbindir | --sbindir | --sbindi | --sbind | --sbin | --sbi | --sb)
     ac_prev=sbindir ;;
   -sbindir=* | --sbindir=* | --sbindi=* | --sbind=* | --sbin=* \
@@ -1927,7 +1938,7 @@ fi
 for ac_var in	exec_prefix prefix bindir sbindir libexecdir datarootdir \
 		datadir sysconfdir sharedstatedir localstatedir includedir \
 		oldincludedir docdir infodir htmldir dvidir pdfdir psdir \
-		libdir localedir mandir
+		libdir localedir mandir runstatedir
 do
   eval ac_val=\$$ac_var
   # Remove trailing slashes.
@@ -2080,6 +2091,7 @@ Fine tuning of the installation directories:
   --sysconfdir=DIR        read-only single-machine data [PREFIX/etc]
   --sharedstatedir=DIR    modifiable architecture-independent data [PREFIX/com]
   --localstatedir=DIR     modifiable single-machine data [PREFIX/var]
+  --runstatedir=DIR       modifiable per-process data [LOCALSTATEDIR/run]
   --libdir=DIR            object code libraries [EPREFIX/lib]
   --includedir=DIR        C header files [PREFIX/include]
   --oldincludedir=DIR     C header files for non-gcc [/usr/include]
@@ -14946,7 +14958,7 @@ else
     We can't simply define LARGE_OFF_T to be 9223372036854775807,
     since some C++ compilers masquerading as C compilers
     incorrectly reject 9223372036854775807.  */
-#define LARGE_OFF_T (((off_t) 1 << 62) - 1 + ((off_t) 1 << 62))
+#define LARGE_OFF_T ((((off_t) 1 << 31) << 31) - 1 + (((off_t) 1 << 31) << 31))
   int off_t_is_large[(LARGE_OFF_T % 2147483629 == 721
 		       && LARGE_OFF_T % 2147483647 == 1)
 		      ? 1 : -1];
@@ -14992,7 +15004,7 @@ else
     We can't simply define LARGE_OFF_T to be 9223372036854775807,
     since some C++ compilers masquerading as C compilers
     incorrectly reject 9223372036854775807.  */
-#define LARGE_OFF_T (((off_t) 1 << 62) - 1 + ((off_t) 1 << 62))
+#define LARGE_OFF_T ((((off_t) 1 << 31) << 31) - 1 + (((off_t) 1 << 31) << 31))
   int off_t_is_large[(LARGE_OFF_T % 2147483629 == 721
 		       && LARGE_OFF_T % 2147483647 == 1)
 		      ? 1 : -1];
@@ -15016,7 +15028,7 @@ rm -f core conftest.err conftest.$ac_objext conftest.$ac_ext
     We can't simply define LARGE_OFF_T to be 9223372036854775807,
     since some C++ compilers masquerading as C compilers
     incorrectly reject 9223372036854775807.  */
-#define LARGE_OFF_T (((off_t) 1 << 62) - 1 + ((off_t) 1 << 62))
+#define LARGE_OFF_T ((((off_t) 1 << 31) << 31) - 1 + (((off_t) 1 << 31) << 31))
   int off_t_is_large[(LARGE_OFF_T % 2147483629 == 721
 		       && LARGE_OFF_T % 2147483647 == 1)
 		      ? 1 : -1];
@@ -15061,7 +15073,7 @@ else
     We can't simply define LARGE_OFF_T to be 9223372036854775807,
     since some C++ compilers masquerading as C compilers
     incorrectly reject 9223372036854775807.  */
-#define LARGE_OFF_T (((off_t) 1 << 62) - 1 + ((off_t) 1 << 62))
+#define LARGE_OFF_T ((((off_t) 1 << 31) << 31) - 1 + (((off_t) 1 << 31) << 31))
   int off_t_is_large[(LARGE_OFF_T % 2147483629 == 721
 		       && LARGE_OFF_T % 2147483647 == 1)
 		      ? 1 : -1];
@@ -15085,7 +15097,7 @@ rm -f core conftest.err conftest.$ac_objext conftest.$ac_ext
     We can't simply define LARGE_OFF_T to be 9223372036854775807,
     since some C++ compilers masquerading as C compilers
     incorrectly reject 9223372036854775807.  */
-#define LARGE_OFF_T (((off_t) 1 << 62) - 1 + ((off_t) 1 << 62))
+#define LARGE_OFF_T ((((off_t) 1 << 31) << 31) - 1 + (((off_t) 1 << 31) << 31))
   int off_t_is_large[(LARGE_OFF_T % 2147483629 == 721
 		       && LARGE_OFF_T % 2147483647 == 1)
 		      ? 1 : -1];
@@ -16385,6 +16397,8 @@ main ()
     if (*(data + i) != *(data3 + i))
       return 14;
   close (fd);
+  free (data);
+  free (data3);
   return 0;
 }
 _ACEOF
@@ -22958,6 +22972,7 @@ fi
 fi
 
 
+
 		if test "x${PBX_MYSQLCLIENT}" != "x1" -a "${USE_MYSQLCLIENT}" != "no"; then
 		PBX_MYSQLCLIENT=0
 		if test -n "$ac_tool_prefix"; then
@@ -23077,6 +23092,61 @@ $as_echo "#define HAVE_MYSQLCLIENT 1" >>confdefs.h
 	fi
 
 
+if test "${PBX_MYSQLCLIENT}" = 1; then
+   { $as_echo "$as_me:${as_lineno-$LINENO}: checking for MySQL client bool support" >&5
+$as_echo_n "checking for MySQL client bool support... " >&6; }
+   cat confdefs.h - <<_ACEOF >conftest.$ac_ext
+/* end confdefs.h.  */
+#include <mysql/mysql.h>
+int
+main ()
+{
+bool test = 1;
+  ;
+  return 0;
+}
+_ACEOF
+if ac_fn_c_try_link "$LINENO"; then :
+  { $as_echo "$as_me:${as_lineno-$LINENO}: result: yes" >&5
+$as_echo "yes" >&6; }
+
+$as_echo "#define HAVE_MYSQLCLIENT_BOOL 1" >>confdefs.h
+
+else
+  { $as_echo "$as_me:${as_lineno-$LINENO}: result: no" >&5
+$as_echo "no" >&6; }
+
+fi
+rm -f core conftest.err conftest.$ac_objext \
+    conftest$ac_exeext conftest.$ac_ext
+   { $as_echo "$as_me:${as_lineno-$LINENO}: checking for MySQL client my_bool support" >&5
+$as_echo_n "checking for MySQL client my_bool support... " >&6; }
+   cat confdefs.h - <<_ACEOF >conftest.$ac_ext
+/* end confdefs.h.  */
+#include <mysql/mysql.h>
+int
+main ()
+{
+my_bool test = 1;
+  ;
+  return 0;
+}
+_ACEOF
+if ac_fn_c_try_link "$LINENO"; then :
+  { $as_echo "$as_me:${as_lineno-$LINENO}: result: yes" >&5
+$as_echo "yes" >&6; }
+
+$as_echo "#define HAVE_MYSQLCLIENT_MY_BOOL 1" >>confdefs.h
+
+else
+  { $as_echo "$as_me:${as_lineno-$LINENO}: result: no" >&5
+$as_echo "no" >&6; }
+
+fi
+rm -f core conftest.err conftest.$ac_objext \
+    conftest$ac_exeext conftest.$ac_ext
+fi
+
 
 if test "x${PBX_NBS}" != "x1" -a "${USE_NBS}" != "no"; then
    pbxlibdir=""

configure.ac

@@ -2279,8 +2279,28 @@ if test "${PBX_MISDN}" = 1; then
    AC_CHECK_MEMBER([Q931_info_t.redirect_dn], [], [PBX_MISDN=0], [#include <mISDNuser/mISDNlib.h>])
 fi
 
+
 AST_EXT_TOOL_CHECK([MYSQLCLIENT], [mysql_config])
 
+if test "${PBX_MYSQLCLIENT}" = 1; then
+   AC_MSG_CHECKING(for MySQL client bool support)
+   AC_LINK_IFELSE(
+           [AC_LANG_PROGRAM([#include <mysql/mysql.h>],
+                   [bool test = 1;])],
+           AC_MSG_RESULT(yes)
+           AC_DEFINE(HAVE_MYSQLCLIENT_BOOL, 1, [Define to 1 if mysql/mysql.h has bool defined.]),
+           AC_MSG_RESULT(no)
+   )
+   AC_MSG_CHECKING(for MySQL client my_bool support)
+   AC_LINK_IFELSE(
+           [AC_LANG_PROGRAM([#include <mysql/mysql.h>],
+                   [my_bool test = 1;])],
+           AC_MSG_RESULT(yes)
+           AC_DEFINE(HAVE_MYSQLCLIENT_MY_BOOL, 1, [Define to 1 if mysql/mysql.h has my_bool defined.]),
+           AC_MSG_RESULT(no)
+   )
+fi
+
 AST_EXT_LIB_CHECK([NBS], [nbs], [nbs_connect], [nbs.h])
 
 AST_EXT_TOOL_CHECK([NEON], [neon-config])

contrib/ast-db-manage/config/versions/fbb7766f17bc_add_playlist_to_moh.py

@@ -0,0 +1,54 @@
+"""add playlist to moh
+
+Revision ID: fbb7766f17bc
+Revises: 3a094a18e75b
+Create Date: 2019-09-18 10:24:18.731798
+
+"""
+
+# revision identifiers, used by Alembic.
+revision = 'fbb7766f17bc'
+down_revision = '3a094a18e75b'
+
+from alembic import op
+import sqlalchemy as sa
+
+
+def enum_update(table_name, column_name, enum_name, enum_values):
+    if op.get_context().bind.dialect.name != 'postgresql':
+        if op.get_context().bind.dialect.name == 'mssql':
+            op.drop_constraint('ck_musiconhold_mode_moh_mode_values', 'musiconhold')
+        op.alter_column(table_name, column_name,
+                        type_=sa.Enum(*enum_values, name=enum_name))
+        return
+
+    # Postgres requires a few more steps
+    tmp = enum_name + '_tmp'
+
+    op.execute('ALTER TYPE ' + enum_name + ' RENAME TO ' + tmp)
+
+    updated = sa.Enum(*enum_values, name=enum_name)
+    updated.create(op.get_bind(), checkfirst=False)
+
+    op.execute('ALTER TABLE ' + table_name + ' ALTER COLUMN ' + column_name +
+               ' TYPE ' + enum_name + ' USING mode::text::' + enum_name)
+
+    op.execute('DROP TYPE ' + tmp)
+
+
+def upgrade():
+    op.create_table(
+        'musiconhold_entry',
+        sa.Column('name', sa.String(80), primary_key=True, nullable=False),
+        sa.Column('position', sa.Integer, primary_key=True, nullable=False),
+        sa.Column('entry', sa.String(1024), nullable=False)
+    )
+    op.create_foreign_key('fk_musiconhold_entry_name_musiconhold', 'musiconhold_entry', 'musiconhold', ['name'], ['name'])
+    enum_update('musiconhold', 'mode', 'moh_mode_values',
+                ['custom', 'files', 'mp3nb', 'quietmp3nb', 'quietmp3', 'playlist'])
+
+
+def downgrade():
+    enum_update('musiconhold', 'mode', 'moh_mode_values',
+                ['custom', 'files', 'mp3nb', 'quietmp3nb', 'quietmp3'])
+    op.drop_table('musiconhold_entry')

contrib/realtime/mysql/mysql_cdr.sql

@@ -0,0 +1,41 @@
+CREATE TABLE alembic_version (
+    version_num VARCHAR(32) NOT NULL, 
+    CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
+);
+
+-- Running upgrade  -> 210693f3123d
+
+CREATE TABLE cdr (
+    accountcode VARCHAR(20), 
+    src VARCHAR(80), 
+    dst VARCHAR(80), 
+    dcontext VARCHAR(80), 
+    clid VARCHAR(80), 
+    channel VARCHAR(80), 
+    dstchannel VARCHAR(80), 
+    lastapp VARCHAR(80), 
+    lastdata VARCHAR(80), 
+    start DATETIME, 
+    answer DATETIME, 
+    end DATETIME, 
+    duration INTEGER, 
+    billsec INTEGER, 
+    disposition VARCHAR(45), 
+    amaflags VARCHAR(45), 
+    userfield VARCHAR(256), 
+    uniqueid VARCHAR(150), 
+    linkedid VARCHAR(150), 
+    peeraccount VARCHAR(20), 
+    sequence INTEGER
+);
+
+INSERT INTO alembic_version (version_num) VALUES ('210693f3123d');
+
+-- Running upgrade 210693f3123d -> 54cde9847798
+
+ALTER TABLE cdr MODIFY accountcode VARCHAR(80) NULL;
+
+ALTER TABLE cdr MODIFY peeraccount VARCHAR(80) NULL;
+
+UPDATE alembic_version SET version_num='54cde9847798' WHERE alembic_version.version_num = '210693f3123d';
+

contrib/realtime/mysql/mysql_voicemail.sql

@@ -0,0 +1,35 @@
+CREATE TABLE alembic_version (
+    version_num VARCHAR(32) NOT NULL, 
+    CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
+);
+
+-- Running upgrade  -> a2e9769475e
+
+CREATE TABLE voicemail_messages (
+    dir VARCHAR(255) NOT NULL, 
+    msgnum INTEGER NOT NULL, 
+    context VARCHAR(80), 
+    macrocontext VARCHAR(80), 
+    callerid VARCHAR(80), 
+    origtime INTEGER, 
+    duration INTEGER, 
+    recording BLOB, 
+    flag VARCHAR(30), 
+    category VARCHAR(30), 
+    mailboxuser VARCHAR(30), 
+    mailboxcontext VARCHAR(30), 
+    msg_id VARCHAR(40)
+);
+
+ALTER TABLE voicemail_messages ADD CONSTRAINT voicemail_messages_dir_msgnum PRIMARY KEY (dir, msgnum);
+
+CREATE INDEX voicemail_messages_dir ON voicemail_messages (dir);
+
+INSERT INTO alembic_version (version_num) VALUES ('a2e9769475e');
+
+-- Running upgrade a2e9769475e -> 39428242f7f5
+
+ALTER TABLE voicemail_messages MODIFY recording BLOB(4294967295) NULL;
+
+UPDATE alembic_version SET version_num='39428242f7f5' WHERE alembic_version.version_num = 'a2e9769475e';
+

contrib/realtime/postgresql/postgresql_cdr.sql

@@ -0,0 +1,45 @@
+BEGIN;
+
+CREATE TABLE alembic_version (
+    version_num VARCHAR(32) NOT NULL, 
+    CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
+);
+
+-- Running upgrade  -> 210693f3123d
+
+CREATE TABLE cdr (
+    accountcode VARCHAR(20), 
+    src VARCHAR(80), 
+    dst VARCHAR(80), 
+    dcontext VARCHAR(80), 
+    clid VARCHAR(80), 
+    channel VARCHAR(80), 
+    dstchannel VARCHAR(80), 
+    lastapp VARCHAR(80), 
+    lastdata VARCHAR(80), 
+    start TIMESTAMP WITHOUT TIME ZONE, 
+    answer TIMESTAMP WITHOUT TIME ZONE, 
+    "end" TIMESTAMP WITHOUT TIME ZONE, 
+    duration INTEGER, 
+    billsec INTEGER, 
+    disposition VARCHAR(45), 
+    amaflags VARCHAR(45), 
+    userfield VARCHAR(256), 
+    uniqueid VARCHAR(150), 
+    linkedid VARCHAR(150), 
+    peeraccount VARCHAR(20), 
+    sequence INTEGER
+);
+
+INSERT INTO alembic_version (version_num) VALUES ('210693f3123d');
+
+-- Running upgrade 210693f3123d -> 54cde9847798
+
+ALTER TABLE cdr ALTER COLUMN accountcode TYPE VARCHAR(80);
+
+ALTER TABLE cdr ALTER COLUMN peeraccount TYPE VARCHAR(80);
+
+UPDATE alembic_version SET version_num='54cde9847798' WHERE alembic_version.version_num = '210693f3123d';
+
+COMMIT;
+

contrib/realtime/postgresql/postgresql_voicemail.sql

@@ -0,0 +1,39 @@
+BEGIN;
+
+CREATE TABLE alembic_version (
+    version_num VARCHAR(32) NOT NULL, 
+    CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
+);
+
+-- Running upgrade  -> a2e9769475e
+
+CREATE TABLE voicemail_messages (
+    dir VARCHAR(255) NOT NULL, 
+    msgnum INTEGER NOT NULL, 
+    context VARCHAR(80), 
+    macrocontext VARCHAR(80), 
+    callerid VARCHAR(80), 
+    origtime INTEGER, 
+    duration INTEGER, 
+    recording BYTEA, 
+    flag VARCHAR(30), 
+    category VARCHAR(30), 
+    mailboxuser VARCHAR(30), 
+    mailboxcontext VARCHAR(30), 
+    msg_id VARCHAR(40)
+);
+
+ALTER TABLE voicemail_messages ADD CONSTRAINT voicemail_messages_dir_msgnum PRIMARY KEY (dir, msgnum);
+
+CREATE INDEX voicemail_messages_dir ON voicemail_messages (dir);
+
+INSERT INTO alembic_version (version_num) VALUES ('a2e9769475e');
+
+-- Running upgrade a2e9769475e -> 39428242f7f5
+
+ALTER TABLE voicemail_messages ALTER COLUMN recording TYPE BYTEA;
+
+UPDATE alembic_version SET version_num='39428242f7f5' WHERE alembic_version.version_num = 'a2e9769475e';
+
+COMMIT;
+

doc/CHANGES-staging/app_attended_transfer.txt

@@ -1,3 +0,0 @@
-Subject: AttendedTransfer
-
-A new application, this will queue up attended transfer to the given extension.

doc/CHANGES-staging/app_blind_transfer.txt

@@ -1,4 +0,0 @@
-Subject: BlindTransfer
-
-A new application, this will redirect all channels currently
-bridged to the caller channel to the specified destination.

doc/CHANGES-staging/app_confbridge_remb_behavior_all.txt

@@ -1,7 +0,0 @@
-Subject: ConfBridge
-
-Add "average_all", "highest_all", and "lowest_all" values for
-the remb_behavior option. These values operate on a bridge
-level instead of a per-source level. This means that a single
-REMB value is calculated and sent to every sender, instead of
-a REMB value that is unique for the specific sender..

doc/CHANGES-staging/app_dial_ringtime_progresstime.txt

@@ -1,12 +0,0 @@
-Subject: Dial
-
-Add RINGTIME and RINGTIME_MS variables containing respectively seconds and
-milliseconds between creation of the dialing channel and receiving the first
-RINGING signal
-
-Add PROGRESSTIME and PROGRESSTIME_MS variables analogous to the above with respect to
-the PROGRESS signal. Shorter of these two times should be equivalent to
-the PDD (Post Dial Delay) value
-
-Add DIALEDTIME_MS and ANSWEREDTIME_MS variables to get millisecond resolution
-versions of DIALEDTIME and ANSWEREDTIME

doc/CHANGES-staging/app_readexten_pound.txt

@@ -1,3 +0,0 @@
-Subject: ReadExten
-
-Add 'p' option to stop reading extension if user presses '#' key.

doc/CHANGES-staging/bridging_stasis_cache.txt

@@ -1,36 +0,0 @@
-Subject: Bridging
-Master-Only: true
-
-The bridging core no longer uses the stasis cache for bridge
-snapshots.  The latest bridge snapshot is now stored on the
-ast_bridge structure itself.
-
-The following APIs are no longer available since the stasis cache
-is no longer used:
-  ast_bridge_topic_cached()
-  ast_bridge_topic_all_cached()
-
-A topic pool is now used for individual bridge topics.
-
-The ast_bridge_cache() function was removed since there's no
-longer a separate container of snapshots.
-
-A new function "ast_bridges()" was created to retrieve the
-container of all bridges.  Users formerly calling
-ast_bridge_cache() can use the new function to iterate over
-bridges and retrieve the latest snapshot directly from the
-bridge.
-
-The ast_bridge_snapshot_get_latest() function was renamed to
-ast_bridge_get_snapshot_by_uniqueid().
-
-A new function "ast_bridge_get_snapshot()" was created to retrieve
-the bridge snapshot directly from the bridge structure.
-
-The ast_bridge_topic_all() function now returns a normal topic
-not a cached one so you can't use stasis cache functions on it
-either.
-
-The ast_bridge_snapshot_type() stasis message now has the
-ast_bridge_snapshot_update structure as it's data.  It contains
-the last snapshot and the new one.

doc/CHANGES-staging/chan_sip_deprecated.txt

@@ -1,7 +0,0 @@
-Subject: chan_sip
-Master-Only: true
-
-The chan_sip module is now deprecated, users should migrate to the
-replacement module chan_pjsip.  See guides at the Asterisk Wiki:
-  https://wiki.asterisk.org/wiki/x/tAHOAQ
-  https://wiki.asterisk.org/wiki/x/hYCLAQ

doc/CHANGES-staging/channels_stasis_cache.txt

@@ -1,16 +0,0 @@
-Subject: Channels
-Master-Only: true
-
-The core no longer uses the stasis cache for channels snapshots.
-The following APIs are no longer available:
-    ast_channel_topic_cached()
-    ast_channel_topic_all_cached()
-The ast_channel_cache_all() and ast_channel_cache_by_name() functions
-now returns an ao2_container of ast_channel_snapshots rather than a
-container of stasis_messages therefore you can't call stasis_cache
-functions on it.
-The ast_channel_topic_all() function now returns a normal topic,
-not a cached one so you can't use stasis cache functions on it either.
-The ast_channel_snapshot_type() stasis message now has the
-ast_channel_snapshot_update structure as it's data.
-ast_channel_snapshot_get_latest() still returns the latest snapshot.

doc/CHANGES-staging/pbx_dundi_ipv6.txt

@@ -1,3 +0,0 @@
-Subject: pbx_dundi
-
-The DUNDi PBX module now supports IPv4/IPv6 dual binding.

doc/CHANGES-staging/res_pjsip_add_norefersub_global_config.txt

@@ -1,13 +0,0 @@
-Subject: res_pjsip
-
-Added a new PJSIP global setting called norefersub.
-Default is true to keep support working as before.
-
-res_pjsip_refer configures PJSIP norefersub capability accordingly.
-
-Checks the PJSIP global setting value.
-If it is true (default) it adds the norefersub capability to PJSIP.
-If it is false (disabled) it does not add the norefersub capability
-to PJSIP.
-
-This is useful for Cisco switches that do not follow RFC4488.

doc/CHANGES-staging/res_rtp_asterisk_dtls_fragmentation.txt

@@ -1,5 +0,0 @@
-Subject: res_rtp_asterisk
-
-DTLS packets will now be fragmented according to the MTU as set in rtp.conf. This
-allows larger certificates to be used for the DTLS negotiation. By default this value
-is 1200.

doc/CHANGES-staging/rtp_ice_include_local_address.txt

@@ -1,5 +0,0 @@
-Subject: RTP/ICE
-
-You can now indicate that you'd like an ice_host_candidate's local address
-to be published as well as the mapped address.  See the sample rtp.conf
-for more information.

doc/UPGRADE-staging/Build_all_3_app_voicemail_variants_at_the_same_time.txt

@@ -1,9 +0,0 @@
-Subject: app_voicemail.c
-
-The "Voicemail Build Options" section of menuselect has been removed along with
-the FILE_STORAGE, ODBC_STORAGE and IMAP_STORAGE menuselect options.  All 3 variants
-of the voicemail app can now be built at the same by enabling app_voicemail,
-app_voicemail_imap, and app_voicemail_odbc under the "Applications" section.
-By default, only app_voicemail is enabled.  Also, the modules.conf sample has
-been updated to "noload" app_voicemail_imap and app_voicemail_odbc should they
-all be built.  Packagers must update their build scripts appropriately.

doc/UPGRADE-staging/applications_jabberstatus.txt

@@ -1,4 +0,0 @@
-Subject: Applications
-Master-Only: true
-
-The JabberStatus application, deprecated in Asterisk 12, has been removed.

doc/UPGRADE-staging/bridging_stasis_cache.txt

@@ -1,36 +0,0 @@
-Subject: Bridging
-Master-Only: true
-
-The bridging core no longer uses the stasis cache for bridge
-snapshots.  The latest bridge snapshot is now stored on the
-ast_bridge structure itself.
-
-The following APIs are no longer available since the stasis cache
-is no longer used:
-  ast_bridge_topic_cached()
-  ast_bridge_topic_all_cached()
-
-A topic pool is now used for individual bridge topics.
-
-The ast_bridge_cache() function was removed since there's no
-longer a separate container of snapshots.
-
-A new function "ast_bridges()" was created to retrieve the
-container of all bridges.  Users formerly calling
-ast_bridge_cache() can use the new function to iterate over
-bridges and retrieve the latest snapshot directly from the
-bridge.
-
-The ast_bridge_snapshot_get_latest() function was renamed to
-ast_bridge_get_snapshot_by_uniqueid().
-
-A new function "ast_bridge_get_snapshot()" was created to retrieve
-the bridge snapshot directly from the bridge structure.
-
-The ast_bridge_topic_all() function now returns a normal topic
-not a cached one so you can't use stasis cache functions on it
-either.
-
-The ast_bridge_snapshot_type() stasis message now has the
-ast_bridge_snapshot_update structure as it's data.  It contains
-the last snapshot and the new one.

doc/UPGRADE-staging/chan_pjsip_refer_fix.txt

@@ -1,5 +0,0 @@
-Subject: chan_pjsip
-Subject: Core
-
-res_pjsip_pubsub is now required so call transfer progress can be monitored
-and reported in the channel variable TRANSFERSTATUS.

doc/UPGRADE-staging/chan_sip_deprecated.txt

@@ -1,7 +0,0 @@
-Subject: chan_sip
-Master-Only: true
-
-The chan_sip module is now deprecated, users should migrate to the
-replacement module chan_pjsip.  See guides at the Asterisk Wiki:
-  https://wiki.asterisk.org/wiki/x/tAHOAQ
-  https://wiki.asterisk.org/wiki/x/hYCLAQ

doc/UPGRADE-staging/channels_stasis_cache.txt

@@ -1,16 +0,0 @@
-Subject: Channels
-Master-Only: true
-
-The core no longer uses the stasis cache for channels snapshots.
-The following APIs are no longer available:
-    ast_channel_topic_cached()
-    ast_channel_topic_all_cached()
-The ast_channel_cache_all() and ast_channel_cache_by_name() functions
-now returns an ao2_container of ast_channel_snapshots rather than a
-container of stasis_messages therefore you can't call stasis_cache
-functions on it.
-The ast_channel_topic_all() function now returns a normal topic,
-not a cached one so you can't use stasis cache functions on it either.
-The ast_channel_snapshot_type() stasis message now has the
-ast_channel_snapshot_update structure as it's data.
-ast_channel_snapshot_get_latest() still returns the latest snapshot.

doc/UPGRADE-staging/func_callerid_callerpres.txt

@@ -1,5 +0,0 @@
-Subject: func_callerid
-Master-Only: true
-
-The CALLERPRES() dialplan function, deprecated in Asterisk 1.8, has been
-removed.

doc/UPGRADE-staging/install_headers.txt

@@ -1,7 +0,0 @@
-Subject: Build
-Master-Only: true
-
-Asterisk headers are no longer installed and uninstalled automatically when
-performing a "make install" or a "make uninstall".  To install/uninstall the
-headers, use "make install-headers" and "make uninstall-headers".  The headers
-also continue to be uninstalled when performing a "make uninstall-all".

doc/UPGRADE-staging/res_parking_parkingslot.txt

@@ -1,5 +0,0 @@
-Subject: res_parking
-Master-Only: true
-
-The PARKINGSLOT channel variable, deprecated in Asterisk 12 in favor of the
-PARKING_SPACE channel variable, will no longer be set.

doc/UPGRADE-staging/res_xmpp_jabberstatus.txt

@@ -1,4 +0,0 @@
-Subject: res_xmpp
-Master-Only: true
-
-The JabberStatus application, deprecated in Asterisk 12, has been removed.

formats/format_g726.c

@@ -199,7 +199,7 @@ static off_t g726_tell(struct ast_filestream *fs)
 	return ftello(fs->f) << 1;
 }
 
-static struct ast_format_def f[] = {
+static struct ast_format_def f_def[] = {
 	{
 		.name = "g726-40",
 		.exts = "g726-40",
@@ -259,9 +259,9 @@ static int unload_module(void)
 {
 	int i;
 
-	for (i = 0; f[i].desc_size ; i++) {
-		if (ast_format_def_unregister(f[i].name))
-			ast_log(LOG_WARNING, "Failed to unregister format %s.\n", f[i].name);
+	for (i = 0; f_def[i].desc_size ; i++) {
+		if (ast_format_def_unregister(f_def[i].name))
+			ast_log(LOG_WARNING, "Failed to unregister format %s.\n", f_def[i].name);
 	}
 	return(0);
 }
@@ -270,10 +270,10 @@ static int load_module(void)
 {
 	int i;
 
-	for (i = 0; f[i].desc_size ; i++) {
-		f[i].format = ast_format_g726;
-		if (ast_format_def_register(&f[i])) {	/* errors are fatal */
-			ast_log(LOG_WARNING, "Failed to register format %s.\n", f[i].name);
+	for (i = 0; f_def[i].desc_size ; i++) {
+		f_def[i].format = ast_format_g726;
+		if (ast_format_def_register(&f_def[i])) {	/* errors are fatal */
+			ast_log(LOG_WARNING, "Failed to register format %s.\n", f_def[i].name);
 			unload_module();
 			return AST_MODULE_LOAD_DECLINE;
 		}

formats/msgsm.h

@@ -1,4 +1,6 @@
 /* Conversion routines derived from code by guido@sienanet.it */
+#ifndef ASTERISK_MSGSM_H
+#define ASTERISK_MSGSM_H
 
 #define GSM_MAGIC 0xD
 
@@ -687,3 +689,5 @@ static inline void conv65( wav_byte * c, gsm_byte * d){
                         writeGSM_33(d+33);
 
 }
+
+#endif /* ASTERISK_MSGSM_H */

funcs/func_curl.c

@@ -121,6 +121,11 @@
 						<para>Include header information in the result
 						(boolean)</para>
 					</enum>
+					<enum name="httpheader">
+						<para>Add HTTP header. Multiple calls add multiple headers.
+						Setting of any header will remove the default
+						"Content-Type application/x-www-form-urlencoded"</para>
+					</enum>
 					<enum name="httptimeout">
 						<para>For HTTP(S) URIs, number of seconds to wait for a
 						server response</para>
@@ -181,7 +186,7 @@
 		</syntax>
 		<description>
 			<para>Options may be set globally or per channel.  Per-channel
-			settings will override global settings.</para>
+			settings will override global settings. Only HTTP headers are added instead of overriding</para>
 		</description>
 		<see-also>
 			<ref type="function">CURL</ref>
@@ -243,6 +248,9 @@ static int parse_curlopt_key(const char *name, CURLoption *key, enum optiontype
 	if (!strcasecmp(name, "header")) {
 		*key = CURLOPT_HEADER;
 		*ot = OT_BOOLEAN;
+	} else if (!strcasecmp(name, "httpheader")) {
+		*key = CURLOPT_HTTPHEADER;
+		*ot = OT_STRING;
 	} else if (!strcasecmp(name, "proxy")) {
 		*key = CURLOPT_PROXY;
 		*ot = OT_STRING;
@@ -412,16 +420,18 @@ yuck:
 		return -1;
 	}
 
-	/* Remove any existing entry */
+	/* Remove any existing entry, only http headers are left */
 	AST_LIST_LOCK(list);
-	AST_LIST_TRAVERSE_SAFE_BEGIN(list, cur, list) {
-		if (cur->key == new->key) {
-			AST_LIST_REMOVE_CURRENT(list);
-			ast_free(cur);
-			break;
+	if (new->key != CURLOPT_HTTPHEADER) {
+		AST_LIST_TRAVERSE_SAFE_BEGIN(list, cur, list) {
+			if (cur->key == new->key) {
+				AST_LIST_REMOVE_CURRENT(list);
+				ast_free(cur);
+				break;
+			}
 		}
+		AST_LIST_TRAVERSE_SAFE_END
 	}
-	AST_LIST_TRAVERSE_SAFE_END
 
 	/* Insert new entry */
 	ast_debug(1, "Inserting entry %p with key %d and value %p\n", new, new->key, new->value);
@@ -639,6 +649,7 @@ static int acf_curl_helper(struct ast_channel *chan, struct curl_args *args)
 	int ret = -1;
 	CURL **curl;
 	struct curl_settings *cur;
+	struct curl_slist *headers = NULL;
 	struct ast_datastore *store = NULL;
 	int hashcompat = 0;
 	AST_LIST_HEAD(global_curl_info, curl_settings) *list = NULL;
@@ -666,6 +677,8 @@ static int acf_curl_helper(struct ast_channel *chan, struct curl_args *args)
 	AST_LIST_TRAVERSE(&global_curl_info, cur, list) {
 		if (cur->key == CURLOPT_SPECIAL_HASHCOMPAT) {
 			hashcompat = (long) cur->value;
+		} else if (cur->key == CURLOPT_HTTPHEADER) {
+			headers = curl_slist_append(headers, (char*) cur->value);
 		} else {
 			curl_easy_setopt(*curl, cur->key, cur->value);
 		}
@@ -682,6 +695,8 @@ static int acf_curl_helper(struct ast_channel *chan, struct curl_args *args)
 			AST_LIST_TRAVERSE(list, cur, list) {
 				if (cur->key == CURLOPT_SPECIAL_HASHCOMPAT) {
 					hashcompat = (long) cur->value;
+				} else if (cur->key == CURLOPT_HTTPHEADER) {
+					headers = curl_slist_append(headers, (char*) cur->value);
 				} else {
 					curl_easy_setopt(*curl, cur->key, cur->value);
 				}
@@ -697,6 +712,10 @@ static int acf_curl_helper(struct ast_channel *chan, struct curl_args *args)
 		curl_easy_setopt(*curl, CURLOPT_POSTFIELDS, args->postdata);
 	}
 
+	if (headers) {
+		curl_easy_setopt(*curl, CURLOPT_HTTPHEADER, headers);
+	}
+
 	/* Temporarily assign a buffer for curl to write errors to. */
 	curl_errbuf[0] = curl_errbuf[CURL_ERROR_SIZE] = '\0';
 	curl_easy_setopt(*curl, CURLOPT_ERRORBUFFER, curl_errbuf);
@@ -714,6 +733,7 @@ static int acf_curl_helper(struct ast_channel *chan, struct curl_args *args)
 	if (store) {
 		AST_LIST_UNLOCK(list);
 	}
+	curl_slist_free_all(headers);
 
 	if (args->postdata) {
 		curl_easy_setopt(*curl, CURLOPT_POST, 0);
@@ -841,6 +861,7 @@ static struct ast_custom_function acf_curlopt = {
 "  ftptext        - For FTP, force a text transfer (boolean)\n"
 "  ftptimeout     - For FTP, the server response timeout\n"
 "  header         - Retrieve header information (boolean)\n"
+"  httpheader     - Add new custom http header (string)\n"
 "  httptimeout    - Number of seconds to wait for HTTP response\n"
 "  maxredirs      - Maximum number of redirects to follow\n"
 "  proxy          - Hostname or IP to use as a proxy\n"

funcs/func_env.c

@@ -709,12 +709,13 @@ static int file_read(struct ast_channel *chan, const char *cmd, char *data, stru
 		ast_debug(3, "offset=%" PRId64 ", length=%" PRId64 ", offset_offset=%" PRId64 ", length_offset=%" PRId64 "\n", offset, length, offset_offset, length_offset);
 		for (i = offset_offset; i < flength; i += sizeof(fbuf)) {
 			char *pos;
-			if (fread(fbuf, 1, sizeof(fbuf), ff) < sizeof(fbuf) && !feof(ff)) {
+			size_t bytes_read;
+			if ((bytes_read = fread(fbuf, 1, sizeof(fbuf), ff)) < sizeof(fbuf) && !feof(ff)) {
 				ast_log(LOG_ERROR, "Short read?!!\n");
 				fclose(ff);
 				return -1;
 			}
-			for (pos = fbuf; pos < fbuf + sizeof(fbuf); pos++) {
+			for (pos = fbuf; pos < fbuf + bytes_read; pos++) {
 				LINE_COUNTER(pos, format, current_length);
 
 				if (current_length == length) {

funcs/func_jitterbuffer.c

@@ -62,8 +62,9 @@
 		</syntax>
 		<description>
 			<para>Jitterbuffers are constructed in two different ways.
-			The first always take three arguments: <replaceable>max_size</replaceable>,
-			<replaceable>resync_threshold</replaceable>, and <replaceable>target_extra</replaceable>.
+			The first always take four arguments: <replaceable>max_size</replaceable>,
+			<replaceable>resync_threshold</replaceable>, <replaceable>target_extra</replaceable>,
+			and <replaceable>sync_video</replaceable>.
 			Alternatively, a single argument of <literal>default</literal> can be provided,
 			which will construct the default jitterbuffer for the given
 			<replaceable>jitterbuffer type</replaceable>.</para>
@@ -76,12 +77,17 @@
 			<para>target_extra: This option only affects the adaptive jitterbuffer. It represents
 			the amount time in milliseconds by which the new jitter buffer will pad its size.
 			Defaults to 40ms.</para>
+			<para>sync_video: This option enables video synchronization with the audio stream. It can be
+			turned on and off. Defaults to off.</para>
 			<example title="Fixed with defaults" language="text">
 			exten => 1,1,Set(JITTERBUFFER(fixed)=default)
 			</example>
 			<example title="Fixed with 200ms max size" language="text">
 			exten => 1,1,Set(JITTERBUFFER(fixed)=200)
 			</example>
+			<example title="Fixed with 200ms max size and video sync support" language="text">
+			exten => 1,1,Set(JITTERBUFFER(fixed)=200,,,yes)
+			</example>
 			<example title="Fixed with 200ms max size, resync threshold 1500" language="text">
 			exten => 1,1,Set(JITTERBUFFER(fixed)=200,1500)
 			</example>
@@ -91,6 +97,9 @@
 			<example title="Adaptive with 200ms max size, 60ms target extra" language="text">
 			exten => 1,1,Set(JITTERBUFFER(adaptive)=200,,60)
 			</example>
+			<example title="Adaptive with 200ms max size and video sync support" language="text">
+			exten => 1,1,Set(JITTERBUFFER(adaptive)=200,,,yes)
+			</example>
 			<example title="Set a fixed jitterbuffer with defaults; then remove it" language="text">
 			exten => 1,1,Set(JITTERBUFFER(fixed)=default)
 			exten => 1,n,Set(JITTERBUFFER(disabled)=)
@@ -133,6 +142,7 @@ static int jb_helper(struct ast_channel *chan, const char *cmd, char *data, cons
 			AST_APP_ARG(max_size);
 			AST_APP_ARG(resync_threshold);
 			AST_APP_ARG(target_extra);
+			AST_APP_ARG(sync_video);
 		);
 
 		AST_STANDARD_APP_ARGS(args, parse);
@@ -151,6 +161,11 @@ static int jb_helper(struct ast_channel *chan, const char *cmd, char *data, cons
 				"jbtargetextra",
 				args.target_extra);
 		}
+		if (!ast_strlen_zero(args.sync_video)) {
+			res |= ast_jb_read_conf(&jb_conf,
+				"jbsyncvideo",
+				args.sync_video);
+		}
 		if (res) {
 			ast_log(LOG_WARNING, "Invalid jitterbuffer parameters %s\n", value);
 		}

funcs/func_talkdetect.c

@@ -203,7 +203,7 @@ static int talk_detect_audiohook_cb(struct ast_audiohook *audiohook, struct ast_
 			int64_t diff_ms = ast_tvdiff_ms(ast_tvnow(), td_params->talking_start);
 			diff_ms -= td_params->dsp_silence_threshold;
 
-			blob = ast_json_pack("{s: i}", "duration", diff_ms);
+			blob = ast_json_pack("{s: I}", "duration", (ast_json_int_t)diff_ms);
 			if (!blob) {
 				return 1;
 			}

include/asterisk/abstract_jb.h

@@ -44,7 +44,8 @@ struct ast_frame;
 enum {
 	AST_JB_ENABLED = (1 << 0),
 	AST_JB_FORCED =  (1 << 1),
-	AST_JB_LOG =     (1 << 2)
+	AST_JB_LOG =     (1 << 2),
+	AST_JB_SYNC_VIDEO =   (1 << 3)
 };
 
 enum ast_jb_type {
@@ -89,6 +90,7 @@ struct ast_jb_conf
 #define AST_JB_CONF_TARGET_EXTRA "targetextra"
 #define AST_JB_CONF_IMPL "impl"
 #define AST_JB_CONF_LOG "log"
+#define AST_JB_CONF_SYNC_VIDEO "syncvideo"
 
 /* Hooks for the abstract jb implementation */
 /*! \brief Create */

include/asterisk/ari.h

@@ -101,7 +101,7 @@ struct ast_ari_response {
 	/*! Corresponding text for the response code */
 	const char *response_text; /* Shouldn't http.c handle this? */
 	/*! Flag to indicate that no further response is needed */
-	int no_response:1;
+	unsigned int no_response:1;
 };
 
 /*!

include/asterisk/astobj2.h

@@ -376,6 +376,8 @@ enum ao2_alloc_opts {
 	 * should never be passed directly to ao2_alloc.
 	 */
 	AO2_ALLOC_OPT_LOCK_OBJ = AO2_ALLOC_OPT_LOCK_MASK,
+	/*! The ao2 object will not record any REF_DEBUG entries */
+	AO2_ALLOC_OPT_NO_REF_DEBUG = (1 << 2),
 };
 
 /*!
@@ -396,8 +398,6 @@ enum ao2_alloc_opts {
  * - the returned pointer cannot be free()'d or realloc()'ed;
  *   rather, we just call ao2_ref(o, -1);
  *
- * \note refdebug logging is skipped if debug_msg is NULL
- *
  * @{
  */
 
@@ -457,7 +457,6 @@ void *__ao2_alloc_with_lockobj(size_t data_size, ao2_destructor_fn destructor_fn
  * can go away is when we release our reference, and it is
  * the last one in existence.
  *
- * \note refdebug logging is skipped if tag is NULL
  * @{
  */
 

include/asterisk/audiohook.h

@@ -64,6 +64,8 @@ enum ast_audiohook_flags {
 	AST_AUDIOHOOK_MUTE_READ     = (1 << 5), /*!< audiohook should be mute frames read */
 	AST_AUDIOHOOK_MUTE_WRITE    = (1 << 6), /*!< audiohook should be mute frames written */
 	AST_AUDIOHOOK_COMPATIBLE    = (1 << 7), /*!< is the audiohook native slin compatible */
+
+	AST_AUDIOHOOK_SUBSTITUTE_SILENCE = (1 << 8), /*!< Substitute silence for missing audio */
 };
 
 enum ast_audiohook_init_flags {

include/asterisk/autoconfig.h.in

@@ -537,6 +537,12 @@
 /* Define if your system has the MYSQLCLIENT libraries. */
 #undef HAVE_MYSQLCLIENT
 
+/* Define to 1 if mysql/mysql.h has bool defined. */
+#undef HAVE_MYSQLCLIENT_BOOL
+
+/* Define to 1 if mysql/mysql.h has my_bool defined. */
+#undef HAVE_MYSQLCLIENT_MY_BOOL
+
 /* Define to 1 if you have the Network Broadcast Sound library. */
 #undef HAVE_NBS
 

include/asterisk/calendar.h

@@ -133,8 +133,8 @@ struct ast_calendar {
 	int timeframe;       /*!< Span (in mins) of calendar data to pull with each request */
 	pthread_t thread;    /*!< The thread that the calendar is loaded/updated in */
 	ast_cond_t unload;
-	int unloading:1;
-	int pending_deletion:1;
+	unsigned int unloading:1;
+	unsigned int pending_deletion:1;
 	struct ao2_container *events;  /*!< The events that are known at this time */
 };
 

include/asterisk/cel.h

@@ -263,6 +263,28 @@ struct ast_event *ast_cel_create_event(struct ast_channel_snapshot *snapshot,
 		enum ast_cel_event_type event_type, const char *userdefevname,
 		struct ast_json *extra, const char *peer_str);
 
+/*!
+ * \brief Allocate and populate a CEL event structure
+ *
+ * \param snapshot An ast_channel_snapshot of the primary channel associated
+ *        with this channel event.
+ * \param event_type The type of call event being reported.
+ * \param event_time The time at which the event occurred.
+ * \param userdefevname Custom name for the call event. (optional)
+ * \param extra An event-specific opaque JSON blob to be rendered and placed
+ *        in the "CEL_EXTRA" information element of the call event. (optional)
+ * \param peer_str A list of comma-separated peer channel names. (optional)
+ *
+ * \since 13.29.0
+ * \since 16.6.0
+ *
+ * \retval The created ast_event structure
+ * \retval NULL on failure
+ */
+struct ast_event *ast_cel_create_event_with_time(struct ast_channel_snapshot *snapshot,
+		enum ast_cel_event_type event_type, const struct timeval *event_time,
+		const char *userdefevname, struct ast_json *extra, const char *peer_str);
+
 /*!
  * \brief CEL backend callback
  */

include/asterisk/channel_internal.h

@@ -18,6 +18,9 @@
  * \brief Internal channel functions for channel.c to use
  */
 
+#ifndef ASTERISK_CHANNEL_INTERNAL_H
+#define ASTERISK_CHANNEL_INTERNAL_H
+
 #define ast_channel_internal_alloc(destructor, assignedid, requestor) __ast_channel_internal_alloc(destructor, assignedid, requestor, __FILE__, __LINE__, __PRETTY_FUNCTION__)
 struct ast_channel *__ast_channel_internal_alloc(void (*destructor)(void *obj), const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *file, int line, const char *function);
 void ast_channel_internal_finalize(struct ast_channel *chan);
@@ -33,3 +36,5 @@ void ast_channel_internal_set_stream_topology_change_source(
 	struct ast_channel *chan, void *change_source);
 void ast_channel_internal_swap_stream_topology(struct ast_channel *chan1,
 	struct ast_channel *chan2);
+
+#endif /* ASTERISK_CHANNEL_INTERNAL_H */

include/asterisk/config.h

@@ -948,6 +948,24 @@ struct ast_variable *ast_variable_list_append_hint(struct ast_variable **head, s
 	struct ast_variable *new_var);
 #define ast_variable_list_append(head, new_var) ast_variable_list_append_hint(head, NULL, new_var)
 
+/*!
+ * \brief Replace a variable in the given list with a new value
+ * \since 13.30.0
+ *
+ * \param head A pointer to an ast_variable * of the existing variable list head. May NOT be NULL
+ * but the content may be to initialize a new list.  If so, upon return, this parameter will be updated
+ * with a pointer to the new list head.
+ * \param replacement The variable that replaces another variable in the list with the
+ * same name.
+ *
+ * \retval 0 if a variable was replaced in the list
+ * \retval -1 if no replacement occured
+ *
+ * \note The variable name comparison is performed case-sensitively
+ * \note If a variable is replaced, its memory is freed.
+ */
+int ast_variable_list_replace(struct ast_variable **head, struct ast_variable *replacement);
+
 /*!
  * \brief Update variable value within a config
  *