tcptls: Change error message to debug.

[asterisk/asterisk.git] / CHANGES

diff --git a/CHANGES b/CHANGES
index 68e4599..67454bc 100644 (file)
--- a/CHANGES
+++ b/CHANGES
@@ -9,6 +9,18 @@
 ==============================================================================
 
 ------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 15 to Asterisk 16 --------------------
+------------------------------------------------------------------------------
+
+chan_sip
+------------------
+ * New function SIP_HEADERS() enumerates all headers in the incoming INVITE.
+
+ * The variable GET_TRANSFERRER_DATA set in the peer channel causes matching
+   headers be retrieved from the REFER message and made accessible to the
+   dialplan in the hash TRANSFER_DATA.
+
+------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 14 to Asterisk 15 --------------------
 ------------------------------------------------------------------------------
 
@@ -17,10 +29,66 @@ app_queue
  * PAUSEALL/UNPAUSEALL now sets the pause reason in the queue_log if it has
    been defined.
 
+ * A new option, "announce-position-only-up," has been added that, when set to
+   yes, causes position announcements to only be played when the caller's
+   queue position has improved since the last time that we annouced their
+   position. This default is no.
+
+Build System
+------------------
+ * '--with-pjproject-bundled' is now the default when running ./configure
+   It can be disabled with '--without-pjproject-bundled'.
+
+ * A '--with-download-cache' option is now available which is equivalent to
+   setting '--with-sounds-cache' and '--with-externals-cache' to the same
+   value.  The download cache can also be set via the AST_DOWNLOAD_CACHE
+   environment variable.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 14.6.0 to Asterisk 14.7.0 ------------
+------------------------------------------------------------------------------
+
+res_pjsip
+------------------
+ * The "external_media_address" on transports is now resolved using dnsmgr and
+   when dnsmgr refreshes are enabled will be automatically updated with the new
+   IP address of a given hostname.
+
+res_musiconhold
+------------------
+ * By default, when res_musiconhold reloads or unloads, it sends a HUP signal
+   to custom applications (and all descendants), waits 100ms, then sends a
+   TERM signal, waits 100ms, then finally sends a KILL signal.  An application
+   which is interacting with an external device and/or spawns children of its
+   own may not be able to exit cleanly in the default times, expecially if sent
+   a KILL signal, or if it's children are getting signals directly from
+   res_musiconhoild.  To allow extra time, the 'kill_escalation_delay'
+   class option can be used to set the number of milliseconds res_musiconhold
+   waits before escalating kill signals, with the default being the current
+   100ms.  To control to whom the signals are sent, the "kill_method"
+   class option can be set to "process_group" (the default, existing behavior),
+   which sends signals to the application and its descendants directly, or
+   "process" which sends signals only to the application itself.
+
+ * New dialplan function PJSIP_DTMF_MODE added to get or change the DTMF mode
+   of a channel on a per-call basis.
+
+res_xmpp
+-----------------
+ * OAuth 2.0 authentication is now supported when contacting Google. Follow the
+   instructions in xmpp.conf.sample to retrieve and configure the necessary
+   tokens.
+
 ------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 14.5.0 to Asterisk 14.6.0 ------------
 ------------------------------------------------------------------------------
 
+app_voicemail
+------------------
+ * A new global option "imap_poll_logout" was added to specify whether need to
+   disconnect from the IMAP server after polling of mailboxes.
+   Default: no
+
 res_pjsip
 ------------------
  * A new endpoint option "refer_blind_progress" was added to turn off notifying
@@ -57,6 +125,14 @@ chan_pjsip
    from the SDP, unless the remote side sends a different codec and we will
    switch to match.
 
+Build System
+------------------
+ * Added a new PJPROJECT_CONFIGURE_OPTS environment variable which can be used
+   to pass arbitrary options to the bundled pjproject configure.
+
+ * Automatically set the bundled pjproject configure --host and --build
+   options to match those supplied for the asterisk configure.
+
 ------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 14.4.0 to Asterisk 14.5.0 ------------
 ------------------------------------------------------------------------------
@@ -1491,7 +1567,7 @@ res_pjsip_endpoint_identifer_ip
 ------------------
  * New CLI commands have been added: "pjsip show identif(y|ies)", which lists
    all configured PJSIP identify objects
-   
+
 ------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 12 to Asterisk 13 --------------------
 ------------------------------------------------------------------------------