Update CHANGES and UPGRADE.txt for 18.7.0

CHANGES

@@ -12,6 +12,124 @@
 ===
 ==============================================================================
 
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 18.6.0 to Asterisk 18.7.0 ------------
+------------------------------------------------------------------------------
+
+Channel-agnostic MF support
+------------------
+ * A SendMF application and PlayMF manager
+   application are now included to send
+   arbitrary standard R1 MF tones on the
+   current channel or another specified channel.
+
+app_milliwatt
+------------------
+ * The Milliwatt application's existing behavior is
+   incorrect in that it plays a constant tone, which
+   is not how digital milliwatt test lines actually
+   work.
+
+   An option is added so that a proper milliwatt test
+   tone can be provided, including a 1 second silent
+   interval every 10 seconds. However, for compatability
+   reasons, the default behavior remains unchanged.
+
+app_morsecode
+------------------
+ * Extends the Morsecode application by adding support for
+   American Morse code and adds a configurable option
+   for the frequency used in off intervals.
+
+app_originate
+------------------
+ * Codecs can now be specified for dialplan-originated
+   calls, as with call files and the manager action.
+   By default, only the slin codec is now used, instead
+   of all the slin* codecs.
+
+app_queue
+------------------
+ * Reload behavior in app_queue has been changed so
+   queue and agent stats are not reset during full
+   app_queue module reloads. The queue reset stats
+   CLI command may still be used to reset stats while
+   Asterisk is running.
+
+app_read
+------------------
+ * A new option allows the digit '#' to be read literally,
+   rather than used exclusively as the input terminator
+   character.
+
+app_voicemail
+------------------
+ * Add a new 'S' option to VoiceMail which prevents the instructions
+   (vm-intro) from being played if a busy/unavailable/temporary greeting
+   from the voicemail user is played. This is similar to the existing 's'
+   option except that instructions will still be played if no user
+   greeting is available.
+
+chan_iax2
+------------------
+ * ANI2 (OLI) is now transmitted over IAX2 calls
+   as an information element.
+
+func_env.c
+------------------
+ * Two new functions, DIRNAME and BASENAME, are now
+   included which allow users to obtain the directory
+   or the base filename of any file.
+
+func_framedrop
+------------------
+ * New function to selectively drop specified frames
+   in either direction on a channel.
+
+func_scramble
+------------------
+ * Adds an audio scrambler function that may be used to
+   distort voice audio on a channel as a privacy
+   enhancement.
+
+func_strings
+------------------
+ * A new STRBETWEEN function is now included which
+   allows a substring to be inserted between characters
+   in a string. This is particularly useful for transforming
+   dial strings, such as adding pauses between digits
+   for a string of digits that are sent to another channel.
+
+res_pjproject
+------------------
+ * In pjproject.conf you can now map pjproject log levels
+   to the Asterisk TRACE log level.  The default mappings
+   have therefore changed so that only pjproject levels
+   3 and 4 are mapped to DEBUG and 5 and 6 are now mapped
+   to TRACE.  Previously 3, 4, 5, and 6 were all mapped to
+   DEBUG.
+
+res_rtp_asterisk
+------------------
+ * When the address of the STUN server (stunaddr) is a name resolved via DNS, the
+   stunaddr will be recurringly resolved when the DNS answer Time-To-Live (TTL)
+   expires. This allows the STUN server to change its IP address without having to
+   reload the res_rtp_asterisk module.
+
+res_tonedetect
+------------------
+ * Arbitrary tone detection is now available through a
+   WaitForTone application (blocking) and a TONE_DETECT
+   function (non-blocking).
+
+say.c
+------------------
+ * Adds SAYFILES function to retrieve the file names that would
+   be played by corresponding Say applications, such as
+   SayDigits, SayAlpha, etc.
+
+   Additionally adds SayMoney and SayOrdinal applications.
+
 ------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 18.5.0 to Asterisk 18.6.0 ------------
 ------------------------------------------------------------------------------

doc/CHANGES-staging/app_milliwatt.txt

@@ -1,11 +0,0 @@
-Subject: app_milliwatt
-
-The Milliwatt application's existing behavior is
-incorrect in that it plays a constant tone, which
-is not how digital milliwatt test lines actually
-work.
-
-An option is added so that a proper milliwatt test
-tone can be provided, including a 1 second silent
-interval every 10 seconds. However, for compatability
-reasons, the default behavior remains unchanged.

doc/CHANGES-staging/app_morsecode.txt

@@ -1,6 +0,0 @@
-Subject: app_morsecode
-
-Extends the Morsecode application by adding support for
-American Morse code and adds a configurable option
-for the frequency used in off intervals.
-

doc/CHANGES-staging/app_originate_codecs.txt

@@ -1,6 +0,0 @@
-Subject: app_originate
-
-Codecs can now be specified for dialplan-originated
-calls, as with call files and the manager action.
-By default, only the slin codec is now used, instead
-of all the slin* codecs.

doc/CHANGES-staging/app_queue_stats.txt

@@ -1,7 +0,0 @@
-Subject: app_queue
-
-Reload behavior in app_queue has been changed so
-queue and agent stats are not reset during full
-app_queue module reloads. The queue reset stats
-CLI command may still be used to reset stats while
-Asterisk is running.

doc/CHANGES-staging/app_read.txt

@@ -1,5 +0,0 @@
-Subject: app_read
-
-A new option allows the digit '#' to be read literally,
-rather than used exclusively as the input terminator
-character.

doc/CHANGES-staging/app_voicemail.txt

@@ -1,7 +0,0 @@
-Subject: app_voicemail
-
-Add a new 'S' option to VoiceMail which prevents the instructions
-(vm-intro) from being played if a busy/unavailable/temporary greeting
-from the voicemail user is played. This is similar to the existing 's'
-option except that instructions will still be played if no user
-greeting is available.

doc/CHANGES-staging/chan_iax2_ani2.txt

@@ -1,4 +0,0 @@
-Subject: chan_iax2
-
-ANI2 (OLI) is now transmitted over IAX2 calls
-as an information element.

doc/CHANGES-staging/func_env.txt

@@ -1,5 +0,0 @@
-Subject: func_env.c
-
-Two new functions, DIRNAME and BASENAME, are now
-included which allow users to obtain the directory
-or the base filename of any file.

doc/CHANGES-staging/func_framedrop.txt

@@ -1,5 +0,0 @@
-Subject: func_framedrop
-
-New function to selectively drop specified frames
-in either direction on a channel.
-

doc/CHANGES-staging/func_scramble.txt

@@ -1,5 +0,0 @@
-Subject: func_scramble
-
-Adds an audio scrambler function that may be used to
-distort voice audio on a channel as a privacy
-enhancement.

doc/CHANGES-staging/func_strings.txt

@@ -1,7 +0,0 @@
-Subject: func_strings
-
-A new STRBETWEEN function is now included which
-allows a substring to be inserted between characters
-in a string. This is particularly useful for transforming
-dial strings, such as adding pauses between digits
-for a string of digits that are sent to another channel.

doc/CHANGES-staging/mf.txt

@@ -1,6 +0,0 @@
-Subject: Channel-agnostic MF support
-
-A SendMF application and PlayMF manager
-application are now included to send
-arbitrary standard R1 MF tones on the
-current channel or another specified channel.

doc/CHANGES-staging/res_pjproject.txt

@@ -1,8 +0,0 @@
-Subject: res_pjproject
-
-In pjproject.conf you can now map pjproject log levels
-to the Asterisk TRACE log level.  The default mappings
-have therefore changed so that only pjproject levels
-3 and 4 are mapped to DEBUG and 5 and 6 are now mapped
-to TRACE.  Previously 3, 4, 5, and 6 were all mapped to
-DEBUG.

doc/CHANGES-staging/res_rtp_asterisk_stunaddr_recurring_resolution.txt

@@ -1,6 +0,0 @@
-Subject: res_rtp_asterisk
-
-When the address of the STUN server (stunaddr) is a name resolved via DNS, the
-stunaddr will be recurringly resolved when the DNS answer Time-To-Live (TTL)
-expires. This allows the STUN server to change its IP address without having to
-reload the res_rtp_asterisk module.

doc/CHANGES-staging/res_tonedetect.txt

@@ -1,5 +0,0 @@
-Subject: res_tonedetect
-
-Arbitrary tone detection is now available through a
-WaitForTone application (blocking) and a TONE_DETECT
-function (non-blocking).

doc/CHANGES-staging/say.txt

@@ -1,7 +0,0 @@
-Subject: say.c
-
-Adds SAYFILES function to retrieve the file names that would
-be played by corresponding Say applications, such as
-SayDigits, SayAlpha, etc.
-
-Additionally adds SayMoney and SayOrdinal applications.