Update CHANGES and UPGRADE.txt for 19.5.0

diff --git a/CHANGES b/CHANGES
index a0b4eccdc37a0986c8776c3c086687d57660e0dd..cf43ed4799797fc94700bf67a9343797a411762f 100644 (file)
--- a/CHANGES
+++ b/CHANGES
@@ -12,6 +12,107 @@
 ===
 ==============================================================================
 
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 19.4.0 to Asterisk 19.5.0 ------------
+------------------------------------------------------------------------------
+
+app_confbridge
+------------------
+ * Added the hear_own_join_sound option to the confbridge user profile to
+   control who hears the sound_join audio file. When set to 'yes' the user
+   entering the conference and the participants already in the conference
+   will hear the sound_join audio file. When set to 'no' the user entering
+   the conference will not hear the sound_join audio file, but the
+   participants already in the conference will hear the sound_join audio file.
+
+ * Adds the CONFBRIDGE_CHANNELS function which can
+   be used to retrieve a list of channels in a ConfBridge,
+   optionally filtered by a particular category. This
+   list can then be used with functions like SHIFT, POP,
+   UNSHIFT, etc.
+
+app_queue
+------------------
+ * The m option now allows an override music on hold
+   class to be specified for the Queue application
+   within the dialplan.
+
+app_voicemail
+------------------
+ * The r option has been added, which prevents deletion
+   of messages from VoiceMailMain, which can be
+   useful for shared mailboxes.
+
+ari
+------------------
+ * Expose channel driver's unique id (which is the Call-ID for SIP/PJSIP)
+   to ARI channel resources as 'protocol_id'.
+
+   ASTERISK-30027
+
+chan_dahdi
+------------------
+ * Previously, cadences were appended on dahdi restart,
+   rather than reloaded. This prevented cadences from
+   being updated and maxed out the available cadences
+   if reloaded multiple times. This behavior is fixed
+   so that reloading cadences is idempotent and cadences
+   can actually be reloaded.
+
+chan_pjsip
+------------------
+ * added global config option "allow_sending_180_after_183"
+
+   Allow Asterisk to send 180 Ringing to an endpoint
+   after 183 Session Progress has been send.
+   If disabled Asterisk will instead send only a
+   183 Session Progress to the endpoint.
+
+ * Hook flash events can now be sent on a PJSIP channel
+   if requested to do so.
+
+chan_sip
+------------------
+ * Session timers get removed on UPDATE
+   Fix if Asterisk receives a SIP REFER with Session-Timers UAC
+   that Asterisk maintains Session-Timers when sending UPDATE request
+
+cli
+------------------
+ * A new CLI command 'dialplan eval function' has been
+   added which allows users to test the behavior of
+   dialplan function calls directly from the CLI.
+
+func_db
+------------------
+ * The function DB_KEYCOUNT has been added, which
+   returns the cardinality of the keys at a specified
+   prefix in AstDB, i.e. the number of keys at a
+   given prefix.
+
+func_evalexten
+------------------
+ * This adds the EVAL_EXTEN function which may be
+   used to evaluate data at dialplan extensions.
+
+res_agi
+------------------
+ * Agi command 'exec' can now be enabled\r
+   to evaluate dialplan functions and variables\r
+   by setting the variable AGIEXECFULL to yes.
+
+res_parking
+------------------
+ * An m option to Park and ParkAndAnnounce now allows
+   specifying a music on hold class override.
+
+stasis_channels
+------------------
+ * Expose channel driver's unique id (which is the Call-ID for SIP/PJSIP)
+   to ARI channel resources as 'protocol_id'.
+
+   ASTERISK-30027
+
 ------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 19.3.1 to Asterisk 19.3.2 ------------
 ------------------------------------------------------------------------------

diff --git a/UPGRADE.txt b/UPGRADE.txt
index fedcff9cf50afeddcc4eed4dd7b78e7446f7d7b5..bfa9f9eb684f8ce9c82d4ee4faa7fa17e1e0f7f0 100644 (file)
--- a/UPGRADE.txt
+++ b/UPGRADE.txt
@@ -18,6 +18,18 @@
 ===
 ===========================================================
 
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 19.4.0 to Asterisk 19.5.0 ------------
+------------------------------------------------------------------------------
+
+res_pjsip
+------------------
+ * The 'async_operations' setting on transports is no longer
+   obeyed and instead is always set to 1. This is due to the
+   functionality not being applicable to Asterisk and causing
+   excess unnecessary memory usage. This setting will now be
+   ignored but can also be removed from the configuration file.
+
 ------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 19.2.0 to Asterisk 19.3.0 ------------
 ------------------------------------------------------------------------------

diff --git a/doc/CHANGES-staging/app_confbridge_channels.txt b/doc/CHANGES-staging/app_confbridge_channels.txt
deleted file mode 100644 (file)
index 485f664..0000000
--- a/doc/CHANGES-staging/app_confbridge_channels.txt
+++ /dev/null
@@ -1,7 +0,0 @@
-Subject: app_confbridge
-
-Adds the CONFBRIDGE_CHANNELS function which can
-be used to retrieve a list of channels in a ConfBridge,
-optionally filtered by a particular category. This
-list can then be used with functions like SHIFT, POP,
-UNSHIFT, etc.

diff --git a/doc/CHANGES-staging/app_confbridge_hear_join.txt b/doc/CHANGES-staging/app_confbridge_hear_join.txt
deleted file mode 100644 (file)
index 40f2383..0000000
--- a/doc/CHANGES-staging/app_confbridge_hear_join.txt
+++ /dev/null
@@ -1,8 +0,0 @@
-Subject: app_confbridge
-
-Added the hear_own_join_sound option to the confbridge user profile to
-control who hears the sound_join audio file. When set to 'yes' the user
-entering the conference and the participants already in the conference
-will hear the sound_join audio file. When set to 'no' the user entering
-the conference will not hear the sound_join audio file, but the
-participants already in the conference will hear the sound_join audio file.

diff --git a/doc/CHANGES-staging/app_queue_music.txt b/doc/CHANGES-staging/app_queue_music.txt
deleted file mode 100644 (file)
index 254a45d..0000000
--- a/doc/CHANGES-staging/app_queue_music.txt
+++ /dev/null
@@ -1,5 +0,0 @@
-Subject: app_queue
-
-The m option now allows an override music on hold
-class to be specified for the Queue application
-within the dialplan.

diff --git a/doc/CHANGES-staging/app_voicemail_nodelete.txt b/doc/CHANGES-staging/app_voicemail_nodelete.txt
deleted file mode 100644 (file)
index ef95896..0000000
--- a/doc/CHANGES-staging/app_voicemail_nodelete.txt
+++ /dev/null
@@ -1,5 +0,0 @@
-Subject: app_voicemail
-
-The r option has been added, which prevents deletion
-of messages from VoiceMailMain, which can be
-useful for shared mailboxes.

diff --git a/doc/CHANGES-staging/ari_add_pvt_id_to_channel_resource.txt b/doc/CHANGES-staging/ari_add_pvt_id_to_channel_resource.txt
deleted file mode 100644 (file)
index a4f008f..0000000
--- a/doc/CHANGES-staging/ari_add_pvt_id_to_channel_resource.txt
+++ /dev/null
@@ -1,7 +0,0 @@
-Subject: ari
-Subject: stasis_channels
-
-Expose channel driver's unique id (which is the Call-ID for SIP/PJSIP)
-to ARI channel resources as 'protocol_id'.
-
-ASTERISK-30027

diff --git a/doc/CHANGES-staging/chan_dahdi_cadences.txt b/doc/CHANGES-staging/chan_dahdi_cadences.txt
deleted file mode 100644 (file)
index b888926..0000000
--- a/doc/CHANGES-staging/chan_dahdi_cadences.txt
+++ /dev/null
@@ -1,8 +0,0 @@
-Subject: chan_dahdi
-
-Previously, cadences were appended on dahdi restart,
-rather than reloaded. This prevented cadences from
-being updated and maxed out the available cadences
-if reloaded multiple times. This behavior is fixed
-so that reloading cadences is idempotent and cadences
-can actually be reloaded.

diff --git a/doc/CHANGES-staging/chan_pjsip_180_sdp.txt b/doc/CHANGES-staging/chan_pjsip_180_sdp.txt
deleted file mode 100644 (file)
index ffd14af..0000000
--- a/doc/CHANGES-staging/chan_pjsip_180_sdp.txt
+++ /dev/null
@@ -1,8 +0,0 @@
-Subject: chan_pjsip
-
-added global config option "allow_sending_180_after_183"
-
-Allow Asterisk to send 180 Ringing to an endpoint
-after 183 Session Progress has been send.
-If disabled Asterisk will instead send only a
-183 Session Progress to the endpoint.

diff --git a/doc/CHANGES-staging/chan_pjsip_flash.txt b/doc/CHANGES-staging/chan_pjsip_flash.txt
deleted file mode 100644 (file)
index 34da796..0000000
--- a/doc/CHANGES-staging/chan_pjsip_flash.txt
+++ /dev/null
@@ -1,4 +0,0 @@
-Subject: chan_pjsip
-
-Hook flash events can now be sent on a PJSIP channel
-if requested to do so.

diff --git a/doc/CHANGES-staging/chan_sip_session-timer_on_update.txt b/doc/CHANGES-staging/chan_sip_session-timer_on_update.txt
deleted file mode 100644 (file)
index 259782f..0000000
--- a/doc/CHANGES-staging/chan_sip_session-timer_on_update.txt
+++ /dev/null
@@ -1,6 +0,0 @@
-Subject: chan_sip
-
-Session timers get removed on UPDATE
-Fix if Asterisk receives a SIP REFER with Session-Timers UAC
-that Asterisk maintains Session-Timers when sending UPDATE request
-

diff --git a/doc/CHANGES-staging/cli_eval_function.txt b/doc/CHANGES-staging/cli_eval_function.txt
deleted file mode 100644 (file)
index 9f7873c..0000000
--- a/doc/CHANGES-staging/cli_eval_function.txt
+++ /dev/null
@@ -1,5 +0,0 @@
-Subject: cli
-
-A new CLI command 'dialplan eval function' has been
-added which allows users to test the behavior of
-dialplan function calls directly from the CLI.

diff --git a/doc/CHANGES-staging/func_db.txt b/doc/CHANGES-staging/func_db.txt
deleted file mode 100644 (file)
index 72e333a..0000000
--- a/doc/CHANGES-staging/func_db.txt
+++ /dev/null
@@ -1,6 +0,0 @@
-Subject: func_db
-
-The function DB_KEYCOUNT has been added, which
-returns the cardinality of the keys at a specified
-prefix in AstDB, i.e. the number of keys at a
-given prefix.

diff --git a/doc/CHANGES-staging/func_evalexten.txt b/doc/CHANGES-staging/func_evalexten.txt
deleted file mode 100644 (file)
index f912bbe..0000000
--- a/doc/CHANGES-staging/func_evalexten.txt
+++ /dev/null
@@ -1,4 +0,0 @@
-Subject: func_evalexten
-
-This adds the EVAL_EXTEN function which may be
-used to evaluate data at dialplan extensions.

diff --git a/doc/CHANGES-staging/res_agi.txt b/doc/CHANGES-staging/res_agi.txt
deleted file mode 100644 (file)
index eb6132d..0000000
--- a/doc/CHANGES-staging/res_agi.txt
+++ /dev/null
@@ -1,5 +0,0 @@
-Subject: res_agi\r
-\r
-Agi command 'exec' can now be enabled\r
-to evaluate dialplan functions and variables\r
-by setting the variable AGIEXECFULL to yes.
\ No newline at end of file

diff --git a/doc/CHANGES-staging/res_parking_moh.txt b/doc/CHANGES-staging/res_parking_moh.txt
deleted file mode 100644 (file)
index 50f589c..0000000
--- a/doc/CHANGES-staging/res_parking_moh.txt
+++ /dev/null
@@ -1,4 +0,0 @@
-Subject: res_parking
-
-An m option to Park and ParkAndAnnounce now allows
-specifying a music on hold class override.

diff --git a/doc/UPGRADE-staging/res_pjsip_async_operations.txt b/doc/UPGRADE-staging/res_pjsip_async_operations.txt
deleted file mode 100644 (file)
index cf9f942..0000000
--- a/doc/UPGRADE-staging/res_pjsip_async_operations.txt
+++ /dev/null
@@ -1,7 +0,0 @@
-Subject: res_pjsip
-
-The 'async_operations' setting on transports is no longer
-obeyed and instead is always set to 1. This is due to the
-functionality not being applicable to Asterisk and causing
-excess unnecessary memory usage. This setting will now be
-ignored but can also be removed from the configuration file.