[Asterisk-code-review] build: Revise CHANGES and UPGRADE.txt handling. (asterisk[master])
Benjamin Keith Ford
asteriskteam at digium.com
Fri Feb 1 12:50:09 CST 2019
Benjamin Keith Ford has uploaded this change for review. ( https://gerrit.asterisk.org/10945
Change subject: build: Revise CHANGES and UPGRADE.txt handling.
......................................................................
build: Revise CHANGES and UPGRADE.txt handling.
This changes the way that we handle adding changes to CHANGES and
UPGRADE.txt. The reason for this is because whenever someone needed to
make a changes to one of these files and someone else had already done
so, you would run into merge conflicts. With this new setup, there will
never be merge conflicts since all changes will be documented in the
doc/<file>-staging directory. The release script is now responsible for
merging all of these changes into the appropriate files.
There is a special format that these files have to follow in order to be
parsed. The files do not need to have a meaningful name, but it is
strongly recommended. For example, if you made a change to pjsip, you
may have something like this "res_pjsip_relative_title", where
"relative_title" is something more descriptive than that. Inside each
file, you will need a subject line for your change, followed by a
description. There can be multiple subject lines. The file may look
something like this:
Subject: res_pjsip
A description that explains the changes made and why. The release
script will handle the bulleting and section separators!
You can still separate with new lines within your
description.
Subject: res_pjsip
You can have more than one subject!
Subject: Core
The subjects don't have to be the same.
The headers (Subject:) are case sensative.
For more information, check out the wiki page:
https://wiki.asterisk.org/wiki/display/AST/CHANGES+and+UPGRADE.txt
Change-Id: I19cf4b569321c88155a65e9b0b80f6d58075dd47
---
M CHANGES
D UPGRADE.txt
A doc/CHANGES-staging/README
A doc/CHANGES-staging/app_voicemail_aliasescontext
A doc/CHANGES-staging/bridge_snapshots_and_cache_changes
A doc/CHANGES-staging/chan_pjsip_PJSIP_PARSE_URI
A doc/CHANGES-staging/chan_sip_deprecated
A doc/CHANGES-staging/channels_remove_stasis_cache_for_channel_snapshots
A doc/CHANGES-staging/core_ast_bt_get_symbols_return
A doc/CHANGES-staging/features_automon_automixmon
A doc/CHANGES-staging/pbx_config_multiple_global_sections
A doc/CHANGES-staging/res_pjsip_send_contact_status_on_update_registration
A doc/CHANGES-staging/res_pjsip_trust_and_send_connected_line_options
A doc/CHANGES-staging/res_rtp_asterisk_strictrtp_seqno_option
A doc/UPGRADE-staging/README
A doc/UPGRADE-staging/bridge_snapshots_and_cache_changes
A doc/UPGRADE-staging/chan_sip_deprecated
A doc/UPGRADE-staging/channels_remove_stasis_cache_for_channel_snapshots
A doc/UPGRADE-staging/func_callerid_remove_CALLERPRES
A doc/UPGRADE-staging/res_parking_PARKINGSLOT_no_longer_set
A doc/UPGRADE-staging/res_xmpp_remove_JabberStatus_application
21 files changed, 251 insertions(+), 218 deletions(-)
git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/45/10945/1
diff --git a/CHANGES b/CHANGES
index 23612fb..62698a9 100644
--- a/CHANGES
+++ b/CHANGES
@@ -9,133 +9,6 @@
==============================================================================
------------------------------------------------------------------------------
---- Functionality changes from Asterisk 16 to Asterisk 17 --------------------
-------------------------------------------------------------------------------
-
-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
-
-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.
-
-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.
-
-------------------------------------------------------------------------------
---- Functionality changes from Asterisk 16.1.0 to Asterisk 16.2.0 ------------
-------------------------------------------------------------------------------
-
-res_pjsip
-------------------
- * Added "send_contact_status_on_update_registration" global configuration option
- to enable sending AMI ContactStatus event when a device refreshes its registration.
-
-Features
-------------------
- * Before Asterisk 12, when using the automon or automixmon features defined
- in features.conf, a channel variable (TOUCH_MIXMONITOR_OUTPUT) was set on
- both channels, indicating the filename of the recording.
-
- When bridging was overhauled in Asterisk 12, the behavior was changed such
- that the variable was only set on the peer channel and not on the channel
- that initiated the automon or automixmon.
-
- The previous behavior has been restored so both channels receive the
- channel variable when one of these features is invoked.
-
-app_voicemail
-------------------
- * You can now specify a special context with the "aliasescontext" parameter
- in voicemail.conf which will allow you to create aliases for physical
- mailboxes.
-
-------------------------------------------------------------------------------
---- Functionality changes from Asterisk 16.0.0 to Asterisk 16.1.0 ------------
-------------------------------------------------------------------------------
-
-pbx_config
-------------------
- * pbx_config will now find and process multiple 'globals' sections from
- extensions.conf. Variables are processed in the order they are found
- and duplicate variables overwrite the previous value.
-
-chan_pjsip
-------------------
- * New dialplan function PJSIP_PARSE_URI added to parse an URI and return
- a specified part of the URI.
-
-Core
-------------------
- * ast_bt_get_symbols() now returns a vector of strings instead of an
- array of strings. This must be freed with ast_bt_free_symbols.
-
-res_pjsip
-------------------
- * New options 'trust_connected_line' and 'send_connected_line' have been
- added to the endpoint. The option 'trust_connected_line' is to control
- if connected line updates are accepted from this endpoint.
- The option 'send_connected_line' is to control if connected line updates
- can be sent to this endpoint.
- The default value is 'yes' for both options.
-
-res_rtp_asterisk
-------------------
- * The existing strictrtp option in rtp.conf has a new choice availabe, called
- 'seqno', which behaves the same way as setting strictrtp to 'yes', but will
- ignore the time interval during learning so that bursts of packets can still
- trigger learning our source.
-
-------------------------------------------------------------------------------
--- Functionality changes from Asterisk 15 to Asterisk 16 --------------------
------------------------------------------------------------------------------
diff --git a/UPGRADE.txt b/UPGRADE.txt
deleted file mode 100644
index 53e9c4a..0000000
--- a/UPGRADE.txt
+++ /dev/null
@@ -1,91 +0,0 @@
-===========================================================
-===
-=== Information for upgrading between Asterisk versions
-===
-=== These files document all the changes that MUST be taken
-=== into account when upgrading between the Asterisk
-=== versions listed below. These changes may require that
-=== you modify your configuration files, dialplan or (in
-=== some cases) source code if you have your own Asterisk
-=== modules or patches. These files also include advance
-=== notice of any functionality that has been marked as
-=== 'deprecated' and may be removed in a future release,
-=== along with the suggested replacement functionality.
-===
-=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
-=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
-=== UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6
-=== UPGRADE-1.8.txt -- Upgrade info for 1.6 to 1.8
-=== UPGRADE-10.txt -- Upgrade info for 1.8 to 10
-=== UPGRADE-11.txt -- Upgrade info for 10 to 11
-=== UPGRADE-12.txt -- Upgrade info for 11 to 12
-=== UPGRADE-13.txt -- Upgrade info for 12 to 13
-=== UPGRADE-14.txt -- Upgrade info for 13 to 14
-=== UPGRADE-15.txt -- Upgrade info for 14 to 15
-=== UPGRADE-16.txt -- Upgrade info for 15 to 16
-===========================================================
-
-New in 17.0.0:
-
-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.
-
-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.
-
-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.
-
diff --git a/doc/CHANGES-staging/README b/doc/CHANGES-staging/README
new file mode 100644
index 0000000..9c3c79f
--- /dev/null
+++ b/doc/CHANGES-staging/README
@@ -0,0 +1,22 @@
+DO NOT REMOVE THIS FILE!
+
+The only files that should be added to this directory are ones that will be
+used by the release script to update the CHANGES file automatically. The only
+time that it is necessary to add something to the CHANGES-staging directory is
+if you are either adding a new feature to Asterisk or adding new functionality
+to an existing feature. The file does not need to have a meaningful name, but
+it probably should. If there are multiple items that need documenting, each can
+be separated with a subject line, which should always start with "SUBJECT:",
+followed by the subject of the change. For example, if you are making a change
+to PJSIP, then you might add the file "res_pjsip_my_cool_feature" to this
+directory, with a short description of what it does. Here's an example:
+
+SUBJECT: res_pjsip
+
+Here's a pretty good description of my new feature that explains exactly what
+it does and how to use it.
+
+SUBJECT: core
+
+Here's another description of something else I added that is a big enough
+change to warrant another entry in the CHANGES file.
diff --git a/doc/CHANGES-staging/app_voicemail_aliasescontext b/doc/CHANGES-staging/app_voicemail_aliasescontext
new file mode 100644
index 0000000..e56dabd
--- /dev/null
+++ b/doc/CHANGES-staging/app_voicemail_aliasescontext
@@ -0,0 +1,5 @@
+Subject: app_voicemail
+
+You can now specify a special context with the "aliasescontext" parameter
+in voicemail.conf which will allow you to create aliases for physical
+mailboxes.
diff --git a/doc/CHANGES-staging/bridge_snapshots_and_cache_changes b/doc/CHANGES-staging/bridge_snapshots_and_cache_changes
new file mode 100644
index 0000000..ebcca2e
--- /dev/null
+++ b/doc/CHANGES-staging/bridge_snapshots_and_cache_changes
@@ -0,0 +1,51 @@
+Subject: 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.
+
+Subject: Bridging
+
+The following APIs are no longer available since the stasis cache
+is no longer used:
+ ast_bridge_topic_cached()
+ ast_bridge_topic_all_cached()
+
+Subject: Bridging
+
+A topic pool is now used for individual bridge topics.
+
+Subject: Bridging
+
+The ast_bridge_cache() function was removed since there's no
+longer a separate container of snapshots.
+
+Subject: Bridging
+
+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.
+
+Subject: Bridging
+
+The ast_bridge_snapshot_get_latest() function was renamed to
+ast_bridge_get_snapshot_by_uniqueid().
+
+Subject: Bridging
+
+A new function "ast_bridge_get_snapshot()" was created to retrieve
+the bridge snapshot directly from the bridge structure.
+
+Subject: Bridging
+
+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.
+
+Subject: Bridging
+
+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.
diff --git a/doc/CHANGES-staging/chan_pjsip_PJSIP_PARSE_URI b/doc/CHANGES-staging/chan_pjsip_PJSIP_PARSE_URI
new file mode 100644
index 0000000..85854ee
--- /dev/null
+++ b/doc/CHANGES-staging/chan_pjsip_PJSIP_PARSE_URI
@@ -0,0 +1,4 @@
+Subject: chan_pjsip
+
+New dialplan function PJSIP_PARSE_URI added to parse an URI and return
+a specified part of the URI.
diff --git a/doc/CHANGES-staging/chan_sip_deprecated b/doc/CHANGES-staging/chan_sip_deprecated
new file mode 100644
index 0000000..04ac21c
--- /dev/null
+++ b/doc/CHANGES-staging/chan_sip_deprecated
@@ -0,0 +1,6 @@
+Subject: 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
diff --git a/doc/CHANGES-staging/channels_remove_stasis_cache_for_channel_snapshots b/doc/CHANGES-staging/channels_remove_stasis_cache_for_channel_snapshots
new file mode 100644
index 0000000..1743fea
--- /dev/null
+++ b/doc/CHANGES-staging/channels_remove_stasis_cache_for_channel_snapshots
@@ -0,0 +1,15 @@
+Subject: 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.
diff --git a/doc/CHANGES-staging/core_ast_bt_get_symbols_return b/doc/CHANGES-staging/core_ast_bt_get_symbols_return
new file mode 100644
index 0000000..a482a4c
--- /dev/null
+++ b/doc/CHANGES-staging/core_ast_bt_get_symbols_return
@@ -0,0 +1,4 @@
+Subject: Core
+
+ast_bt_get_symbols() now returns a vector of strings instead of an
+array of strings. This must be freed with ast_bt_free_symbols.
diff --git a/doc/CHANGES-staging/features_automon_automixmon b/doc/CHANGES-staging/features_automon_automixmon
new file mode 100644
index 0000000..b97c883
--- /dev/null
+++ b/doc/CHANGES-staging/features_automon_automixmon
@@ -0,0 +1,12 @@
+Subject: Features
+
+Before Asterisk 12, when using the automon or automixmon features defined
+in features.conf, a channel variable (TOUCH_MIXMONITOR_OUTPUT) was set on
+both channels, indicating the filename of the recording.
+
+When bridging was overhauled in Asterisk 12, the behavior was changed such
+that the variable was only set on the peer channel and not on the channel
+that initiated the automon or automixmon.
+
+The previous behavior has been restored so both channels receive the
+channel variable when one of these features is invoked.
diff --git a/doc/CHANGES-staging/pbx_config_multiple_global_sections b/doc/CHANGES-staging/pbx_config_multiple_global_sections
new file mode 100644
index 0000000..822cf42
--- /dev/null
+++ b/doc/CHANGES-staging/pbx_config_multiple_global_sections
@@ -0,0 +1,5 @@
+Subject: pbx_config
+
+pbx_config will now find and process multiple 'globals' sections from
+extensions.conf. Variables are processed in the order they are found
+and duplicate variables overwrite the previous value.
diff --git a/doc/CHANGES-staging/res_pjsip_send_contact_status_on_update_registration b/doc/CHANGES-staging/res_pjsip_send_contact_status_on_update_registration
new file mode 100644
index 0000000..e27a0c0
--- /dev/null
+++ b/doc/CHANGES-staging/res_pjsip_send_contact_status_on_update_registration
@@ -0,0 +1,4 @@
+Subject: res_pjsip
+
+Added "send_contact_status_on_update_registration" global configuration option
+to enable sending AMI ContactStatus event when a device refreshes its registration.
diff --git a/doc/CHANGES-staging/res_pjsip_trust_and_send_connected_line_options b/doc/CHANGES-staging/res_pjsip_trust_and_send_connected_line_options
new file mode 100644
index 0000000..44a4c05
--- /dev/null
+++ b/doc/CHANGES-staging/res_pjsip_trust_and_send_connected_line_options
@@ -0,0 +1,8 @@
+Subject: res_pjsip
+
+New options 'trust_connected_line' and 'send_connected_line' have been
+added to the endpoint. The option 'trust_connected_line' is to control
+if connected line updates are accepted from this endpoint.
+The option 'send_connected_line' is to control if connected line updates
+can be sent to this endpoint.
+The default value is 'yes' for both options.
diff --git a/doc/CHANGES-staging/res_rtp_asterisk_strictrtp_seqno_option b/doc/CHANGES-staging/res_rtp_asterisk_strictrtp_seqno_option
new file mode 100644
index 0000000..a2db17b
--- /dev/null
+++ b/doc/CHANGES-staging/res_rtp_asterisk_strictrtp_seqno_option
@@ -0,0 +1,6 @@
+Subject: res_rtp_asterisk
+
+The existing strictrtp option in rtp.conf has a new choice availabe, called
+'seqno', which behaves the same way as setting strictrtp to 'yes', but will
+ignore the time interval during learning so that bursts of packets can still
+trigger learning our source.
diff --git a/doc/UPGRADE-staging/README b/doc/UPGRADE-staging/README
new file mode 100644
index 0000000..95ec1ff
--- /dev/null
+++ b/doc/UPGRADE-staging/README
@@ -0,0 +1,22 @@
+DO NOT REMOVE THIS FILE!
+
+The only files that should be added to this directory are ones that will be
+used by the release script to update the UPGRADE.txt file automatically. The
+only time that it is necessary to add something to the UPGRADE-staging directory
+is if you are making a breaking change to an existing feature in Asterisk. The
+file does not need to have a meaningful name, but it probably should. If there
+are multiple items that need documenting, each can be separated with a subject
+line, which should always start with "SUBJECT:", followed by the subject of the
+change. For example, if you are making a change to PJSIP, then you might add the
+file "res_pjsip_breaking_change" to this directory, with a short description of
+what it does. Here's an example:
+
+SUBJECT: res_pjsip
+
+Here's a pretty good description of what I changed that explains exactly what
+it does and why it breaks things (and why they needed to be broken).
+
+SUBJECT: core
+
+Here's another description of something else I added that is a big enough
+change to warrant another entry in the UPDATE.txt file.
diff --git a/doc/UPGRADE-staging/bridge_snapshots_and_cache_changes b/doc/UPGRADE-staging/bridge_snapshots_and_cache_changes
new file mode 100644
index 0000000..ebcca2e
--- /dev/null
+++ b/doc/UPGRADE-staging/bridge_snapshots_and_cache_changes
@@ -0,0 +1,51 @@
+Subject: 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.
+
+Subject: Bridging
+
+The following APIs are no longer available since the stasis cache
+is no longer used:
+ ast_bridge_topic_cached()
+ ast_bridge_topic_all_cached()
+
+Subject: Bridging
+
+A topic pool is now used for individual bridge topics.
+
+Subject: Bridging
+
+The ast_bridge_cache() function was removed since there's no
+longer a separate container of snapshots.
+
+Subject: Bridging
+
+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.
+
+Subject: Bridging
+
+The ast_bridge_snapshot_get_latest() function was renamed to
+ast_bridge_get_snapshot_by_uniqueid().
+
+Subject: Bridging
+
+A new function "ast_bridge_get_snapshot()" was created to retrieve
+the bridge snapshot directly from the bridge structure.
+
+Subject: Bridging
+
+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.
+
+Subject: Bridging
+
+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.
diff --git a/doc/UPGRADE-staging/chan_sip_deprecated b/doc/UPGRADE-staging/chan_sip_deprecated
new file mode 100644
index 0000000..04ac21c
--- /dev/null
+++ b/doc/UPGRADE-staging/chan_sip_deprecated
@@ -0,0 +1,6 @@
+Subject: 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
diff --git a/doc/UPGRADE-staging/channels_remove_stasis_cache_for_channel_snapshots b/doc/UPGRADE-staging/channels_remove_stasis_cache_for_channel_snapshots
new file mode 100644
index 0000000..1743fea
--- /dev/null
+++ b/doc/UPGRADE-staging/channels_remove_stasis_cache_for_channel_snapshots
@@ -0,0 +1,15 @@
+Subject: 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.
diff --git a/doc/UPGRADE-staging/func_callerid_remove_CALLERPRES b/doc/UPGRADE-staging/func_callerid_remove_CALLERPRES
new file mode 100644
index 0000000..b128cc6
--- /dev/null
+++ b/doc/UPGRADE-staging/func_callerid_remove_CALLERPRES
@@ -0,0 +1,4 @@
+Subject: func_callerid
+
+The CALLERPRES() dialplan function, deprecated in Asterisk 1.8, has been
+removed.
diff --git a/doc/UPGRADE-staging/res_parking_PARKINGSLOT_no_longer_set b/doc/UPGRADE-staging/res_parking_PARKINGSLOT_no_longer_set
new file mode 100644
index 0000000..1ddcaeb
--- /dev/null
+++ b/doc/UPGRADE-staging/res_parking_PARKINGSLOT_no_longer_set
@@ -0,0 +1,4 @@
+Subject: res_parking
+
+The PARKINGSLOT channel variable, deprecated in Asterisk 12 in favor of the
+PARKING_SPACE channel variable, will no longer be set.
diff --git a/doc/UPGRADE-staging/res_xmpp_remove_JabberStatus_application b/doc/UPGRADE-staging/res_xmpp_remove_JabberStatus_application
new file mode 100644
index 0000000..b0aac75
--- /dev/null
+++ b/doc/UPGRADE-staging/res_xmpp_remove_JabberStatus_application
@@ -0,0 +1,7 @@
+Subject: res_xmpp
+
+The JabberStatus application, deprecated in Asterisk 12, has been removed.
+
+Subject: Applications
+
+The JabberStatus application, deprecated in Asterisk 12, has been removed.
--
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Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-MessageType: newchange
Gerrit-Change-Id: I19cf4b569321c88155a65e9b0b80f6d58075dd47
Gerrit-Change-Number: 10945
Gerrit-PatchSet: 1
Gerrit-Owner: Benjamin Keith Ford <bford at digium.com>
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