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Update CHANGES and UPGRADE.txt for 17.0.0

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17.1
Asterisk Development Team 6 years ago
parent 4f2ad5a55d
commit f1e0c9d1b8
24 changed files with 243 additions and 212 deletions
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131
CHANGES
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@ -12,6 +12,137 @@
=== ===
============================================================================== ==============================================================================
------------------------------------------------------------------------------
--- 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 ---------- --- Functionality changes from Asterisk 16.2.0 to Asterisk 16.3.0 ----------
------------------------------------------------------------------------------ ------------------------------------------------------------------------------

112
UPGRADE.txt
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@ -18,6 +18,118 @@
=== ===
=========================================================== ===========================================================
------------------------------------------------------------------------------
--- 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: New in 16.0.0:
app_fax: app_fax:

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

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

7
doc/CHANGES-staging/app_confbridge_remb_behavior_all.txt
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@ -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..

12
doc/CHANGES-staging/app_dial_ringtime_progresstime.txt
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@ -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

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

36
doc/CHANGES-staging/bridging_stasis_cache.txt
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@ -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.

7
doc/CHANGES-staging/chan_sip_deprecated.txt
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@ -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

16
doc/CHANGES-staging/channels_stasis_cache.txt
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@ -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.

3
doc/CHANGES-staging/pbx_dundi_ipv6.txt
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@ -1,3 +0,0 @@
Subject: pbx_dundi
The DUNDi PBX module now supports IPv4/IPv6 dual binding.

13
doc/CHANGES-staging/res_pjsip_add_norefersub_global_config.txt
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@ -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.

5
doc/CHANGES-staging/res_rtp_asterisk_dtls_fragmentation.txt
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@ -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.

5
doc/CHANGES-staging/rtp_ice_include_local_address.txt
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@ -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.

9
doc/UPGRADE-staging/Build_all_3_app_voicemail_variants_at_the_same_time.txt
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@ -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.

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

36
doc/UPGRADE-staging/bridging_stasis_cache.txt
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@ -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.

5
doc/UPGRADE-staging/chan_pjsip_refer_fix.txt
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@ -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.

7
doc/UPGRADE-staging/chan_sip_deprecated.txt
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@ -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

16
doc/UPGRADE-staging/channels_stasis_cache.txt
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@ -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.

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

7
doc/UPGRADE-staging/install_headers.txt
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@ -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".

5
doc/UPGRADE-staging/res_parking_parkingslot.txt
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@ -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.

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