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asterisk/ChangeLogs/ChangeLog-20.3.0.md

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Change Log for Release 20.3.0
========================================
Summary:
----------------------------------------
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
User Notes:
----------------------------------------
- ### cel: add local optimization begin event
The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used
by itself or in conert with the existing
AST_CEL_LOCAL_OPTIMIZE to book-end local channel optimizaion.
- ### chan_dahdi: Add dialmode option for FXS lines.
A "dialmode" option has been added which allows
specifying, on a per-channel basis, what methods of
subscriber dialing (pulse and/or tone) are permitted.
Additionally, this can be changed on a channel
at any point during a call using the CHANNEL
function.
- ### pbx_dundi: Add PJSIP support.
DUNDi now supports chan_pjsip. Outgoing calls using
PJSIP require the pjsip_outgoing_endpoint option
to be set in dundi.conf.
- ### cli: increase channel column width
This change increases the display width on 'core show channels'
amd 'core show channels verbose'
For 'core show channels', the Channel name field is increased to
64 characters and the Location name field is increased to 32
characters.
For 'core show channels verbose', the Channel name field is
increased to 80 characters, the Context is increased to 24
characters and the Extension is increased to 24 characters.
- ### app_senddtmf: Add SendFlash AMI action.
The SendFlash AMI action now allows sending
a hook flash event on a channel.
- ### res_http_media_cache: Introduce options and customize
The res_http_media_cache module now attempts to load
configuration from the res_http_media_cache.conf file.
The following options were added:
* timeout_secs
* user_agent
* follow_location
* max_redirects
* protocols
* redirect_protocols
* dns_cache_timeout_secs
- ### test.c: Fix counting of tests and add 2 new tests
The "tests" attribute of the "testsuite" element in the
output XML now reflects only the tests actually requested
to be executed instead of all the tests registered.
The "failures" attribute was added to the "testsuite"
element.
Also added two new unit tests that just pass and fail
to be used for testing CI itself.
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
It is now possible to specify the MixMonitorID when calling
the manager action: MixMonitorMute. This will allow an
individual MixMonitor instance to be muted via ID.
The MixMonitorID can be stored as a channel variable using
the 'i' MixMonitor option and is returned upon creation if
this option is used.
As part of this change, if no MixMonitorID is specified in
the manager action MixMonitorMute, Asterisk will set the mute
flag on all MixMonitor audiohooks on the channel. Previous
behavior would set the flag on the first MixMonitor audiohook
found.
- ### bridge_builtin_features: add beep via touch variable
Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid
interval in seconds will result in a periodic beep being
played to the monitored channel upon MixMontior/Monitor
feature start.
If an interval less than 5 seconds is specified, the interval
will default to 5 seconds. If the value is set to an invalid
interval, the default of 15 seconds will be used.
- ### format_sln: add .slin as supported file extension
format_sln now recognizes '.slin' as a valid
file extension in addition to the existing
'.sln' and '.raw'.
Upgrade Notes:
----------------------------------------
- ### cel: add local optimization begin event
The existing AST_CEL_LOCAL_OPTIMIZE can continue
to be used as-is and the AST_CEL_LOCAL_OPTIMIZE_BEGIN event
can be ignored if desired.
Closed Issues:
----------------------------------------
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
Commits By Author:
----------------------------------------
- ### Asterisk Development Team (1):
- Update for 20.3.0-rc1
- ### Fabrice Fontaine (2):
- main/iostream.c: fix build with libressl
- configure: fix detection of re-entrant resolver functions
- ### George Joseph (13):
- make_version: Strip svn stuff and suppress ref HEAD errors
- test.c: Fix counting of tests and add 2 new tests
- Initial GitHub Issue Templates
- Initial GitHub PRs
- .github: Remove unnecessary parameter in CherryPickTest
- .github: Update issue templates
- .github: Add cherry-pick test progress labels
- .github: Refactor CP progress and add new PR test progress
- .github: Remove separate set labels step from new PR
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Fix CherryPickTest to only run when it should
- .github: Add AsteriskReleaser
- Set up new ChangeLogs directory
- ### Henning Westerholt (2):
- chan_pjsip: fix music on hold continues after INVITE with replaces
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- ### Holger Hans Peter Freyther (1):
- res_http_media_cache: Introduce options and customize
- ### Jaco Kroon (2):
- app_queue: periodic announcement configurable start time.
- res_calendar: output busy state as part of show calendar.
- ### Joshua C. Colp (2):
- pbx_dundi: Fix PJSIP endpoint configuration check.
- LICENSE: Update link to trademark policy.
- ### Joshua Colp (1):
- Revert "app_queue: periodic announcement configurable start time."
- ### Maximilian Fridrich (1):
- res_pjsip: mediasec: Add Security-Client headers after 401
- ### Mike Bradeen (5):
- cli: increase channel column width
- format_sln: add .slin as supported file extension
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- bridge_builtin_features: add beep via touch variable
- cel: add local optimization begin event
- ### Naveen Albert (8):
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- app_senddtmf: Add SendFlash AMI action.
- func_json: Fix JSON parsing issues.
- app_queue: Fix minor xmldoc duplication and vagueness.
- voicemail.conf: Fix incorrect comment about #include.
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- chan_dahdi: Add dialmode option for FXS lines.
- ### Sean Bright (5):
- contrib: rc.archlinux.asterisk uses invalid redirect.
- res_agi: RECORD FILE plays 2 beeps.
- ael: Regenerate lexers and parsers.
- loader.c: Minor module key check simplification.
- core: Cleanup gerrit and JIRA references. (#57)
- ### The_Blode (1):
- install_prereq: Add Linux Mint support.
Detail:
----------------------------------------
- ### Set up new ChangeLogs directory
Author: George Joseph
Date: 2023-05-09
- ### .github: Add AsteriskReleaser
Author: George Joseph
Date: 2023-05-05
- ### chan_pjsip: also return all codecs on empty re-INVITE for late offers
Author: Henning Westerholt
Date: 2023-05-03
We should also return all codecs on an re-INVITE without SDP for a
call that used late offer (e.g. no SDP in the initial INVITE, SDP
in the ACK). Bugfix for feature introduced in ASTERISK-30193
(https://issues.asterisk.org/jira/browse/ASTERISK-30193)
Migration from previous gerrit change that was not merged.
- ### cel: add local optimization begin event
Author: Mike Bradeen
Date: 2023-05-02
The current AST_CEL_LOCAL_OPTIMIZE event is and has been
triggered on a local optimization end to serve as a flag
indicating the event occurred. This change adds a second
AST_CEL_LOCAL_OPTIMIZE_BEGIN event for further detail.
Resolves: #52
UpgradeNote: The existing AST_CEL_LOCAL_OPTIMIZE can continue
to be used as-is and the AST_CEL_LOCAL_OPTIMIZE_BEGIN event
can be ignored if desired.
UserNote: The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used
by itself or in conert with the existing
AST_CEL_LOCAL_OPTIMIZE to book-end local channel optimizaion.
- ### core: Cleanup gerrit and JIRA references. (#57)
Author: Sean Bright
Date: 2023-05-03
* Remove .gitreview and switch to pulling the main asterisk branch
version from configure.ac instead.
* Replace references to JIRA with GitHub.
* Other minor cleanup found along the way.
Resolves: #39
- ### .github: Fix CherryPickTest to only run when it should
Author: George Joseph
Date: 2023-05-03
Fixed CherryPickTest so it triggers only on the
"cherry-pick-test" label instead of all labels.
- ### .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
Author: George Joseph
Date: 2023-05-02
- ### .github: Remove separate set labels step from new PR
Author: George Joseph
Date: 2023-05-02
- ### .github: Refactor CP progress and add new PR test progress
Author: George Joseph
Date: 2023-05-02
- ### res_pjsip: mediasec: Add Security-Client headers after 401
Author: Maximilian Fridrich
Date: 2023-05-02
When using mediasec, requests sent after a 401 must still contain the
Security-Client header according to
draft-dawes-sipcore-mediasec-parameter.
Resolves: #48
- ### .github: Add cherry-pick test progress labels
Author: George Joseph
Date: 2023-05-02
- ### LICENSE: Update link to trademark policy.
Author: Joshua C. Colp
Date: 2023-05-01
Resolves: #43
- ### chan_dahdi: Add dialmode option for FXS lines.
Author: Naveen Albert
Date: 2023-04-28
Currently, both pulse and tone dialing are always enabled
on all FXS lines, with no way of disabling one or the other.
In some circumstances, it is desirable or necessary to
disable one of these, and this behavior can be problematic.
A new "dialmode" option is added which allows setting the
methods to support on a per channel basis for FXS (FXO
signalled lines). The four options are "both", "pulse",
"dtmf"/"tone", and "none".
Additionally, integration with the CHANNEL function is
added so that this setting can be updated for a channel
during a call.
Resolves: #35
ASTERISK-29992
UserNote: A "dialmode" option has been added which allows
specifying, on a per-channel basis, what methods of
subscriber dialing (pulse and/or tone) are permitted.
Additionally, this can be changed on a channel
at any point during a call using the CHANNEL
function.
- ### .github: Update issue templates
Author: George Joseph
Date: 2023-05-01
- ### .github: Remove unnecessary parameter in CherryPickTest
Author: George Joseph
Date: 2023-05-01
- ### Initial GitHub PRs
Author: George Joseph
Date: 2023-04-28
- ### Initial GitHub Issue Templates
Author: George Joseph
Date: 2023-04-28
- ### pbx_dundi: Fix PJSIP endpoint configuration check.
Author: Joshua C. Colp
Date: 2023-04-13
ASTERISK-28233
- ### Revert "app_queue: periodic announcement configurable start time."
Author: Joshua Colp
Date: 2023-04-11
This reverts commit 3fd0b65bae4b1b14434737ffcf0da4aa9ff717f6.
Reason for revert: Causes segmentation fault.
- ### res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
Author: Naveen Albert
Date: 2023-02-17
The current STIR/SHAKEN signing process is inconsistent with the
RFCs in a couple ways that can cause interoperability issues.
RFC8225 specifies that the keys must be ordered lexicographically, but
currently the fields are simply ordered according to the order
in which they were added to the JSON object, which is not
compliant with the RFC and can cause issues with some carriers.
To fix this, we now leverage libjansson's ability to dump a JSON
object sorted by key value, yielding the correct field ordering.
Additionally, telephone numbers must have any leading + prefix removed
and must not contain characters outside of 0-9, *, and # in order
to comply with the RFCs. Numbers are now properly formatted as such.
ASTERISK-30407 #close
- ### pbx_dundi: Add PJSIP support.
Author: Naveen Albert
Date: 2022-12-09
Adds PJSIP as a supported technology to DUNDi.
To facilitate this, we now allow an endpoint to be specified
for outgoing PJSIP calls. We also allow users to force a specific
channel technology for outgoing SIP-protocol calls.
ASTERISK-28109 #close
ASTERISK-28233 #close
- ### install_prereq: Add Linux Mint support.
Author: The_Blode
Date: 2023-03-17
ASTERISK-30359 #close
- ### chan_pjsip: fix music on hold continues after INVITE with replaces
Author: Henning Westerholt
Date: 2023-03-21
In a three party scenario with INVITE with replaces, we need to
unhold the call, otherwise one party continues to get music on
hold, and the call is not properly bridged between them.
ASTERISK-30428
- ### voicemail.conf: Fix incorrect comment about #include.
Author: Naveen Albert
Date: 2023-03-28
A comment at the top of voicemail.conf says that #include
cannot be used in voicemail.conf because this breaks
the ability for app_voicemail to auto-update passwords.
This is factually incorrect, since Asterisk has no problem
updating files that are #include'd in the main configuration
file, and this does work in voicemail.conf as well.
ASTERISK-30479 #close
- ### app_queue: Fix minor xmldoc duplication and vagueness.
Author: Naveen Albert
Date: 2023-04-03
The F option in the xmldocs for the Queue application
was erroneously duplicated, causing it to display
twice on the wiki. The two sections are now merged into one.
Additionally, the description for the d option was quite
vague. Some more details are added to provide context
as to what this actually does.
ASTERISK-30486 #close
- ### test.c: Fix counting of tests and add 2 new tests
Author: George Joseph
Date: 2023-03-28
The unit test XML output was counting all registered tests as "run"
even when only a subset were actually requested to be run and
the "failures" attribute was missing.
* The "tests" attribute of the "testsuite" element in the
output XML now reflects only the tests actually requested
to be executed instead of all the tests registered.
* The "failures" attribute was added to the "testsuite"
element.
Also added 2 new unit tests that just pass and fail to be
used for CI testing.
- ### res_calendar: output busy state as part of show calendar.
Author: Jaco Kroon
Date: 2023-03-23
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
- ### loader.c: Minor module key check simplification.
Author: Sean Bright
Date: 2023-03-23
- ### ael: Regenerate lexers and parsers.
Author: Sean Bright
Date: 2023-03-21
Various changes to ensure that the lexers and parsers can be correctly
generated when REBUILD_PARSERS is enabled.
Some notes:
* Because of the version of flex we are using to generate the lexers
(2.5.35) some post-processing in the Makefile is still required.
* The generated lexers do not contain the problematic C99 check that
was being replaced by the call to sed in the respective Makefiles so
it was removed.
* Since these files are generated, they will include trailing
whitespace in some places. This does not need to be corrected.
- ### bridge_builtin_features: add beep via touch variable
Author: Mike Bradeen
Date: 2023-03-01
Add periodic beep option to one-touch recording by setting
the touch variable TOUCH_MONITOR_BEEP or
TOUCH_MIXMONITOR_BEEP to the desired interval in seconds.
If the interval is less than 5 seconds, a minimum of 5
seconds will be imposed. If the interval is set to an
invalid value, it will default to 15 seconds.
A new test event PERIODIC_HOOK_ENABLED was added to the
func_periodic_hook hook_on function to indicate when
a hook is started. This is so we can test that the touch
variable starts the hook as expected.
ASTERISK-30446
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
Author: Mike Bradeen
Date: 2023-03-13
While it is possible to create multiple mixmonitor instances
on a channel, it was not previously possible to mute individual
instances.
This change includes the ability to specify the MixMonitorID
when calling the manager action: MixMonitorMute. This will
allow an individual MixMonitor instance to be muted via id.
This id can be stored as a channel variable using the 'i'
MixMonitor option.
As part of this change, if no MixMonitorID is specified in
the manager action MixMonitorMute, Asterisk will set the mute
flag on all MixMonitor spy-type audiohooks on the channel.
This is done via the new audiohook function:
ast_audiohook_set_mute_all.
ASTERISK-30464
- ### format_sln: add .slin as supported file extension
Author: Mike Bradeen
Date: 2023-03-14
Adds '.slin' to existing supported file extensions:
.sln and .raw
ASTERISK-30465
- ### res_agi: RECORD FILE plays 2 beeps.
Author: Sean Bright
Date: 2023-03-08
Sending the "RECORD FILE" command without the optional
`offset_samples` argument can result in two beeps playing on the
channel.
This bug has been present since Asterisk 0.3.0 (2003-02-06).
ASTERISK-30457 #close
- ### func_json: Fix JSON parsing issues.
Author: Naveen Albert
Date: 2023-02-26
Fix issue with returning empty instead of dumping
the JSON string when recursing.
Also adds a unit test to capture this fix.
ASTERISK-30441 #close
- ### app_senddtmf: Add SendFlash AMI action.
Author: Naveen Albert
Date: 2023-02-26
Adds an AMI action to send a flash event
on a channel.
ASTERISK-30440 #close
- ### app_dial: Fix DTMF not relayed to caller on unanswered calls.
Author: Naveen Albert
Date: 2023-03-04
DTMF frames are not handled in app_dial when sent towards the
caller. This means that if DTMF is sent to the calling party
and the call has not yet been answered, the DTMF is not audible.
This is now fixed by relaying DTMF frames if only a single
destination is being dialed.
ASTERISK-29516 #close
- ### configure: fix detection of re-entrant resolver functions
Author: Fabrice Fontaine
Date: 2023-03-08
uClibc does not provide res_nsearch:
asterisk-16.0.0/main/dns.c:506: undefined reference to `res_nsearch'
Patch coded by Yann E. MORIN:
http://lists.busybox.net/pipermail/buildroot/2018-October/232630.html
ASTERISK-21795 #close
Signed-off-by: Bernd Kuhls <bernd.kuhls@t-online.de>
[Retrieved from:
https: //git.buildroot.net/buildroot/tree/package/asterisk/0005-configure-fix-detection-of-re-entrant-resolver-funct.patch]
Signed-off-by: Fabrice Fontaine <fontaine.fabrice@gmail.com>
- ### cli: increase channel column width
Author: Mike Bradeen
Date: 2023-03-06
For 'core show channels', the Channel name field is increased
to 64 characters and the Location name field is increased to
32 characters.
For 'core show channels verbose', the Channel name field is
increased to 80 characters, the Context is increased to 24
characters and the Extension is increased to 24 characters.
ASTERISK-30455
- ### app_queue: periodic announcement configurable start time.
Author: Jaco Kroon
Date: 2023-02-21
This newly introduced periodic-announce-startdelay makes it possible to
configure the initial start delay of the first periodic announcement
after which periodic-announce-frequency takes over.
ASTERISK-30437 #close
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
- ### make_version: Strip svn stuff and suppress ref HEAD errors
Author: George Joseph
Date: 2023-03-13
* All of the code that used subversion has been removed.
* When Asterisk is checked out from a tag or commit instead
of one of the regular branches, git would emit messages like
"fatal: ref HEAD is not a symbolic ref" which weren't fatal
at all. Those are now suppressed.
- ### res_http_media_cache: Introduce options and customize
Author: Holger Hans Peter Freyther
Date: 2022-10-16
Make the existing CURL parameters configurable and allow
to specify the usable protocols, proxy and DNS timeout.
ASTERISK-30340
- ### main/iostream.c: fix build with libressl
Author: Fabrice Fontaine
Date: 2023-02-25
Fix the following build failure with libressl by using SSL_is_server
which is available since version 2.7.0 and
https://github.com/libressl-portable/openbsd/commit/d7ec516916c5eaac29b02d7a8ac6570f63b458f7:
iostream.c: In function 'ast_iostream_close':
iostream.c:559:41: error: invalid use of incomplete typedef 'SSL' {aka 'struct ssl_st'}
559 | if (!stream->ssl->server) {
| ^~
ASTERISK-30107 #close
Fixes: - http://autobuild.buildroot.org/results/ce4d62d00bb77ba5b303cacf6be7e350581a62f9
- ### contrib: rc.archlinux.asterisk uses invalid redirect.
Author: Sean Bright
Date: 2023-03-02
`rc.archlinux.asterisk`, which explicitly requests bash in its
shebang, uses the following command syntax:
${DAEMON} -rx "core stop now" > /dev/null 2&>1
The intent of which is to execute:
${DAEMON} -rx "core stop now"
While sending both stdout and stderr to `/dev/null`. Unfortunately,
because the `&` is in the wrong place, bash is interpreting the `2` as
just an additional argument to the `$DAEMON` command and not as a file
descriptor and proceeds to use the bashism `&>` to send stderr and
stdout to a file named `1`.
So we clean it up and just use bash's shortcut syntax.
Issue raised and a fix suggested (but not used) by peutch on GitHub¹.
ASTERISK-30449 #close
1. https://github.com/asterisk/asterisk/pull/31
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