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*: recursive bump for jpeg -> libjpeg-turbo switch
*: recursive bump for default Kerberos implementation switch
*: recursive bump for icu 77 and libxml2 2.14
asterisk[19,21,22]: Fix invalid XML documentation building
revbump packages due to devel/libslang2 removal
*: recursive bump for pango requiring fontconfig 2.15
Correct PJPROJ version number.
asterisk19: revert previous while packages should not download files during build, that's not a reason to mark them BROKEN.
asterisk19: Mark BROKEN
*: recursive bump for default-on option of at-spi2-core
*: recursive bump for icu 76 shlib major version bump
*: revbump for icu downgrade
*: recursive bump for icu 76.1 shlib bump
*: recursive bump for merging at-spi2-atk and atk into at2-spi-core Remove at-spi2-atk and atk
Update to Asterisk 19.8.1.
Note that the Asterisk 19.* series is EOL and this package will be
scheduled for deletion in one to two quarters.
pkgsrc changes:
- MKPIE_SUPPORTED=NO -- eol, so not worth effort to fix
- various new/obsoleted config files / docs
- new/obsoleted features
+ app_sf
+ func_evalexten
+ func_export
+ func_json
+ res_ari_mailboxes
+ res_geolocation
+ res_mwi_external
+ res_mwi_external_ami
+ res_pjsip_geolocation
+ res_pjsip_rfc3329
+ res_speech_aeap
+ res_stasis_playback
Change Log for Release 19.8.1
========================================
Summary:
----------------------------------------
- apply_patches: Use globbing instead of file/sort.
- bundled_pjproject: Backport 2 SSL patches from upstream
- bundled_pjproject: Backport security fixes from pjproject 2.13.1
- apply_patches: Sort patch list before applying
Closed Issues:
----------------------------------------
- #188: [improvement]: pjsip: Upgrade bundled version to pjproject 2.13.1 #187
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before applying
- #194: [bug]: Segfault/double-free in bundled pjproject using TLS transport
Commits By Author:
----------------------------------------
- ### George Joseph (3):
- apply_patches: Sort patch list before applying
- bundled_pjproject: Backport security fixes from pjproject 2.13.1
- bundled_pjproject: Backport 2 SSL patches from upstream
- ### Sean Bright (1):
- apply_patches: Use globbing instead of file/sort.
-----
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 19.7.0 to Asterisk 19.8.0 ------------
------------------------------------------------------------------------------
cdr
------------------
* Two new options have been added which allow
bridging and dial state changes to be ignored
in CDRs, which can be useful if a single CDR
is desired for a channel.
res_pjsip
------------------
* Added options "security_negotiation" and "security_mechanisms" to pjsip
endpoints and registrations. "security_negotiation" can be set to "no" (default)
or "mediasec", and "security_mechanisms" can be a list of comma-separated
security_mechanisms in the form defined by RFC 3329 section 2.2.
* A new option named "all_codecs_on_empty_reinvite" has been added to the
global section. When this option is enabled, on reception of a re-INVITE
without SDP, Asterisk will send an SDP offer in the 200 OK response containing
all configured codecs on the endpoint, instead of simply those that have
already been negotiated. RFC 3261 specifies this as a SHOULD requirement.
The default value is "off".
res_pjsip_logger
------------------
* SIP messages can now be filtered by SIP request method
(INVITE, CANCEL, ACK, BYE, REGISTER, OPTION,
SUBSCRIBE, NOTIFY, PUBLISH, INFO, and MESSAGE),
allowing for more granular debugging to be done
in the CLI. This applies to requests but not responses.
res_pjsip_notify
------------------
* Allows using the config options in pjsip_notify.conf
from AMI actions as with the existing CLI commands.
res_tonedetect
------------------
* The TONE_DETECT function now supports
detection of audible ringback tone
using the p option.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 19.6.0 to Asterisk 19.7.0 ------------
------------------------------------------------------------------------------
New EXPORT function
------------------
* A new function, EXPORT, allows writing variables
and functions on other channels, the complement
of the IMPORT function.
app_amd
------------------
* An audio file to play during AMD processing can
now be specified to the AMD application or configured
in the amd.conf configuration file.
app_bridgewait
------------------
* Adds the n option to not answer the channel when
the BridgeWait application is called.
features
------------------
* The Bridge application now has the n "no answer" option
that can be used to prevent the channel from being
automatically answered prior to bridging.
func_strings
------------------
* Three new functions, TRIM, LTRIM, and RTRIM, are
now available for trimming leading and trailing
whitespace.
res_pjsip
------------------
* A new option named "peer_supported" has been added to the endpoint option
100rel. When set to this option, Asterisk sends provisional responses
reliably if the peer supports it. If the peer does not support reliable
provisional responses, Asterisk sends them normally.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 19.6.0 to Asterisk 19.7.0 ------------
------------------------------------------------------------------------------
Transfer feature
------------------
* The following capabilities have been added to the
transfer feature:
- The transfer initiation announcement prompt can
now be customized in features.conf.
- The TRANSFER_EXTEN variable now can be set on the
transferer's channel in order to allow the transfer
function to automatically attempt to go to the extension
contained in this variable, if it exists. The transfer
context behavior is not changed (TRANSFER_CONTEXT is used
if it exists; otherwise the default context is used).
app_confbridge
------------------
* Adds the end_marked_any option which can be used
to kick users from a conference after any
marked user leaves (including marked users).
locks
------------------
* A new AMI event, DeadlockStart, is now available
when Asterisk is compiled with DETECT_DEADLOCKS,
and can indicate that a deadlock has occured.
res_geolocation
------------------
* Added 4 built-in profiles:
"<prefer_config>"
"<discard_config>"
"<prefer_incoming>"
"<discard_incoming>"
The profiles are empty except for having their precedence
set.
Added profile parameter "suppress_empty_ca_elements" that
will cause Civic Address elements that are empty to be
suppressed from the outgoing PIDF-LO document.
You can now specify the location object's format, location_info,
method, location_source and confidence parameters directly on
a profile object for simple scenarios where the location
information isn't common with any other profiles. This is
mutually exclusive with setting location_reference on the
profile.
Added an 'a' option to the GEOLOC_PROFILE function to allow
variable lists like location_info_refinement to be appended
to instead of replacing the entire list.
Added an 'r' option to the GEOLOC_PROFILE function to resolve all
variables before a read operation and after a Set operation.
res_musiconhold_answeredonly
------------------
* This change adds an option, answeredonly, that will prevent music
on hold on channels that are not answered.
res_pjsip
------------------
* TLS transports in res_pjsip can now reload their TLS certificate
and private key files, provided the filename of them has not
changed.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 19.5.0 to Asterisk 19.6.0 ------------
------------------------------------------------------------------------------
res_geolocation
------------------
* * Added processing for the 'confidence' element.
* Added documentation to some APIs.
* removed a lot of complex code related to the very-off-nominal
case of needing to process multiple location info sources.
* Create a new 'ast_geoloc_eprofile_to_pidf' API that just takes
one eprofile instead of a datastore of multiples.
* Plugged a huge leak in XML processing that arose from
insufficient documentation by the libxml/libxslt authors.
* Refactored stylesheets to be more efficient.
* Renamed 'profile_action' to 'profile_precedence' to better
reflect it's purpose.
* Added the config option for 'allow_routing_use' which
sets the value of the 'Geolocation-Routing' header.
* Removed the GeolocProfileCreate and GeolocProfileDelete
dialplan apps.
* Changed the GEOLOC_PROFILE dialplan function as follows:
* Removed the 'profile' argument.
* Automatically create a profile if it doesn't exist.
* Delete a profile if 'inheritable' is set to no.
* Fixed various bugs and leaks
* Updated Asterisk WiKi documentation.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 19.5.0 to Asterisk 19.6.0 ------------
------------------------------------------------------------------------------
chan_dahdi
------------------
* A POLARITY function is now available that allows
getting or setting the polarity on a channel
from the dialplan.
db
------------------
* The DBPrefixGet AMI action now allows retrieving
all of the DB keys beginning with a particular
prefix.
res_cliexec
------------------
* A new CLI command, dialplan exec application, has
been added which allows dialplan applications to be
executed at the CLI, useful for some quick testing
without needing to write dialplan.
res_geolocation
------------------
* Added res_geolocation which creates the core capabilities
to manipulate Geolocation information on SIP INVITEs.
res_pjsip
------------------
* A new transport option 'allow_wildcard_certs' has been added that when it
and 'verify_server' are both set to 'yes', enables verification against
wildcards, i.e. '*.' in certs for common, and subject alt names of type DNS
for TLS transport types. Names must start with the wildcard. Partial wildcards,
e.g. 'f*.example.com' and 'foo.*.com' are not allowed. As well, names only
match against a single level meaning '*.example.com' matches 'foo.example.com',
but not 'foo.bar.example.com'.
res_pjsip_geolocation
------------------
* Added res_pjsip_geolocation which gives chan_pjsip
the ability to use the core geolocation capabilities.
res_pjsip_header_funcs
------------------
* Add function PJSIP_RESPONSE_HEADERS() to get list of header names from 200 response, in the same way as PJSIP_HEADERS() from the request.
Add function PJSIP_RESPONSE_HEADER() to read header from 200 response, in the same way as PJSIP_HEADER() from the request.
------------------------------------------------------------------------------
--- 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
to evaluate dialplan functions and variables
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 ------------
------------------------------------------------------------------------------
func_odbc
------------------
* A SQL_ESC_BACKSLASHES dialplan function has been added which
escapes backslashes. Usage of this is dependent on whether the
database in use can use backslashes to escape ticks or not. If
it can, then usage of this prevents a broken SQL query depending
on how the SQL query is constructed.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 19.2.0 to Asterisk 19.3.0 ------------
------------------------------------------------------------------------------
ami
------------------
* AMI events can now be globally disabled using
the disabledevents [general] setting.
app_mf
------------------
* Adds an option to ReceiveMF to cap the
number of digits read at a user-specified
maximum.
app_queue
------------------
* Load queues and members from Realtime for
AMI actions: QueuePause, QueueStatus and QueueSummary,
Applications: PauseQueueMember and UnpauseQueueMember.
* Added a new AMI action: QueueWithdrawCaller
This AMI action makes it possible to withdraw a caller from a queue
back to the dialplan. The call will be signaled to leave the queue
whenever it can, hence, it not guaranteed that the call will leave
the queue.
Optional custom data can be passed in the request, in the WithdrawInfo
parameter. If the call successfully withdrawn the queue,
it can be retrieved using the QUEUE_WITHDRAW_INFO variable.
This can be useful for certain uses, such as dispatching the call
to a specific extension.
channel_internal_api
------------------
* CHANNEL(lastcontext) and CHANNEL(lastexten)
are now available for use in the dialplan.
res_pjsip_pubsub
------------------
* A new resource_list option, resource_display_name, indicates
whether display name of resource or the resource name being
provided for RLS entries.
If this option is enabled, the Display Name will be provided.
This option is disabled by default to remain the previous behavior.
If the 'event' set to 'presence' or 'dialog' the non-empty HINT name
will be set as the Display Name.
The 'message-summary' is not supported yet.
* The Resource List Subscriptions (RLS) is dynamic now.
The asterisk now updates current subscriptions to reflect the changes
to the list on subscription refresh. If list items are added,
removed, updated or do not exist anymore, the asterisk regenerates
the resource list.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 19.1.0 to Asterisk 19.2.0 ------------
------------------------------------------------------------------------------
Applications
------------------
* added support for Danish syntax, playing the correct plural sound file
dependen on where you have 1 or multipe messages
based on the existing SE/NO code
* added that we set DIALEDPEERNUMBER on the outgoing channels
so it is avalible in b(content^extension^line)
this add the same behaviour as Dial
Core
------------------
* Bundled PJProject Build
The build process has been updated to make pjproject troubleshooting
and development easier. See third-party/pjproject/README-hacking.md or
https://wiki.asterisk.org/wiki/display/AST/Bundled+PJProject
for more info.
ami
------------------
* An AMI event now exists for "Wink".
app_mf
------------------
* Adds MF receiver and sender applications to support
the R1 MF signaling protocol, including integration
with the Dial application.
app_queue
------------------
* added that we set DIALEDPEERNUMBER on the outgoing channels
so it is avalible in b(content^extension^line)
this add the same behaviour as Dial
app_queues
------------------
* adding support for playing the correct en/et for nordic languages
* Don't play sound_thanks if there is no leading hold_time message
When the only announcement is hold time, and there is no hold time (0 min, 0 sec), asterisk will say "thank you for your patience"
app_sendtext
------------------
* A ReceiveText application has been added that can be
used in conjunction with the SendText application.
app_voicemail
------------------
* added support for Danish syntax, playing the correct plural sound file
dependen on where you have 1 or multipe messages
based on the existing SE/NO code
cdr
------------------
* A new CDR option, channeldefaultenabled, allows controlling
whether CDR is enabled or disabled by default on
newly created channels. The default behavior remains
unchanged from previous versions of Asterisk (new
channels will have CDR enabled, as long as CDR is
enabled globally).
chan_sip.c
------------------
* resolve issue with pickup on device that uses "183" and not "180"
cli
------------------
* The "module refresh" command has been added,
which allows unloading and then loading a
module with a single command.
func_json
------------------
* The JSON_DECODE dialplan function can now be used
to parse JSON strings, such as in conjunction with
CURL for using API responses.
res_fax_spandsp
------------------
* Adds support for spandsp 3.0.0.
comms/asterisk19: Roll back partial update to 19.8.1 A partial update to 19.8.1 rode along with fetching from https, but distinfo and more importantly patches did not. I tried moving the patches forward, which was doable but then there were pjsip issues. To get a buildable version on the branch, I have backed out the partial update (but left the https change).
revbump after icu and protobuf updates
switch to https
*: recursive bump for gnutls p11-kit option (existing installations need the bl3.mk included, but it's now only optionally included)
*: bump for cairo buildlink3.mk change lzo was made an option
asterisk*: Attempt to fix PLIST on SunOS
*: recursive bump for cairo dependency changes
*: revebump for new brotli option for freetype2 Addresses PR 57693
*: recursive bump for icu 74.1
*: bump for openssl 3
recursive revbump for tiff update
*: recursive bump for Python 3.11 as new default
revbump after textproc/icu update
*: Recursive revbup from graphics/freetype2
*: recursive bump for tiff shlib major bump
asterisk[18,19]: Add missing libedit dependency (included in base on NetBSD)
massive revision bump after textproc/icu update
*: bump PKGREVISION for libunistring shlib major bump
*: Revbump packages that use Python at runtime without a PKGNAME prefix
*: recursive bump for perl 5.36
asterisk*: Check for NetBSD properly. Use OPSYS_VERSION.
asterisk*: Use OPSYS_VERSION to numerically compare NetBSD versions
revbump for textproc/icu update
Update to Asterisk 19.1.0.
The Asterisk Development Team would like to announce the release
of Asterisk 19.1.0.
The release of Asterisk 19.1.0 resolves several issues reported by
the community and would have not been possible without your
participation.
Thank you!
The following issues are resolved in this release:
New Features made in this release:
-----------------------------------
* ASTERISK-29720 - res_tonedetect: Add call progress tone detection
(Reported by N A)
* ASTERISK-18069 - [patch] app_queue Add Login Time and Last
Paused Times to Queue Members
(Reported by Jamuel Starkey)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-29779 - progdocs: Hidden code sections with syntax errors.
(Reported by Alexander Traud)
* ASTERISK-29732 - progdocs: Fix grouping for latest Doxygen
(Reported by Alexander Traud)
* ASTERISK-29771 - Crash occurs when 2 realtime sippeers mysql
connections are configured and we have a schema warning
(Reported by Mario Ban)
* ASTERISK-29776 - stir/shaken: Requires GNU designator
(Reported by Alexander Traud)
* ASTERISK-29773 - progdocs: doxyref.h outdated
(Reported by Alexander Traud)
* ASTERISK-29765 - xmldoc: Fix for Doxygen
(Reported by Alexander Traud)
* ASTERISK-29762 - channels: Fix for Doxygen
(Reported by Alexander Traud)
* ASTERISK-29730 - Segfault in __ao2_ref if refdebug = yes
(Reported by Alexei Gradinari)
* ASTERISK-29748 - bridging: Infinite loop when both Local
channel halves in same bridge
(Reported by Joshua C. Colp)
* ASTERISK-29754 - odbc: Fix for Doxygen
(Reported by Alexander Traud)
* ASTERISK-29753 - parking: Fix for Doxygen
(Reported by Alexander Traud)
* ASTERISK-29756 - res_ari: Fix for Doxygen
(Reported by Alexander Traud)
* ASTERISK-29755 - frame: Fix for Doxygen
(Reported by Alexander Traud)
* ASTERISK-29751 - channel: Fix for Doxygen
(Reported by Alexander Traud)
* ASTERISK-29752 - app: Fix for Doxygen
(Reported by Alexander Traud)
* ASTERISK-29750 - stasis: Fix for Doxygen
(Reported by Alexander Traud)
* ASTERISK-29749 - res_xmpp: Fix for Doxygen
(Reported by Alexander Traud)
* ASTERISK-29742 - addons: Fix for Doxygen.
(Reported by Alexander Traud)
* ASTERISK-29747 - res_pjsip: Fix for Doxygen
(Reported by Alexander Traud)
* ASTERISK-29737 - chan_iax2: Fix for Doxygen
(Reported by Alexander Traud)
* ASTERISK-29743 - bridges: Fix for Doxygen
(Reported by Alexander Traud)
* ASTERISK-29740 - apps: Fix for Doxygen
(Reported by Alexander Traud)
* ASTERISK-29741 - tests: Fix for Doxygen
(Reported by Alexander Traud)
* ASTERISK-29736 - bridge_channel: Fix for Doxygen
(Reported by Alexander Traud)
* ASTERISK-29734 - progdocs: Use Doxygen \example correctly
(Reported by Alexander Traud)
* ASTERISK-29735 - progdocs: Avoid multiple use of section labels
(Reported by Alexander Traud)
* ASTERISK-29733 - progdocs: Avoid name with Doxygen \file
(Reported by Alexander Traud)
* ASTERISK-29744 - app_morsecode: Fix deadlock
(Reported by N A)
* ASTERISK-29705 - app_read: Fix custom terminator
functionality regression
(Reported by N A)
* ASTERISK-29703 - res_pjsip_callerid: Fix OLI parsing
(Reported by N A)
* ASTERISK-29724 - BuildSystem: In POSIX sh, == in place of = is undefined.
(Reported by Alexander Traud)
* ASTERISK-29702 - sig_analog: Fix truncated buffer copy
(Reported by N A)
* ASTERISK-28040 - pbx: "dialplan reload" is removing minus
symbol from dynamic hints
(Reported by Daniel Zanutti)
* ASTERISK-29391 - VoiceMail does not cancel recording on
rerecord hangup
(Reported by N A)
* ASTERISK-29709 - res_snmp: Not build on recent Debian distributions.
(Reported by Alexander Traud)
* ASTERISK-29717 - res_config_sqlite: not removed in makeopts.in
(Reported by Alexander Traud)
* ASTERISK-29710 - stasis: Clang 13 warns about the unused but
set variable dispatched.
(Reported by Alexander Traud)
* ASTERISK-29711 - aelparse: GCC 11.2 found two maybe uninitialized
(Reported by Alexander Traud)
* ASTERISK-29713 - GCC 11.2: two stringop-overread
(Reported by Alexander Traud)
* ASTERISK-29682 - Squash compiler issues generated by gcc 11
(Reported by George Joseph)
* ASTERISK-29693 - Using --with-crypto and --with-ssl fails on a recompile
(Reported by George Joseph)
* ASTERISK-27816 - func_talkdetect's logic is completely broken
(Reported by Moritz Fain)
* ASTERISK-26497 - make install downloads x86_32 variants of
external modules on non Intel architectures
(Reported by Corey Farrell)
* ASTERISK-29691 - stun: Not all users provide a dst to ast_stun_request
(Reported by Dennis Haney)
Improvements made in this release:
-----------------------------------
* ASTERISK-29777 - documentation: Standardize example syntax
(Reported by N A)
* ASTERISK-29715 - app_voicemail: Refactor email generation functions
(Reported by N A)
* ASTERISK-29727 - Add type for JSON stasis message RTCP Report Received/Sent
(Reported by Boris P. Korzun)
* ASTERISK-29714 - Spelling errors
(Reported by Josh Soref)
* ASTERISK-29707 - chan_iax2: Allow both key and secret to be
specified at dial time
(Reported by N A)
* ASTERISK-29662 - Add mix option to Playback application for
say and filename
(Reported by Shloime Rosenblum)
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
revbump for icu and libffi
asterisk19: Forward-port NetBSD timerfd fix from asterisk18
comms/asterisk19: import asterisk-19.0.0
This is a standard release. As such, it only gets updates
for one year and security fixes for two years.
A variety of items have been deprecated. Some have been
removed in this release, and some are scheduled to be removed in
the next couple of releases. Please take note of those items in
the list below. Out of the box, several of the deprecated items
have been turned off by default (i.e. chan_sip), but I have turned
on everything that was turned on by default in previous versions.
If you are using any of the deprecated items, you should work on
moving away from them as soon as possible.
In ${PREFIX}/share/doc/asterisk, you will find UPGRAGE.txt
and asterisk-19.0.0-summary.txt. Be sure to read them for more
information on updating.
The Asterisk Development Team would like to announce the release
of Asterisk 19.0.0.
The release of Asterisk 19.0.0 resolves several issues reported by
the community and would have not been possible without your
participation.
Thank you!
The following issues are resolved in this release:
Deprecations made in this release:
-----------------------------------
* ASTERISK-29601 - moduleinfo: Add replacement module information
(Reported by N A)
* ASTERISK-29602 - res_monitor: Disable building by default.
(Reported by Joshua C. Colp)
* ASTERISK-29600 - muted: Remove deprecated application
(Reported by Joshua C. Colp)
* ASTERISK-29599 - conf2ael: Remove deprecated application
(Reported by Joshua C. Colp)
* ASTERISK-29598 - res_config_sqlite: Remove deprecated module
(Reported by Joshua C. Colp)
* ASTERISK-29597 - chan_vpb: Remove deprecated module
(Reported by Joshua C. Colp)
* ASTERISK-29596 - chan_misdn: Remove deprecated module
(Reported by Joshua C. Colp)
* ASTERISK-29595 - chan_nbs: Remove deprecated module
(Reported by Joshua C. Colp)
* ASTERISK-29594 - chan_phone: Remove deprecated module
(Reported by Joshua C. Colp)
* ASTERISK-29593 - chan_oss: Remove deprecated module
(Reported by Joshua C. Colp)
* ASTERISK-29592 - cdr_syslog: Remove deprecated module
(Reported by Joshua C. Colp)
* ASTERISK-29591 - app_dahdiras: Remove deprecated module
(Reported by Joshua C. Colp)
* ASTERISK-29590 - app_nbscat: Remove deprecated module
(Reported by Joshua C. Colp)
* ASTERISK-29589 - app_image: Remove deprecated module
(Reported by Joshua C. Colp)
* ASTERISK-29588 - app_url: Remove deprecated module
(Reported by Joshua C. Colp)
* ASTERISK-29587 - app_fax: Remove deprecated module
(Reported by Joshua C. Colp)
* ASTERISK-29586 - app_ices: Remove deprecated module
(Reported by Joshua C. Colp)
* ASTERISK-29585 - app_mysql: Remove deprecated module
(Reported by Joshua C. Colp)
* ASTERISK-29584 - cdr_mysql: Remove deprecated module
(Reported by Joshua C. Colp)
* ASTERISK-29548 - app_meetme: Deprecated in 19, to be removed in 21
(Reported by Joshua C. Colp)
* ASTERISK-29549 - app_osploop: Deprecated in 19, to be removed in 21
(Reported by Joshua C. Colp)
* ASTERISK-29550 - chan_alsa: Deprecated in 19, to be removed in 21
(Reported by Joshua C. Colp)
* ASTERISK-29551 - chan_mgcp: Deprecated in 19, to be removed in 21
(Reported by Joshua C. Colp)
* ASTERISK-29552 - chan_skinny: Deprecated in 19, to be removed in 21
(Reported by Joshua C. Colp)
* ASTERISK-29553 - res_pktccops: Deprecated in 19, to be removed in 21
(Reported by Joshua C. Colp)
* ASTERISK-29558 - app_macro: Deprecated in 16, to be removed in 21
(Reported by Joshua C. Colp)
* ASTERISK-29567 - chan_sip: Deprecated in 17, to be removed in 21
(Reported by Joshua C. Colp)
* ASTERISK-29572 - res_monitor: Deprecated in 16, to be removed in 21
(Reported by Joshua C. Colp)
Security bugs fixed in this release:
-----------------------------------
* ASTERISK-29381 - chan_pjsip: Remote denial of service by an
authenticated user
(Reported by Ivan Poddubny)
* ASTERISK-29415 - Crash in PJSIP TLS transport
(Reported by Andrew Yager)
* ASTERISK-29305 - ASTERISK-29203 / AST-2021-002 -- Another
scenario is causing a crash
(Reported by Gregory Massel)
* ASTERISK-29260 - sRTP Replay Protection ignored; even tears
down long calls
(Reported by Alexander Traud)
* ASTERISK-29227 - res_pjsip_diversion: sending multiple 181
responses causes memory corruption and crash
(Reported by Ivan Poddubny)
* ASTERISK-29219 - res_pjsip_diversion: Crash if Tel URI
contains History-Info
(Reported by Torrey Searle)
* ASTERISK-29057 - pjsip: Crash on call rejection during high load
(Reported by Sandro Gauci)
New Features made in this release:
-----------------------------------
* ASTERISK-29656 - Add CHANNEL_EXISTS function
(Reported by N A)
* ASTERISK-29496 - Add SendMF application
(Reported by N A)
* ASTERISK-29627 - Add STRBETWEEN function
(Reported by N A)
* ASTERISK-29628 - Add file and directory functions
(Reported by N A)
* ASTERISK-29531 - Add SAYFILES function
(Reported by N A)
* ASTERISK-29546 - Add tone detection module
(Reported by N A)
* ASTERISK-18454 - Option for Read to be able to accept #
(Reported by Sta Retji)
* ASTERISK-29542 - Add audio scrambler
(Reported by N A)
* ASTERISK-29478 - Function to drop frames in the TX or RX
directions
(Reported by N A)
* ASTERISK-29389 - Add PJSIP_HEADERS() and ability to read
header by pattern
(Reported by Igor Goncharovsky)
* ASTERISK-11 - AGI channel_status failure
(Reported by bbawkon)
* ASTERISK-29477 - Function to asynchronously store digits dialed
(Reported by N A)
* ASTERISK-29454 - New application to reload modules
(Reported by N A)
* ASTERISK-29444 - Add application to wait for condition
(Reported by N A)
* ASTERISK-29442 - app_dial: Expand A option to allow
announcement playback to caller
(Reported by N A)
* ASTERISK-29446 - app_confbridge: New ConfKick application
(Reported by N A)
* ASTERISK-29440 - app_confbridge: Allow ConfBridge answer to
be suppressed
(Reported by N A)
* ASTERISK-29431 - Minimum and maximum dialplan functions
(Reported by N A)
* ASTERISK-29439 - func_volume: Volume function can't be read
(Reported by N A)
* ASTERISK-27477 - Chan_pjsip does not support unauthenticated
OPTIONS ping
(Reported by Ross Beer)
* ASTERISK-29027 - Implement support for History-Info
(Reported by Torrey Searle)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-20219 - [patch] - IAX2 Call Encryption Fails with
RSA authentication
(Reported by Michael Munger)
* ASTERISK-29402 - res_pjsip_t38: Socket is bound to IPv4/IPv6
but platform does not support it
(Reported by Matthew Kern)
* ASTERISK-29673 - app_read: Fix null pointer crash regression
(Reported by N A)
* ASTERISK-29671 - res_rtp_asterisk: memory leak
(Reported by Jean Aunis - Prescom)
* ASTERISK-29668 - ari: Listing bridges fails when dialing
bridge exists
(Reported by Joshua C. Colp)
* ASTERISK-29663 - messaging: AMI MessageSend does not support
same parameters as dialplan application
(Reported by Brian J. Murrell)
* ASTERISK-29578 - app_queue: Custom device state using
included hints do not update
(Reported by N A)
* ASTERISK-29660 - Build failure when disabling PJSIP support
(Reported by Guido Falsi)
* ASTERISK-29654 - pjproject includes trailing whitespace in
sdp format attributes
(Reported by George Joseph)
* ASTERISK-29635 - MP3Player don' t work with actual mpg123 versions
(Reported by Carlos Oliva)
* ASTERISK-29629 - ARI external media channel creation doesn't
set option data
(Reported by sungtae kim)
* ASTERISK-27176 - test_abstract_jb: frames leak
(Reported by Corey Farrell)
* ASTERISK-29634 - res_snmp: gcc 11 needs -fPIC to compile correctly
(Reported by George Joseph)
* ASTERISK-29630 - Asterisk is unable to read extended number
format terminfo files
(Reported by Sean Bright)
* ASTERISK-28004 - dns: Core ast_dns_get_nameservers does not
support configured IPv6 servers
(Reported by Isaac McDonald)
* ASTERISK-29618 - ConfBridge errors on creation conference room
(Reported by Alexander Zharov)
* ASTERISK-29622 - ARI: external media create doesn't use body parameter
(Reported by sungtae kim)
* ASTERISK-29614 - app_agent_pool: XML Doc: unterminated entity reference
(Reported by Alexander Traud)
* ASTERISK-29609 - Subsequent 'ael reload' will cause a lock up
(Reported by Mark Murawski)
* ASTERISK-28701 - app_queue: Core reload resets queue stats,
even when keepstats=yes
(Reported by Luke Escude)
* ASTERISK-29616 - res_rtp_asterisk: sqrt(.) requires the
header math.h.
(Reported by Alexander Traud)
* ASTERISK-29518 - sig_analog: FCG_CAMA fails to signal ANI
spill when using MF signaling
(Reported by Sarah Autumn)
* ASTERISK-29582 - res_pjproject: Can't map pjproject log
messages to Asterisk TRACE
(Reported by George Joseph)
* ASTERISK-29575 - app_milliwatt: Milliwatt application doesn't
use the proper timings
(Reported by N A)
* ASTERISK-20339 - chan_mgcp, resp_pktccops ast_debug support
(Reported by Tomas Maldonado)
* ASTERISK-29540 - aelparse: include of context with timings fails
(Reported by Alexander Traud)
* ASTERISK-29539 - Segmentation fault at ast_writestream() when
write handler not defined (happens with OGG/Speex)
(Reported by Ernani José Camargo Azevedo)
* ASTERISK-29494 - cdr_adaptive_odbc: Prevent throwing warnings
if CDR filtering is used
(Reported by N A)
* ASTERISK-29513 - statsd: Remove non-standard metric type Meter
(Reported by Rijnhard Hessel)
* ASTERISK-12 - app_voicemail2 became a bit silent, lately
(Reported by siggi)
* ASTERISK-29526 - G729 audio gets corrupted by Asterisk due to smoother
(Reported by under)
* ASTERISK-29392 - chan_iax2: Asterisk crashes when queueing
video with format
(Reported by Michael Welk)
* ASTERISK-27871 - Remote URL in playback must end with file extension
(Reported by Caesar)
* ASTERISK-29507 - STUN timeout is silently delaying calls
(Reported by Sébastien Duthil)
* ASTERISK-29514 - ari: Audiosocket segfault when no data specified
(Reported by Igor Goncharovsky)
* ASTERISK-29503 - Updated identify/match syntax not supported
by config wizard
(Reported by Sean Bright)
* ASTERISK-29480 - fixedjitterbuffer contains an un-wrappered
assert that triggers on a negative time slew
(Reported by Dan Cropp)
* ASTERISK-29485 - core: Inband generation of tones for Busy()
and Congestion() may not occur
(Reported by Joshua C. Colp)
* ASTERISK-29479 - [patch] Channels are not put on hold for
Session Progress with inactive audio
(Reported by Bernd Zobl)
* ASTERISK-29475 - SayNumber triggers WARNING if caller hangs
up during application execution
(Reported by N A)
* ASTERISK-29404 - Consolidate res_pjsip_messaging fixes for
domain name
(Reported by George Joseph)
* ASTERISK-29441 - Core reload making TCP endpoints go offline
(Reported by Luke Escude)
* ASTERISK-28237 - "FRACK!, Failed assertion bad magic number"
happens when unsubscribe an application from an event source
(Reported by Lucas Tardioli Silveira)
* ASTERISK-28393 - Multidomain support issue
(Reported by Andrea Sannucci)
* ASTERISK-29433 - res_rtp_asterisk: Server reflexive
candidates use incorrect raddr for RTCP
(Reported by Chris)
* ASTERISK-29397 - pjsip: Asterisk isn't tolerant of RFC8760 UASs
(Reported by George Joseph)
* ASTERISK-24601 - [patch]Missing RFC4235 tags and attributes
in PJSIP NOTIFY event: dialog XML body
(Reported by Marco Paland)
* ASTERISK-29372 - file.c switch does not account for flash events
(Reported by N A)
* ASTERISK-29370 - chan_sip does not recognize
application/hook-flash
(Reported by N A)
* ASTERISK-29377 - cpool_release_pool "double free or
corruption (out)"
(Reported by Robert Sutton)
* ASTERISK-29358 - chan_pjsip: Trace message for progress is
output even if frame is not queued
(Reported by Michael Maier)
* ASTERISK-29030 - res_rtp_asterisk: Additional RTP-frame (with
wrong SSRC) gets inserted when switching from progress to established
(Reported by Matthias Hensler)
* ASTERISK-29407 - chan_local: Filtering audio formats should
not occur on removed streams
(Reported by Joshua C. Colp)
* ASTERISK-29328 - translate.c: possible buffer overflow when upsampling
(Reported by Jean Aunis - Prescom)
* ASTERISK-29379 - Segfault - ast_channel_is_multistream
(chan=0x0) at channel_internal_api.c:1590
(Reported by Ross Beer)
* ASTERISK-29130 - prometheus: Crash when scraping bridge
(Reported by Francisco Correia)
* ASTERISK-29364 - res_rtp_asterisk: standard deviation
miscalculation
(Reported by Kevin Harwell)
* ASTERISK-29373 - res_rtp_asterisk: Flash events are duplicated
(Reported by N A)
* ASTERISK-28356 - app_queue: CLI set ringinuse for realtime
member not working
(Reported by Michael)
* ASTERISK-24434 - Fix differing usage of assignment operators
in modules.conf
(Reported by Rusty Newton)
* ASTERISK-26614 - app_queue: updatecdr option in queues.conf
does effectively nothing
(Reported by Alexander Gonchiy)
* ASTERISK-24631 - Incorrect description of option "context" in
queues.conf.sample
(Reported by Etienne Lessard)
* ASTERISK-25358 - dateformat not read from logger.conf by
remote console
(Reported by Igor Liferenko)
* ASTERISK-27542 - app_queue: When "queue show" CLI command is
executed a crash occurs
(Reported by Miguel Sanz)
* ASTERISK-29215 - res_pjsip_session: NULL active_media_state
topology caused asterisk crash
(Reported by sungtae kim)
* ASTERISK-29355 - app_queue: Queue member status message sent
even if status doesn't change
(Reported by Roman Pertsev)
* ASTERISK-29035 - chan_local: Multistream support breaks T.38 faxing
(Reported by Matthias Hensler)
* ASTERISK-29354 - res_pjsip: Allow partial reloading of transports
(Reported by Joshua C. Colp)
* ASTERISK-29348 - menuselect doesn't return errors in many cases
(Reported by George Joseph)
* ASTERISK-29352 - res_rtp_asterisk: Fix frame delivery time
when SSRC changes
(Reported by Joshua C. Colp)
* ASTERISK-29071 - app_confbridge: Memory rises when
jitterbuffer enabled and muting over AMI occurs
(Reported by Stefan Ruf)
* ASTERISK-29329 - app_dial: DTMF to 'D' option gets duplicated
if there are multiple progress events
(Reported by N A)
* ASTERISK-29306 - strings: Incorrect use of
__attribute__((pure)) in ast_str_to_lower definition
(Reported by Vitezslav Novy)
* ASTERISK-29300 - res_rtp_asterisk: When native local bridging
the remote SSRC becomes permanent
(Reported by Sebastian Damm)
* ASTERISK-29235 - res_pjsip_nat: Contact is rewritten on
REGISTER responses with external_signaling_address
(Reported by Brian Paboojian)
* ASTERISK-29266 - ICE Role conflict with an unauthorized session
(Reported by Salah Ahmed)
* ASTERISK-29105 - chan_pjsip: 180 Ringing with SDP not changed
into progress
(Reported by Sebastian Damm)
* ASTERISK-29297 - say: Y2021 problem â<80><93> Asterisk cannot say
year 2021 in Dutch
(Reported by Jacek Konieczny)
* ASTERISK-29315 - res_pjsip: re-registration gets stuck if
setting initial auth credentials fails
(Reported by Nick French)
* ASTERISK-29312 - res_fax: asterisk fails to publish the
Stasis and ReceiveFax status messages if the remote Station ID
contains invalid UTF-8 characters
(Reported by Alexei Gradinari)
* ASTERISK-16799 - Callee declined when 'beep' audio file does
not exist
(Reported by IAMJames_)
* ASTERISK-29313 - res_pjsip_refer: Segfault in progress notify
(Reported by George Joseph)
* ASTERISK-28452 - pjsip: <sess-version> of SDP is not
incremented though SDP may be changed on reinvite without SDP offer
(Reported by Michael Maier)
* ASTERISK-29311 - res_odbc_transaction sets forcecommit
default value based on isolation level instead of forcecommit
(Reported by Jaco Kroon)
* ASTERISK-29303 - pjsip: Re-invite occurs when it shouldn't
(Reported by Benjamin Keith Ford)
* ASTERISK-29293 - res_config_pgsql: Limit realtime_pgsql() to
return one (no more) record
(Reported by Boris P. Korzun)
* ASTERISK-28369 - app_queue: Member device state "invalid"
when second call is ringing and hint is used
(Reported by Boolah)
* ASTERISK-29287 - app.h: C++ compatibility broken
(Reported by Jean Aunis - Prescom)
* ASTERISK-29203 - res_pjsip_t38: Crash when changing state
(Reported by Gregory Massel)
* ASTERISK-29205 - res_rtp_asterisk: Asterisk crashes when
making hold/unhold from webrtc client
(Reported by Edvin Vidmar)
* ASTERISK-29196 - res_pjsip: Segmentation fault
(Reported by Mauri de Souza Meneguzzo (3CPlus))
* ASTERISK-29280 - chan_sip: Allow peers without audio (text+video).
(Reported by Alexander Traud)
* ASTERISK-29265 - chan_sip: Allow text+video media streams, again.
(Reported by Alexander Traud)
* ASTERISK-29259 - channel: Allow text+video media streams, again.
(Reported by Alexander Traud)
* ASTERISK-29261 - res_pjsip: user=phone validation fail for
isup numbers containing *#
(Reported by Mark Petersen)
* ASTERISK-29258 - chan_sip: Audio stream rejected, Other
stream present: Invalid SDP.
(Reported by Alexander Traud)
* ASTERISK-29248 - res_pjsip_session: res sometimes
uninitialized reported by compiler Clang.
(Reported by Alexander Traud)
* ASTERISK-29220 - After T38 reinvite response of 488 a
subsequent G711 reinvite is not processed correctly. Instead the
previous T38 session media is used
(Reported by Robert Cripps)
* ASTERISK-29229 - Stasis/messaging: text messages not
dispatched to all subscribers when using generic subscription
(Reported by Jean Aunis - Prescom)
* ASTERISK-29238 - chan_sip: SDP: Offers without any enabled
stream are accepted.
(Reported by Alexander Traud)
* ASTERISK-29237 - chan_sip: SDP: m=video is parsed even when disabled.
(Reported by Alexander Traud)
* ASTERISK-29222 - chan_sip: Hold/Resume an sRTP call on a
video enabled user-agent.
(Reported by Alexander Traud)
* ASTERISK-29240 - chan_pjsip: Incoming PJSIP calls set global
SIPDOMAIN instead of a channel variable
(Reported by Ivan Poddubny)
* ASTERISK-27902 - chan_pjsip isn't updating hangupcause on 4XX responses
(Reported by George Joseph)
* ASTERISK-28016 - PJSIP sends duplicate 183 Progress responses
(Reported by Alex Hermann)
* ASTERISK-28185 - chan_pjsip: Subsequent same responses are not stopped
(Reported by Julien)
* ASTERISK-29230 - pjsip: Asterisk goes crazy and massively
spams logfile if registration can't be send
(Reported by Michael Maier)
* ASTERISK-29231 - pjsip: SIGSEGV in CLI if no trunk is registered
(Reported by Michael Maier)
* ASTERISK-29217 - LOCK() can grant the same lock to multiple
channels spuriously
(Reported by Jaco Kroon)
* ASTERISK-28947 - Segmentation fault in mixmonitor_ds_destroy
(Reported by Robert Sutton)
* ASTERISK-29201 - Crash occurs when Transfer and execute
Hangup before the Transfer result
(Reported by Dan Cropp)
* ASTERISK-29168 - Asterisk crashes during call transfer
(Reported by Dalius Mockevicius)
* ASTERISK-29210 - res_pjsip: Crash when examining transport
(Reported by N GM )
* ASTERISK-29191 - tel: URI in Diversion header causes crash
(Reported by Mikhail Ivanov)
* ASTERISK-28883 - Spyee information ist missing in ChanSpyStop
AMI Event
(Reported by Hendrik Wedhorn)
* ASTERISK-29188 - null media causing the Asterisk crash
(Reported by sungtae kim)
* ASTERISK-29209 - Debug messages printed by scope trace might
be missing newlines
(Reported by Alexander Traud)
* ASTERISK-29024 - pjsip: Route Header in Cancel request
incorrectly set
(Reported by Flole Systems)
* ASTERISK-29211 - res_musiconhold: Segfault on realtime music
on hold without entries
(Reported by Nathan Bruning)
* ASTERISK-29022 - Crash when manipulating PJSIP invite dlg ref counts
(Reported by Sean Bright)
* ASTERISK-29173 - Media cache URL requests allow infinite redirects
(Reported by Sean Bright)
* ASTERISK-29175 - res_pjsip_stir_shaken: Fix module description
(Reported by Stanislav Abramenkov)
* ASTERISK-29148 - AST_MODULE_INFO no, MODULEINFO depend
(Reported by Alexander Traud)
* ASTERISK-29165 - res_pjsip: malformed header Accept-Encoding
in OPTIONS response
(Reported by Alexander Greiner-Baer)
* ASTERISK-28798 - [patch] chan_sip: TCP/TLS client without server.
(Reported by Alexander Traud)
* ASTERISK-29161 - Incorrect setup of recall channels
(Reported by Boris P. Korzun)
* ASTERISK-29155 - app_queue: Deadlock between queues container
and individual queues
(Reported by George Joseph)
* ASTERISK-28933 - res_pjsip.so fails to load when bundled
pjproject is compiled without libssl
(Reported by Walter Doekes)
* ASTERISK-28825 - Any curl response checks out as valid even
if 404 is returned.
(Reported by dovid)
* ASTERISK-29013 - res_pjsip: Asterisk doesn't stop sending
invites (with auth) on 407 replies
(Reported by Sebastian Damm)
* ASTERISK-29142 - sip_to_pjsip.py: doesn't read globbed includes
(Reported by Michael Newton)
* ASTERISK-29144 - GCC Warnings with OPTIMIZE=-Og make
(Reported by Alexander Traud)
* ASTERISK-29146 - GCC Warnings: %s> directive argument is null.
(Reported by Alexander Traud)
* ASTERISK-29145 - GCC Warnings with OPTIMIZE=-Os make
(Reported by Alexander Traud)
* ASTERISK-29124 - res_pjsip: flow transport broken for
outbound requests
(Reported by Nick French)
* ASTERISK-29136 - config: Sample features.conf incorrectly
includes " around sound files
(Reported by Benjamin M.)
* ASTERISK-29123 - logger.conf.sample missing comment mark on line 115
(Reported by Andrew Siplas)
* ASTERISK-29109 - res_pjsip_session: Asterisk 18 does not
progress calls due to codec negotiation after upgrading from
Asterisk 16
(Reported by Ross Beer)
* ASTERISK-28430 - res_rtp_asterisk.c: FRACK!, Failed assertion
errno != EBADF
(Reported by under)
* ASTERISK-29108 - resource_endpoints.c : Memory leak if
endpoint not found
(Reported by Jean Aunis - Prescom)
* ASTERISK-29097 - res_pjsip_config_wizard: Crash when freeing
string when failing to add extension
(Reported by Vieri)
* ASTERISK-26424 - app_voicemail: Undocumented behavior from VMSayName
(Reported by Eric Smith)
* ASTERISK-29051 - res_pjsip_sdp_rtp: Does not set correct
values on RTP instance when "auto" DTMF is used
(Reported by Sebastian Damm)
* ASTERISK-29099 - res_musiconhold: Realtime MOH only loads a
single entry
(Reported by laszlovl)
* ASTERISK-29091 - Crash when ast_translator_build_path fails
(Reported by Jasper van der Neut)
* ASTERISK-28311 - dsp: ast_dsp_silence_noise_with_energy wrong
judgment of frame format
(Reported by å<91>¨å®¶å»º)
* ASTERISK-29085 - func_curl: Segmentation fault when using
CURL after setting httpheader CURLOPT
(Reported by Péter
Juhász)
* ASTERISK-24329 - Music On Hold announcement cuts intro of
music the first time it is played
(Reported by Thomas Frederiksen)
* ASTERISK-29089 - RTP Ports not cleared after hangup
(Reported by Ross Beer)
* ASTERISK-29081 - res_stasis: Add compare function for bridges
moh container
(Reported by Hajek Michal)
* ASTERISK-28416 - Unable to get rtp codec payload code for slin
(Reported by Brian J. Murrell)
* ASTERISK-29014 - res_pjsip_session: Re-INVITE collisions
aren't handled correctly
(Reported by George Joseph)
* ASTERISK-25665 - Duplicate logging in queue log for EXITEMPTY events
(Reported by Ove Aursand)
* ASTERISK-29043 - app_queue: Leave empty sometimes not
recorded as abandoned
(Reported by Kfir Itzhak)
* ASTERISK-29042 - res_parking: Parker UUID is no longer copied
(Reported by Misha Vodsedalek)
* ASTERISK-28878 - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16
(Reported by Joseph Ades)
* ASTERISK-29046 - pbx: Deadlock when doing a reload, while
simultaneously doing an ExtensionState on a pattern match hint
that ends up adding an extension
(Reported by Ramarajan)
* ASTERISK-29040 - res_speech: Assertion on format
(Reported by Nickolay V. Shmyrev)
* ASTERISK-29001 - chan_pjsip does not process or forward 181 responses
(Reported by Torrey Searle)
* ASTERISK-27273 - app_voicemail: When a voicemail is marked as
"Urgent", it is not sent by email/processed by the mailcmd command
(Reported by Leandro Dardini)
* ASTERISK-29034 - Lastpause of realtime members is reseting
(Reported by Evandro César Arruda)
* ASTERISK-29033 - res_pjsip_session: Aggressively terminates
session on failed re-INVITE
(Reported by Joshua C. Colp)
* ASTERISK-28974 - res_rtp_asterisk: T.140 messages have
appended RTP string to each message block. (Reported by
Thomas Johnson)
* ASTERISK-29011 - chan_sip: ToHost property not cleared on reload
(Reported by Dennis)
* ASTERISK-29021 - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on
certified versions
(Reported by cmaj)
* ASTERISK-28927 - Asterisk crash in music on hold
(Reported by David Cunningham)
* ASTERISK-28973 - Malformed IP address in SDP of 2nd SIP timer
triggered INVITE when NAT is active (UDP transport with
external_media_address)
(Reported by Michael Neuhauser)
* ASTERISK-28995 - res_pjsip_registrar: Expires on statically
configured contacts is not correct
(Reported by tootai)
* ASTERISK-28987 - BridgeCreated ARI event shows wrong video_mode info
(Reported by sungtae kim)
* ASTERISK-28978 - acl: named_acl rule misconfiguration results
in segfault on reading rule from realtime
(Reported by Andrew Yager)
Improvements made in this release:
-----------------------------------
* ASTERISK-29637 - Add support for future dates in Say.c
(Reported by Shloime Rosenblum)
* ASTERISK-29525 - PJSIP remove_existing unavailable contacts
(Reported by Joseph Nadiv)
* ASTERISK-29661 - func_vmcount: Add support for multiple mailboxes
(Reported by N A)
* ASTERISK-29275 - Support of MIME-type for wav16
(Reported by Boris P. Korzun)
* ASTERISK-29529 - Add custom logging level
(Reported by N A)
* ASTERISK-29472 - res_pjsip: OLI/ANI2 support missing
(Reported by N A)
* ASTERISK-29626 - app_stack: Include calling location if
attempting to branch to nonexistent location
(Reported by N A)
* ASTERISK-29632 - Add option to Application_VoiceMail to
suppress instructions only when a custom greeting is present
(Reported by Charlie Smurthwaite)
* ASTERISK-29605 - chan_iax2: Add ANI2
(Reported by N A)
* ASTERISK-29508 - STUN server address refresh
(Reported by Sébastien Duthil)
* ASTERISK-29612 - bridge_basic: Don't throw warning if
attended transfer is cancelled
(Reported by N A)
* ASTERISK-29544 - Media Cache - Delayed remote sound file
retrieve delays all playbacks
(Reported by Andre Barbosa)
* ASTERISK-29495 - Return integer instead of float if response
is a whole number
(Reported by N A)
* ASTERISK-29541 - app_morsecode: Add American Morse code
(Reported by N A)
* ASTERISK-29543 - app_originate: Allow specifying codec(s) to use
(Reported by N A)
* ASTERISK-29528 - Add support for multiple files for agent announcements
(Reported by N A)
* ASTERISK-29527 - res_http_media_cache: Cleanup audio format
lookup in HTTP requests
(Reported by Sean Bright)
* ASTERISK-29501 - ARI - Stasis Playback doesn't hangup call
when processing a list of invalid files
(Reported by Andre Barbosa)
* ASTERISK-29464 - ARI - PlaybackFinish skip error events
(Reported by Andre Barbosa)
* ASTERISK-29450 - Allow setting channel variables using
Originate application
(Reported by N A)
* ASTERISK-29460 - Recognize application/hook-flash in PJSIP
(Reported by N A)
* ASTERISK-29459 - Missing configuration from PJSIP to SIP
conversion script
(Reported by N A)
* ASTERISK-29434 - Asterisk reveals pjproject version in STUN packets
(Reported by Jeremy Lainé)
* ASTERISK-29380 - Add Flash AMI event to handle flash events
(Reported by N A)
* ASTERISK-29349 - Silent voicemail option is not completely silent
(Reported by N A)
* ASTERISK-29339 - loader: Let's output warnings for deprecated modules!
(Reported by Joshua C. Colp)
* ASTERISK-29337 - menuselect: Add ability to set deprecated in
and removed in versions for modules
(Reported by Joshua C. Colp)
* ASTERISK-29335 - xml: Embed module information into core XML documentation.
(Reported by Joshua C. Colp)
* ASTERISK-29336 - documentation: Fix inconsistent support levels
(Reported by Joshua C. Colp)
* ASTERISK-29321 - sorcery: Add support for more intelligent reloading.
(Reported by Joshua C. Colp)
* ASTERISK-29325 - res_pjsip_registrar: Include source IP
address and port in log messages
(Reported by Joshua C. Colp)
* ASTERISK-29326 - asterisk: Update copyright/company
(Reported by Joshua C. Colp)
* ASTERISK-29244 - Add MixMonitorStart / Stop / Mute AMI events
(Reported by Sébastien Duthil)
* ASTERISK-29252 - TRANSFERSTATUSPROTOCOL variable to report
Transfer (REFER) failure SIP code
(Reported by Dan Cropp)
* ASTERISK-29262 - Support of various URL-schemes by MoH
(Reported by Boris P. Korzun)
* ASTERISK-28549 - Two repeated 183
(Reported by Gant Liu)
* ASTERISK-29216 - contrib: systemd asterisk service for
centos8 or other newer linux versions
(Reported by Mark Petersen)
* ASTERISK-29143 - res_http_media_cache: HTTP media cache
stored hardcoded in /tmp
(Reported by laszlovl)
* ASTERISK-29118 - VoiceMail() should have an option to play
greetings as Early Media
(Reported by Juan Carlos Castro y Castro)
* ASTERISK-29054 - Logger: Add debug logging categories
(Reported by Kevin Harwell)
* ASTERISK-29083 - Do not build chan_sip by default as it is
now deprecated
(Reported by Sean Bright)
* ASTERISK-29056 - Increase reg_server column size for
ps_contacts table realtime
(Reported by sungtae kim)
* ASTERISK-29055 - Create a Bridge with video_single mode
(Reported by sungtae kim)
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
Thank you for your continued support of Asterisk!