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asterisk/asterisk-18.3.0-summary.txt
Asterisk Development Team cd5f686bc3 Update for 18.3.0
2021-03-25 12:34:21 -05:00

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Release Summary
asterisk-18.3.0
Date: 2021-03-25
<asteriskteam@digium.com>
----------------------------------------------------------------------
Table of Contents
1. Summary
2. Contributors
3. Closed Issues
4. Other Changes
5. Diffstat
----------------------------------------------------------------------
Summary
[Back to Top]
This release is a point release of an existing major version. The changes
included were made to address problems that have been identified in this
release series, or are minor, backwards compatible new features or
improvements. Users should be able to safely upgrade to this version if
this release series is already in use. Users considering upgrading from a
previous version are strongly encouraged to review the UPGRADE.txt
document as well as the CHANGES document for information about upgrading
to this release series.
The data in this summary reflects changes that have been made since the
previous release, asterisk-18.2.0.
----------------------------------------------------------------------
Contributors
[Back to Top]
This table lists the people who have submitted code, those that have
tested patches, as well as those that reported issues on the issue tracker
that were resolved in this release. For coders, the number is how many of
their patches (of any size) were committed into this release. For testers,
the number is the number of times their name was listed as assisting with
testing a patch. Finally, for reporters, the number is the number of
issues that they reported that were affected by commits that went into
this release.
Coders Testers Reporters
11 Alexander Traud 1 Mark Petersen 6 Alexander Traud
9 Joshua C. Colp 3 Boris P. Korzun
8 Sean Bright 3 Joshua C. Colp
6 Jaco Kroon 2 Matthias Hensler
5 George Joseph 2 Stefan Ruf
4 Ben Ford 2 Sebastian Damm
3 Kevin Harwell 2 Gregory Massel
3 Asterisk Development Team 1 Rusty Newton
3 Ivan Poddubnyi 1 Alexei Gradinari
3 Boris P. Korzun 1 Ivan Poddubny
1 Salah Ahmed 1 Jacek Konieczny
1 Dan Cropp 1 Jaco Kroon
1 Holger Hans Peter Freyther 1 Edvin Vidmar
1 Nico Kooijman 1 Sébastien Duthil
1 Alexei Gradinari 1 Jean Aunis - Prescom
1 Torrey Searle 1 sungtae kim
1 Nick French 1 Benjamin Keith Ford
1 Robert Cripps 1 Boolah
1 Sebastien Duthil 1 Nick French
1 Mark Petersen 1 Salah Ahmed
1 Mauri de Souza Meneguzzo
(3CPlus)
1 N A
1 N A
1 Jacek Konieczny
1 IAMJames_
1 Mark Petersen
1 Dan Cropp
1 Ivan Poddubny
1 Vitezslav Novy
1 Mark Petersen
1 Michael Maier
1 George Joseph
1 Alexander Traud
1 Brian Paboojian
1 Dan Cropp
1 Robert Cripps
----------------------------------------------------------------------
Closed Issues
[Back to Top]
This is a list of all issues from the issue tracker that were closed by
changes that went into this release.
Security
Category: Resources/res_pjsip_t38
ASTERISK-29305: ASTERISK-29203 / AST-2021-002 -- Another scenario is
causing a crash
Reported by: Gregory Massel
* [77328142b4] Ben Ford -- AST-2021-006 - res_pjsip_t38.c: Check for
session_media on reinvite.
Category: Resources/res_srtp
ASTERISK-29260: sRTP Replay Protection ignored; even tears down long calls
Reported by: Alexander Traud
* [703158b903] Alexander Traud -- rtp: Enable srtp replay protection
Category: pjproject/pjsip
ASTERISK-29227: res_pjsip_diversion: sending multiple 181 responses causes
memory corruption and crash
Reported by: Ivan Poddubny
* [2770cc5872] Ivan Poddubnyi -- res_pjsip_diversion: Fix adding more
than one histinfo to Supported
Bug
Category: Applications/General
ASTERISK-29287: app.h: C++ compatibility broken
Reported by: Jean Aunis - Prescom
* [916d5d5e45] Jaco Kroon -- app.h: Restore C++ compatibility for macro
AST_DECLARE_APP_ARGS
Category: Applications/app_confbridge
ASTERISK-29071: app_confbridge: Memory rises when jitterbuffer enabled and
muting over AMI occurs
Reported by: Stefan Ruf
* [f7bda066bb] Joshua C. Colp -- channel: Fix crash in suppress API.
* [b43b81d953] Joshua C. Colp -- channel: Fix memory leak in suppress
API.
Category: Applications/app_dial
ASTERISK-29329: app_dial: DTMF to 'D' option gets duplicated if there are
multiple progress events
Reported by: N A
* [94debe5085] Sean Bright -- app_dial.c: Only send DTMF on first
progress event.
Category: Applications/app_page
ASTERISK-16799: Callee declined when 'beep' audio file does not exist
Reported by: IAMJames_
* [6673c1b177] Sean Bright -- app_page.c: Don't fail to Page if beep
sound file is missing
Category: Applications/app_queue
ASTERISK-28369: app_queue: Member device state "invalid" when second call
is ringing and hint is used
Reported by: Boolah
* [985d3e4940] Ivan Poddubnyi -- app_queue: Fix conversion of complex
extension states into device states
Category: Channels/chan_local
ASTERISK-29035: chan_local: Multistream support breaks T.38 faxing
Reported by: Matthias Hensler
* [47e9ce96ea] Joshua C. Colp -- core_unreal: Fix deadlock with T.38
control frames.
* [62e2dd484d] Ben Ford -- core_unreal: Fix T.38 faxing when using local
channels.
Category: Channels/chan_sip/CodecHandling
ASTERISK-29280: chan_sip: Allow peers without audio (text+video).
Reported by: Alexander Traud
* [45e48e387c] Alexander Traud -- chan_sip: Allow [peer] without audio
(text+video).
ASTERISK-29265: chan_sip: Allow text+video media streams, again.
Reported by: Alexander Traud
* [87ad1138ff] Alexander Traud -- chan_sip: Set up calls without audio
(text+video), again.
ASTERISK-29258: chan_sip: Audio stream rejected, Other stream present:
Invalid SDP.
Reported by: Alexander Traud
* [4c154f3431] Alexander Traud -- chan_sip: SDP: Reject audio streams
correctly.
Category: Core/Bridging
ASTERISK-29071: app_confbridge: Memory rises when jitterbuffer enabled and
muting over AMI occurs
Reported by: Stefan Ruf
* [f7bda066bb] Joshua C. Colp -- channel: Fix crash in suppress API.
* [b43b81d953] Joshua C. Colp -- channel: Fix memory leak in suppress
API.
Category: Core/Channels
ASTERISK-29259: channel: Allow text+video media streams, again.
Reported by: Alexander Traud
* [f64ddf3db3] Alexander Traud -- channel: Set up calls without audio
(text+video), again.
Category: Core/General
ASTERISK-29306: strings: Incorrect use of __attribute__((pure)) in
ast_str_to_lower definition
Reported by: Vitezslav Novy
* [e4cd7a7d0b] Sean Bright -- strings.h: ast_str_to_upper() and
_to_lower() are not pure.
Category: Core/Internationalization
ASTERISK-29297: say: Y2021 problem – Asterisk cannot say year 2021 in
Dutch
Reported by: Jacek Konieczny
* [7b052ec965] Nico Kooijman -- main: With Dutch language year after
2020 is not spoken in say.c
Category: Documentation
ASTERISK-24434: Fix differing usage of assignment operators in
modules.conf
Reported by: Rusty Newton
* [3084084648] Sean Bright -- modules.conf: Fix differing usage of
assignment operators.
Category: Resources/res_config_pgsql
ASTERISK-29293: res_config_pgsql: Limit realtime_pgsql() to return one (no
more) record
Reported by: Boris P. Korzun
* [beb579bc99] Boris P. Korzun -- res_config_pgsql: Limit
realtime_pgsql() to return one (no more) record.
Category: Resources/res_fax
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
* [d5e73d2121] Alexei Gradinari -- res_fax: validate the remote/local
Station ID for UTF-8 format
Category: Resources/res_odbc
ASTERISK-29311: res_odbc_transaction sets forcecommit default value based
on isolation level instead of forcecommit
Reported by: Jaco Kroon
* [7ab53fce7a] Jaco Kroon -- res_odbc_transaction: correctly initialise
forcecommit value from DSN.
Category: Resources/res_pjsip
ASTERISK-29196: res_pjsip: Segmentation fault
Reported by: Mauri de Souza Meneguzzo (3CPlus)
* [acb7ce4fe7] Joshua C. Colp -- pjsip: Make modify_local_offer2
tolerate previous failed SDP.
ASTERISK-29261: res_pjsip: user=phone validation fail for isup numbers
containing *#
Reported by: Mark Petersen
* [176274caa4] Mark Petersen -- res/res_pjsip.c: allow user=phone when
number contain *#
Category: Resources/res_pjsip_nat
ASTERISK-29235: res_pjsip_nat: Contact is rewritten on REGISTER responses
with external_signaling_address
Reported by: Brian Paboojian
* [976b1a1d7a] Joshua C. Colp -- res_pjsip_nat: Don't rewrite Contact on
REGISTER responses.
Category: Resources/res_pjsip_outbound_registration
ASTERISK-29315: res_pjsip: re-registration gets stuck if setting initial
auth credentials fails
Reported by: Nick French
* [dedfb334bd] Nick French -- res_pjsip: dont return early from
registration if init auth fails
Category: Resources/res_pjsip_refer
ASTERISK-29313: res_pjsip_refer: Segfault in progress notify
Reported by: George Joseph
* [15afabdf8e] George Joseph -- res_pjsip_refer: Refactor progress
locking and serialization
Category: Resources/res_pjsip_registrar
ASTERISK-29235: res_pjsip_nat: Contact is rewritten on REGISTER responses
with external_signaling_address
Reported by: Brian Paboojian
* [976b1a1d7a] Joshua C. Colp -- res_pjsip_nat: Don't rewrite Contact on
REGISTER responses.
Category: Resources/res_pjsip_sdp_rtp
ASTERISK-29105: chan_pjsip: 180 Ringing with SDP not changed into progress
Reported by: Sebastian Damm
* [3286c04856] Holger Hans Peter Freyther -- pjsip: Generate progress
(once) when receiving a 180 with a SDP
ASTERISK-28452: pjsip: of SDP is not incremented though SDP may be changed
on reinvite without SDP offer
Reported by: Michael Maier
* [1af2a84c8b] Joshua C. Colp -- res_pjsip_session: Always produce offer
on re-INVITE without SDP.
Category: Resources/res_pjsip_session
ASTERISK-29215: res_pjsip_session: NULL active_media_state topology caused
asterisk crash
Reported by: sungtae kim
* [bffff6e2d0] George Joseph -- res_pjsip_session: Make
reschedule_reinvite check for NULL topologies
ASTERISK-29303: pjsip: Re-invite occurs when it shouldn't
Reported by: Benjamin Keith Ford
* [83b0f5963f] Ben Ford -- res_pjsip_session.c: Check topology on
re-invite.
ASTERISK-29203: res_pjsip_t38: Crash when changing state
Reported by: Gregory Massel
* [fad0cf12e6] Kevin Harwell -- AST-2021-002: Remote crash possible when
negotiating T.38
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
* [017e09b40a] Robert Cripps -- res/res_pjsip_session.c: Check that
media type matches in
ASTERISK-29248: res_pjsip_session: res sometimes uninitialized reported by
compiler Clang.
Reported by: Alexander Traud
* [3f119192bb] Alexander Traud -- res_pjsip_session: Avoid
sometimes-uninitialized warning with Clang.
Category: Resources/res_pjsip_t38
ASTERISK-29203: res_pjsip_t38: Crash when changing state
Reported by: Gregory Massel
* [fad0cf12e6] Kevin Harwell -- AST-2021-002: Remote crash possible when
negotiating T.38
Category: Resources/res_rtp_asterisk
ASTERISK-29300: res_rtp_asterisk: When native local bridging the remote
SSRC becomes permanent
Reported by: Sebastian Damm
* [90ef6a14a7] Torrey Searle -- res/res_rtp_asterisk: generate new SSRC
on native bridge end
ASTERISK-29266: ICE Role conflict with an unauthorized session
Reported by: Salah Ahmed
* [df8d335ad1] Salah Ahmed -- res_rtp_asterisk: Check remote ICE reset
and reset local ice attrb
ASTERISK-29205: res_rtp_asterisk: Asterisk crashes when making hold/unhold
from webrtc client
Reported by: Edvin Vidmar
* [5a6f2f913b] Sean Bright -- res_rtp_asterisk.c: Fix signed mismatch
that leads to overflow
Category: pjproject/pjsip
ASTERISK-28452: pjsip: of SDP is not incremented though SDP may be changed
on reinvite without SDP offer
Reported by: Michael Maier
* [1af2a84c8b] Joshua C. Colp -- res_pjsip_session: Always produce offer
on re-INVITE without SDP.
Improvement
Category: Applications/app_mixmonitor
ASTERISK-29244: Add MixMonitorStart / Stop / Mute AMI events
Reported by: Sébastien Duthil
* [092628c982] Sebastien Duthil -- app_mixmonitor: Add AMI events
MixMonitorStart, -Stop and -Mute.
Category: Applications/app_transfer
ASTERISK-29252: TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
failure SIP code
Reported by: Dan Cropp
* [088816284a] Dan Cropp -- chan_pjsip, app_transfer: Add
TRANSFERSTATUSPROTOCOL variable
Category: Channels/chan_pjsip
ASTERISK-29252: TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
failure SIP code
Reported by: Dan Cropp
* [088816284a] Dan Cropp -- chan_pjsip, app_transfer: Add
TRANSFERSTATUSPROTOCOL variable
Category: Core/General
ASTERISK-29326: asterisk: Update copyright/company
Reported by: Joshua C. Colp
* [682f7d9437] Joshua C. Colp -- asterisk: Update copyright.
Category: Core/Sorcery
ASTERISK-29321: sorcery: Add support for more intelligent reloading.
Reported by: Joshua C. Colp
* [a9acbd19f3] Joshua C. Colp -- sorcery: Add support for more
intelligent reloading.
Category: Formats/format_wav
ASTERISK-29275: Support of MIME-type for wav16
Reported by: Boris P. Korzun
* [57d130d3aa] Boris P. Korzun -- format_wav: Support of MIME-type for
wav16
Category: Resources/res_musiconhold
ASTERISK-29262: Support of various URL-schemes by MoH
Reported by: Boris P. Korzun
* [f1c88a497b] Boris P. Korzun -- res_musiconhold: Add support of
various URL-schemes by MoH.
Category: Resources/res_pjsip_registrar
ASTERISK-29325: res_pjsip_registrar: Include source IP address and port in
log messages
Reported by: Joshua C. Colp
* [5f1c21e4ca] Joshua C. Colp -- res_pjsip_registrar: Include source IP
and port in log messages.
----------------------------------------------------------------------
Commits Not Associated with an Issue
[Back to Top]
This is a list of all changes that went into this release that did not
reference a JIRA issue.
+------------------------------------------------------------------------+
| Revision | Author | Summary |
|------------+------------------+----------------------------------------|
| 2c0e6bac06 | Asterisk | Update for 18.3.0-rc2 |
| | Development Team | |
|------------+------------------+----------------------------------------|
| ae4a3da557 | Asterisk | Update for 18.3.0-rc1 |
| | Development Team | |
|------------+------------------+----------------------------------------|
| 263f906af4 | Kevin Harwell | manager: Increase the non breaking AMI |
| | | version number |
|------------+------------------+----------------------------------------|
| 0afd37e3b5 | Asterisk | Update CHANGES and UPGRADE.txt for |
| | Development Team | 18.3.0 |
|------------+------------------+----------------------------------------|
| | | func_callerid+res_agi: Fix compile |
| 23e41313a8 | Jaco Kroon | errors related to |
| | | -Werror=zero-length-bounds |
|------------+------------------+----------------------------------------|
| 52707fba7f | Jaco Kroon | app.h: Fix -Werror=zero-length-bounds |
| | | compile errors in dev mode. |
|------------+------------------+----------------------------------------|
| 262473c6d9 | Alexander Traud | res_format_attr_*: Parameter Names are |
| | | Case-Insensitive. |
|------------+------------------+----------------------------------------|
| 4fc0e16838 | Alexander Traud | chan_iax2: System Header strings is |
| | | included via asterisk.h/compat.h. |
|------------+------------------+----------------------------------------|
| 16e4d1f36f | Sean Bright | res_musiconhold.c: Plug ref leak |
| | | caused by ao2_replace() misuse. |
|------------+------------------+----------------------------------------|
| 269bb08ea2 | George Joseph | res_pjsip_refer: Move the progress dlg |
| | | release to a serializer |
|------------+------------------+----------------------------------------|
| 0323293142 | Alexander Traud | res_format_attr_h263: Generate valid |
| | | SDP fmtp for H.263+. |
|------------+------------------+----------------------------------------|
| be0a61bc3d | Kevin Harwell | res_rtp_asterisk: Add packet subtype |
| | | during RTCP debug when relevant |
|------------+------------------+----------------------------------------|
| 1adf9368ee | Alexander Traud | chan_sip: Filter pass-through |
| | | audio/video formats away, again. |
|------------+------------------+----------------------------------------|
| bee35fe04a | Jaco Kroon | func_odbc: Introduce minargs config |
| | | and expose ARGC in addition to ARGn. |
|------------+------------------+----------------------------------------|
| dbd8908f8d | George Joseph | res_pjsip_refer: Always serialize |
| | | calls to refer_progress_notify |
|------------+------------------+----------------------------------------|
| 28f187d6c5 | George Joseph | chan_iax2.c: Require secret and auth |
| | | method if encryption is enabled |
|------------+------------------+----------------------------------------|
| 24d6adfe99 | Sean Bright | app_read: Release tone zone reference |
| | | on early return. |
|------------+------------------+----------------------------------------|
| 7c0fbaf010 | Ivan Poddubnyi | main/frame: Add missing control frame |
| | | names to ast_frame_subclass2str |
|------------+------------------+----------------------------------------|
| fb42b60326 | Sean Bright | res_pjsip_pubsub: Fix truncation of |
| | | persisted SUBSCRIBE packet |
|------------+------------------+----------------------------------------|
| 9c56870929 | Jaco Kroon | AC_HEADER_STDC causes a compile |
| | | failure with autoconf 2.70 |
|------------+------------------+----------------------------------------|
| | | pjsip_scheduler: Fix pjsip show |
| a25bcf70ed | Alexander Traud | scheduled_tasks like for compiler |
| | | Clang. |
|------------+------------------+----------------------------------------|
| 87a35f8e94 | Ben Ford | chan_pjsip.c: Add parameters to frame |
| | | in indicate. |
+------------------------------------------------------------------------+
----------------------------------------------------------------------
Diffstat Results
[Back to Top]
This is a summary of the changes to the source code that went into this
release that was generated using the diffstat utility.
asterisk-18.2.0-summary.html | 169 -----
asterisk-18.2.0-summary.txt | 508 -----------------
b/.version | 2
b/CHANGES | 53 +
b/ChangeLog | 923 ++++++++++++++++++++++++++++++-
b/README.md | 8
b/UPGRADE.txt | 14
b/apps/app_dial.c | 14
b/apps/app_mixmonitor.c | 75 ++
b/apps/app_page.c | 13
b/apps/app_queue.c | 6
b/apps/app_read.c | 3
b/apps/app_transfer.c | 24
b/asterisk-18.3.0-rc2-summary.html | 14
b/asterisk-18.3.0-rc2-summary.txt | 102 +++
b/channels/chan_iax2.c | 40 +
b/channels/chan_pjsip.c | 41 +
b/channels/chan_sip.c | 60 --
b/configs/samples/func_odbc.conf.sample | 11
b/configs/samples/iax.conf.sample | 9
b/configs/samples/modules.conf.sample | 16
b/configs/samples/rtp.conf.sample | 12
b/configs/samples/stasis.conf.sample | 2
b/configure | 116 ---
b/configure.ac | 5
b/formats/format_wav.c | 3
b/funcs/func_callerid.c | 146 ++--
b/funcs/func_odbc.c | 31 -
b/include/asterisk/app.h | 7
b/include/asterisk/channel.h | 12
b/include/asterisk/core_unreal.h | 2
b/include/asterisk/manager.h | 2
b/include/asterisk/sorcery.h | 22
b/include/asterisk/stasis_channels.h | 26
b/include/asterisk/strings.h | 4
b/main/asterisk.c | 8
b/main/channel.c | 45 +
b/main/core_unreal.c | 92 +++
b/main/frame.c | 9
b/main/manager_channels.c | 56 +
b/main/say.c | 4
b/main/sorcery.c | 17
b/main/stasis.c | 3
b/main/stasis_channels.c | 9
b/main/translate.c | 24
b/res/res_agi.c | 6
b/res/res_config_pgsql.c | 32 -
b/res/res_fax.c | 12
b/res/res_format_attr_celt.c | 14
b/res/res_format_attr_h263.c | 141 ++++
b/res/res_format_attr_ilbc.c | 12
b/res/res_format_attr_opus.c | 31 -
b/res/res_format_attr_silk.c | 17
b/res/res_format_attr_siren14.c | 13
b/res/res_format_attr_siren7.c | 13
b/res/res_format_attr_vp8.c | 12
b/res/res_musiconhold.c | 10
b/res/res_odbc_transaction.c | 5
b/res/res_pjsip.c | 2
b/res/res_pjsip/pjsip_scheduler.c | 2
b/res/res_pjsip_diversion.c | 14
b/res/res_pjsip_endpoint_identifier_ip.c | 3
b/res/res_pjsip_nat.c | 24
b/res/res_pjsip_outbound_registration.c | 13
b/res/res_pjsip_path.c | 12
b/res/res_pjsip_pubsub.c | 2
b/res/res_pjsip_refer.c | 163 +++--
b/res/res_pjsip_registrar.c | 21
b/res/res_pjsip_sdp_rtp.c | 42 +
b/res/res_pjsip_session.c | 197 +++---
b/res/res_pjsip_t38.c | 9
b/res/res_rtp_asterisk.c | 75 ++
b/res/res_sorcery_config.c | 6
73 files changed, 2450 insertions(+), 1215 deletions(-)
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