From: Asterisk Development Team Date: Thu, 25 Mar 2021 17:34:21 +0000 (-0500) Subject: Update for 18.3.0 X-Git-Tag: 18.3.0^0 X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=refs%2Fheads%2F18.3;p=thirdparty%2Fasterisk.git Update for 18.3.0 --- diff --git a/.version b/.version index 7d6cefdd26..1667c1afed 100644 --- a/.version +++ b/.version @@ -1 +1 @@ -18.3.0-rc2 \ No newline at end of file +18.3.0 \ No newline at end of file diff --git a/ChangeLog b/ChangeLog index 89209f20cf..bc3da5bb2f 100644 --- a/ChangeLog +++ b/ChangeLog @@ -1,3 +1,7 @@ +2021-03-25 17:34 +0000 Asterisk Development Team + + * asterisk 18.3.0 Released. + 2021-03-22 15:41 +0000 Asterisk Development Team * asterisk 18.3.0-rc2 Released. diff --git a/asterisk-18.3.0-rc2-summary.html b/asterisk-18.3.0-rc2-summary.html deleted file mode 100644 index 6d797ba76c..0000000000 --- a/asterisk-18.3.0-rc2-summary.html +++ /dev/null @@ -1,14 +0,0 @@ -Release Summary - asterisk-18.3.0-rc2

Release Summary

asterisk-18.3.0-rc2

Date: 2021-03-22

<asteriskteam@digium.com>

Table of Contents

    -
  1. Summary
    2. -
  2. Contributors
    3. -
  3. Closed Issues
    4. -
  4. Open Issues
    5. -
  5. Diffstat
    6. -

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-18.3.0-rc1.

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.

- - -
CodersTestersReporters
1 Joshua C. Colp
1 George Joseph
1 sungtae kim
1 Matthias Hensler

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.

Bug

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.
    • -

Open Issues

[Back to Top]

This is a list of all open issues from the issue tracker that were referenced by changes that went into this release.

Bug

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
    • -

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.

0 files changed

\ No newline at end of file diff --git a/asterisk-18.3.0-rc2-summary.txt b/asterisk-18.3.0-rc2-summary.txt deleted file mode 100644 index 24dc63c73c..0000000000 --- a/asterisk-18.3.0-rc2-summary.txt +++ /dev/null @@ -1,102 +0,0 @@ - Release Summary - - asterisk-18.3.0-rc2 - - Date: 2021-03-22 - - - - ---------------------------------------------------------------------- - - Table of Contents - - 1. Summary - 2. Contributors - 3. Closed Issues - 4. Open Issues - 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-18.3.0-rc1. - - ---------------------------------------------------------------------- - - 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 - 1 Joshua C. Colp 1 sungtae kim - 1 George Joseph 1 Matthias Hensler - - ---------------------------------------------------------------------- - - 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. - - Bug - - 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. - - ---------------------------------------------------------------------- - - Open Issues - - [Back to Top] - - This is a list of all open issues from the issue tracker that were - referenced by changes that went into this release. - - Bug - - 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 - - ---------------------------------------------------------------------- - - 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. - - 0 files changed diff --git a/asterisk-18.3.0-summary.html b/asterisk-18.3.0-summary.html new file mode 100644 index 0000000000..9b2221400b --- /dev/null +++ b/asterisk-18.3.0-summary.html @@ -0,0 +1,202 @@ +Release Summary - asterisk-18.3.0

Release Summary

asterisk-18.3.0

Date: 2021-03-25

<asteriskteam@digium.com>

Table of Contents

    +
  1. Summary
    2. +
  2. Contributors
    3. +
  3. Closed Issues
    4. +
  4. Other Changes
    5. +
  5. Diffstat
    6. +

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.

+ + +
CodersTestersReporters
11 Alexander Traud
9 Joshua C. Colp
8 Sean Bright
6 Jaco Kroon
5 George Joseph
4 Ben Ford
3 Kevin Harwell
3 Asterisk Development Team
3 Ivan Poddubnyi
3 Boris P. Korzun
1 Salah Ahmed
1 Dan Cropp
1 Holger Hans Peter Freyther
1 Nico Kooijman
1 Alexei Gradinari
1 Torrey Searle
1 Nick French
1 Robert Cripps
1 Sebastien Duthil
1 Mark Petersen
1 Mark Petersen
6 Alexander Traud
3 Boris P. Korzun
3 Joshua C. Colp
2 Matthias Hensler
2 Stefan Ruf
2 Sebastian Damm
2 Gregory Massel
1 Rusty Newton
1 Alexei Gradinari
1 Ivan Poddubny
1 Jacek Konieczny
1 Jaco Kroon
1 Edvin Vidmar
1 Sébastien Duthil
1 Jean Aunis - Prescom
1 sungtae kim
1 Benjamin Keith Ford
1 Boolah
1 Nick French
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.

+ + + + + + + + + + + + + + + + + + + + + + + +
RevisionAuthorSummary
2c0e6bac06Asterisk Development TeamUpdate for 18.3.0-rc2
ae4a3da557Asterisk Development TeamUpdate for 18.3.0-rc1
263f906af4Kevin Harwellmanager: Increase the non breaking AMI version number
0afd37e3b5Asterisk Development TeamUpdate CHANGES and UPGRADE.txt for 18.3.0
23e41313a8Jaco Kroonfunc_callerid+res_agi: Fix compile errors related to -Werror=zero-length-bounds
52707fba7fJaco Kroonapp.h: Fix -Werror=zero-length-bounds compile errors in dev mode.
262473c6d9Alexander Traudres_format_attr_*: Parameter Names are Case-Insensitive.
4fc0e16838Alexander Traudchan_iax2: System Header strings is included via asterisk.h/compat.h.
16e4d1f36fSean Brightres_musiconhold.c: Plug ref leak caused by ao2_replace() misuse.
269bb08ea2George Josephres_pjsip_refer: Move the progress dlg release to a serializer
0323293142Alexander Traudres_format_attr_h263: Generate valid SDP fmtp for H.263+.
be0a61bc3dKevin Harwellres_rtp_asterisk: Add packet subtype during RTCP debug when relevant
1adf9368eeAlexander Traudchan_sip: Filter pass-through audio/video formats away, again.
bee35fe04aJaco Kroonfunc_odbc: Introduce minargs config and expose ARGC in addition to ARGn.
dbd8908f8dGeorge Josephres_pjsip_refer: Always serialize calls to refer_progress_notify
28f187d6c5George Josephchan_iax2.c: Require secret and auth method if encryption is enabled
24d6adfe99Sean Brightapp_read: Release tone zone reference on early return.
7c0fbaf010Ivan Poddubnyimain/frame: Add missing control frame names to ast_frame_subclass2str
fb42b60326Sean Brightres_pjsip_pubsub: Fix truncation of persisted SUBSCRIBE packet
9c56870929Jaco KroonAC_HEADER_STDC causes a compile failure with autoconf 2.70
a25bcf70edAlexander Traudpjsip_scheduler: Fix pjsip show scheduled_tasks like for compiler Clang.
87a35f8e94Ben Fordchan_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(-)

\ No newline at end of file diff --git a/asterisk-18.3.0-summary.txt b/asterisk-18.3.0-summary.txt new file mode 100644 index 0000000000..3995c9a89c --- /dev/null +++ b/asterisk-18.3.0-summary.txt @@ -0,0 +1,592 @@ + Release Summary + + asterisk-18.3.0 + + Date: 2021-03-25 + + + + ---------------------------------------------------------------------- + + 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(-)