From: Asterisk Development Team Date: Thu, 18 Feb 2021 16:48:09 +0000 (-0500) Subject: Update for 16.16.1 X-Git-Tag: 16.16.1^0 X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=3cd8afbdf2c6dffe80c631a5336e21bb1827a5cd;p=thirdparty%2Fasterisk.git Update for 16.16.1 --- diff --git a/.version b/.version index 0ff38047bb..140075a329 100644 --- a/.version +++ b/.version @@ -1 +1 @@ -16.16.0 \ No newline at end of file +16.16.1 \ No newline at end of file diff --git a/ChangeLog b/ChangeLog index 1eacb69941..f5aa31e5ba 100644 --- a/ChangeLog +++ b/ChangeLog @@ -1,3 +1,104 @@ +2021-02-18 16:48 +0000 Asterisk Development Team + + * asterisk 16.16.1 Released. + +2021-02-01 15:24 +0000 [a5619097cd] Kevin Harwell + + * AST-2021-002: Remote crash possible when negotiating T.38 + + When an endpoint requests to re-negotiate for fax and the incoming + re-invite is received prior to Asterisk sending out the 200 OK for + the initial invite the re-invite gets delayed. When Asterisk does + finally send the re-inivite the SDP includes streams for both audio + and T.38. + + This happens because when the pending topology and active topologies + differ (pending stream is not in the active) in the delayed scenario + the pending stream is appended to the active topology. However, in + the fax case the pending stream should replace the active. + + This patch makes it so when a delay occurs during fax negotiation, + to or from, the audio stream is replaced by the T.38 stream, or vice + versa instead of being appended. + + Further when Asterisk sent the re-invite with both audio and T.38, + and the endpoint responded with a declined T.38 stream then Asterisk + would crash when attempting to change the T.38 state. + + This patch also puts in a check that ensures the media state has a + valid fax session (associated udptl object) before changing the + T.38 state internally. + + ASTERISK-29203 #close + + Change-Id: I407f4fa58651255b6a9030d34fd6578cf65ccf09 + +2021-01-26 11:09 +0000 [3f4dfd5c02] Alexander Traud + + * rtp: Enable srtp replay protection + + Add option "srtpreplayprotection" rtp.conf to enable srtp + replay protection. + + ASTERISK-29260 + Reported by: Alexander Traud + + Change-Id: I5cd346e3c6b6812039d1901aa4b7be688173b458 + +2020-12-28 06:43 +0000 [17561b5e64] Ivan Poddubnyi + + * res_pjsip_diversion: Fix adding more than one histinfo to Supported + + New responses sent within a PJSIP sessions are based on those that were + sent before. Therefore, adding/modifying a header once causes it to be + sent on all responses that follow. + + Sending 181 Call Is Being Forwarded many times first adds "histinfo" + duplicated more and more, and eventually overflows past the array + boundary. + + This commit adds a check preventing adding "histinfo" more than once, + and skipping it if there is no more space in the header. + + Similar overflow situations can also occur in res_pjsip_path and + res_pjsip_outbound_registration so those were also modified to + check the bounds and suppress duplicate Supported values. + + ASTERISK-29227 + Reported by: Ivan Poddubny + + Change-Id: Id43704a1f1a0293e35cc7f844026f0b04f2ac322 + +2020-12-11 14:49 +0000 [4cea145aa9] Sean Bright + + * res_rtp_asterisk.c: Fix signed mismatch that leads to overflow + + ASTERISK-29205 #close + + Change-Id: Ib7aa65644e8df76e2378d7613ee7cf751b9d0bea + +2021-02-05 05:26 +0000 [321632b02e] Joshua C. Colp + + * pjsip: Make modify_local_offer2 tolerate previous failed SDP. + + If a remote side is broken and sends an SDP that can not be + negotiated the call will be torn down but there is a window + where a second 183 Session Progress or 200 OK that is forked + can be received that also attempts to negotiate SDP. Since + the code marked the SDP negotiation as being done and complete + prior to this it assumes that there is an active local and remote + SDP which it can modify, while in fact there is not as the SDP + did not successfully negotiate. Since there is no local or remote + SDP a crash occurs. + + This patch changes the pjmedia_sdp_neg_modify_local_offer2 + function to no longer assume that a previous SDP negotiation + was successful. + + ASTERISK-29196 + + Change-Id: I22de45916d3b05fdc2a67da92b3a38271ee5949e + 2021-01-21 16:28 +0000 Asterisk Development Team * asterisk 16.16.0 Released. diff --git a/asterisk-16.16.0-summary.html b/asterisk-16.16.0-summary.html deleted file mode 100644 index a139c952b1..0000000000 --- a/asterisk-16.16.0-summary.html +++ /dev/null @@ -1,158 +0,0 @@ -Release Summary - asterisk-16.16.0

Release Summary

asterisk-16.16.0

Date: 2021-01-21

<asteriskteam@digium.com>

Table of Contents

    -
  1. Summary
    2. -
  2. Contributors
    3. -
  3. Closed Issues
    4. -
  4. Open Issues
    5. -
  5. Other Changes
    6. -
  6. Diffstat
    7. -

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-16.15.0.

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
6 Sean Bright
4 Alexander Traud
3 George Joseph
3 Jaco Kroon
3 Joshua C. Colp
2 Asterisk Development Team
2 Ivan Poddubnyi
2 Sungtae Kim
1 Dan Cropp
1 Kevin Harwell
1 Boris P. Korzun
1 Jean Aunis
1 Torrey Searle
1 laszlovl
1 Richard Mudgett
1 Nathan Bruning
1 Pirmin Walthert
1 Stanislav
1 Alexander Greiner-Baer
1 Mark Petersen
4 Alexander Traud
2 Sean Bright
2 sungtae kim
2 George Joseph
1 Flole Systems
1 Michael Maier
1 Ivan Poddubny
1 Julien
1 Jaco Kroon
1 Jean Aunis - Prescom
1 Hendrik Wedhorn
1 Robert Sutton
1 Alex Hermann
1 Alex Hermann
1 Juan Carlos Castro y Castro
1 Boris P. Korzun
1 Alexander Greiner-Baer
1 Alexander Traud
1 Mark Petersen
1 Dan Cropp
1 Nathan Bruning
1 Mark Petersen
1 Michael Maier
1 Gant Liu
1 Schneur Rosenberg
1 Dan Cropp
1 Stanislav Abramenkov
1 Torrey Searle
1 laszlovl
1 Mikhail Ivanov

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_diversion

ASTERISK-29219: res_pjsip_diversion: Crash if Tel URI contains History-Info
Reported by: Torrey Searle
    -
  • [9196e0d1d5] Torrey Searle -- res/res_pjsip_diversion: prevent crash on tel: uri in History-Info
    • -

Bug

Category: Applications/app_chanspy

ASTERISK-28883: Spyee information ist missing in ChanSpyStop AMI Event
Reported by: Hendrik Wedhorn
    -
  • [0a23296834] Sean Bright -- app_chanspy: Spyee information missing in ChanSpyStop AMI Event
    • -

Category: Applications/app_mixmonitor

ASTERISK-28947: Segmentation fault in mixmonitor_ds_destroy
Reported by: Robert Sutton
    -
  • [e96f744816] Kevin Harwell -- app_mixmonitor: cleanup datastore when monitor thread fails to launch
    • -

Category: Applications/app_queue

ASTERISK-29155: app_queue: Deadlock between queues container and individual queues
Reported by: George Joseph
    -
  • [8d8c9db618] George Joseph -- app_queue: Fix deadlock between update and show queues
    • -

Category: Bridges/bridge_simple

ASTERISK-29161: Incorrect setup of recall channels
Reported by: Boris P. Korzun
    -
  • [89d3de37ca] Boris P. Korzun -- bridge_basic: Fixed setup of recall channels
    • -

Category: Channels/chan_pjsip

ASTERISK-29240: chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN instead of a channel variable
Reported by: Ivan Poddubny
    -
  • [97afc9055f] Ivan Poddubnyi -- chan_pjsip: Assign SIPDOMAIN after creating a channel
    • -
ASTERISK-27902: chan_pjsip isn't updating hangupcause on 4XX responses
Reported by: George Joseph
    -
  • [17fa9c93d0] Ivan Poddubnyi -- chan_pjsip: Stop queueing control frames twice on outgoing channels
    • -
ASTERISK-28016: PJSIP sends duplicate 183 Progress responses
Reported by: Alex Hermann
    -
  • [17fa9c93d0] Ivan Poddubnyi -- chan_pjsip: Stop queueing control frames twice on outgoing channels
    • -
ASTERISK-28185: chan_pjsip: Subsequent same responses are not stopped
Reported by: Julien
    -
  • [17fa9c93d0] Ivan Poddubnyi -- chan_pjsip: Stop queueing control frames twice on outgoing channels
    • -
ASTERISK-29230: pjsip: Asterisk goes crazy and massively spams logfile if registration can't be send
Reported by: Michael Maier
    -
  • [7ed20b9d3b] George Joseph -- Revert "res_pjsip_outbound_registration.c: Use our own scheduler and other stuff"
    • -
ASTERISK-29201: Crash occurs when Transfer and execute Hangup before the Transfer result
Reported by: Dan Cropp
    -
  • [e127a57761] Dan Cropp -- chan_pjsip: Incorporate channel reference count into transfer_refer().
    • -
ASTERISK-29022: Crash when manipulating PJSIP invite dlg ref counts
Reported by: Sean Bright
    -
  • [ea744ca7c2] Joshua C. Colp -- pjsip: Match lifetime of INVITE session to our session.
    • -

Category: Channels/chan_sip/CodecHandling

ASTERISK-29238: chan_sip: SDP: Offers without any enabled stream are accepted.
Reported by: Alexander Traud
    -
  • [34b0960310] Alexander Traud -- chan_sip: SDP: Sidestep stream parsing when its media is disabled.
    • -
ASTERISK-29237: chan_sip: SDP: m=video is parsed even when disabled.
Reported by: Alexander Traud
    -
  • [34b0960310] Alexander Traud -- chan_sip: SDP: Sidestep stream parsing when its media is disabled.
    • -

Category: Channels/chan_sip/SRTP

ASTERISK-29222: chan_sip: Hold/Resume an sRTP call on a video enabled user-agent.
Reported by: Alexander Traud
    -
  • [34b0960310] Alexander Traud -- chan_sip: SDP: Sidestep stream parsing when its media is disabled.
    • -

Category: Channels/chan_sip/TCP-TLS

ASTERISK-28798: [patch] chan_sip: TCP/TLS client without server.
Reported by: Alexander Traud
    -
  • [f667c5a781] Alexander Traud -- chan_sip: Remove unused sip_socket->port.
    • -

Category: Channels/chan_sip/Video

ASTERISK-29238: chan_sip: SDP: Offers without any enabled stream are accepted.
Reported by: Alexander Traud
    -
  • [34b0960310] Alexander Traud -- chan_sip: SDP: Sidestep stream parsing when its media is disabled.
    • -
ASTERISK-29237: chan_sip: SDP: m=video is parsed even when disabled.
Reported by: Alexander Traud
    -
  • [34b0960310] Alexander Traud -- chan_sip: SDP: Sidestep stream parsing when its media is disabled.
    • -

Category: Core/Logging

ASTERISK-29209: Debug messages printed by scope trace might be missing newlines
Reported by: Alexander Traud
    -
  • [5a2867efa9] George Joseph -- logger.c: Automatically add a newline to formats that don't have one
    • -

Category: Functions/func_lock

ASTERISK-29217: LOCK() can grant the same lock to multiple channels spuriously
Reported by: Jaco Kroon
    -
  • [32e36144c7] Jaco Kroon -- func_lock: fix multiple-channel-grant problems.
    • -

Category: General

ASTERISK-29148: AST_MODULE_INFO no, MODULEINFO depend
Reported by: Alexander Traud
    -
  • [4c79bc19d1] Alexander Traud -- loader: Sync load- and build-time deps.
    • -

Category: Resources/res_ari_channels

ASTERISK-29188: null media causing the Asterisk crash
Reported by: sungtae kim
    -
  • [a47e6965b3] Sungtae Kim -- res_ari: Fix wrong media uri handle for channel play
    • -

Category: Resources/res_http_media_cache

ASTERISK-29173: Media cache URL requests allow infinite redirects
Reported by: Sean Bright
    -
  • [0c185c9e21] Sean Bright -- res_http_media_cache.c: Set reasonable number of redirects
    • -

Category: Resources/res_musiconhold

ASTERISK-29211: res_musiconhold: Segfault on realtime music on hold without entries
Reported by: Nathan Bruning
    -
  • [bb46595799] Nathan Bruning -- res_musiconhold: Don't crash when real-time doesn't return any entries
    • -

Category: Resources/res_pjsip

ASTERISK-29165: res_pjsip: malformed header Accept-Encoding in OPTIONS response
Reported by: Alexander Greiner-Baer
    -
  • [a8f6238cc8] Alexander Greiner-Baer -- res_pjsip: set Accept-Encoding to identity in OPTIONS response
    • -

Category: Resources/res_pjsip_diversion

ASTERISK-29191: tel: URI in Diversion header causes crash
Reported by: Mikhail Ivanov
    -
  • [9196e0d1d5] Torrey Searle -- res/res_pjsip_diversion: prevent crash on tel: uri in History-Info
    • -

Category: Resources/res_pjsip_outbound_registration

ASTERISK-29231: pjsip: SIGSEGV in CLI if no trunk is registered
Reported by: Michael Maier
    -
  • [7ed20b9d3b] George Joseph -- Revert "res_pjsip_outbound_registration.c: Use our own scheduler and other stuff"
    • -

Category: Resources/res_pjsip_session

ASTERISK-29240: chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN instead of a channel variable
Reported by: Ivan Poddubny
    -
  • [97afc9055f] Ivan Poddubnyi -- chan_pjsip: Assign SIPDOMAIN after creating a channel
    • -

Category: Resources/res_stasis

ASTERISK-29229: Stasis/messaging: text messages not dispatched to all subscribers when using generic subscription
Reported by: Jean Aunis - Prescom
    -
  • [45e1d89135] Jean Aunis -- Stasis/messaging: tech subscriptions conflict with endpoint subscriptions.
    • -

Category: Resources/res_stir_shaken

ASTERISK-29175: res_pjsip_stir_shaken: Fix module description
Reported by: Stanislav Abramenkov
    -
  • [159522003a] Stanislav -- res_pjsip_stir_shaken: Fix module description
    • -

Category: pjproject/pjsip

ASTERISK-29191: tel: URI in Diversion header causes crash
Reported by: Mikhail Ivanov
    -
  • [9196e0d1d5] Torrey Searle -- res/res_pjsip_diversion: prevent crash on tel: uri in History-Info
    • -
ASTERISK-29024: pjsip: Route Header in Cancel request incorrectly set
Reported by: Flole Systems
    -
  • [11def974a8] Pirmin Walthert -- res_pjsip_nat.c: Create deep copies of strings when appropriate
    • -

Improvement

Category: Applications/app_voicemail/NewFeature

ASTERISK-29118: VoiceMail() should have an option to play greetings as Early Media
Reported by: Juan Carlos Castro y Castro
    -
  • [15566494f9] Joshua C. Colp -- voicemail: add option 'e' to play greetings as early media
    • -

Category: Channels/chan_pjsip

ASTERISK-28549: Two repeated 183
Reported by: Gant Liu
    -
  • [17fa9c93d0] Ivan Poddubnyi -- chan_pjsip: Stop queueing control frames twice on outgoing channels
    • -

Category: Contrib/General

ASTERISK-29216: contrib: systemd asterisk service for centos8 or other newer linux versions
Reported by: Mark Petersen
    -
  • [7df88c98d0] Jaco Kroon -- contrib/systemd: Added note on common issues with systemd and asterisk
    • -

Category: Resources/res_http_media_cache

ASTERISK-29143: res_http_media_cache: HTTP media cache stored hardcoded in /tmp
Reported by: laszlovl
    -
  • [8d2558209b] laszlovl -- Introduce astcachedir, to be used for temporary bucket files
    • -

Category: Resources/res_pjsip_session

ASTERISK-28549: Two repeated 183
Reported by: Gant Liu
    -
  • [17fa9c93d0] Ivan Poddubnyi -- chan_pjsip: Stop queueing control frames twice on outgoing channels
    • -

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: Applications/app_voicemail/ODBC

ASTERISK-28992: app_voicemail: Deadlock in ODBC when retrieving file
Reported by: Schneur Rosenberg
    -
  • [2b7af3eb27] Sean Bright -- app_voicemail: Prevent deadlocks when out of ODBC database connections
    • -

Category: Resources/res_pjsip_session

ASTERISK-29215: res_pjsip_session: NULL active_media_state topology caused asterisk crash
Reported by: sungtae kim
    -
  • [ab3f57d88f] Sungtae Kim -- res_pjsip_session: Fixed NULL active media topology handle
    • -

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
e08d866884Asterisk Development TeamUpdate for 16.16.0-rc1
6056818467Asterisk Development TeamUpdate CHANGES and UPGRADE.txt for 16.16.0
aca435dfe7Jaco Kroonpbx_lua: Add LUA_VERSIONS environment variable to ./configure.
4c5bffb217Sean Brightasterisk: Export additional manager functions
89cf7899beRichard Mudgettres_pjsip_session.c: Fix compiler warnings.
7e4bb4ed11Joshua C. Colpres_pjsip_pidf_digium_body_supplement: Support Sangoma user agent.
ddbf3a7f73Sean Brightmedia_cache: Fix reference leak with bucket file metadata
a360150ee0Sean BrightCHANGES: Remove already applied CHANGES update
d1a78e047dAlexander Traudmodules.conf: Align the comments for more conclusiveness.

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-16.15.0-summary.html                 |  213 ----------
-asterisk-16.15.0-summary.txt                  |  543 --------------------------
-b/.version                                    |    2
-b/CHANGES                                     |   18
-b/ChangeLog                                   |  507 ++++++++++++++++++++++++
-b/Makefile                                    |    6
-b/apps/app_chanspy.c                          |    6
-b/apps/app_mixmonitor.c                       |   23 +
-b/apps/app_queue.c                            |  245 ++++++-----
-b/apps/app_voicemail.c                        |   36 +
-b/asterisk-16.16.0-rc1-summary.html           |  164 +++++++
-b/asterisk-16.16.0-rc1-summary.txt            |  492 +++++++++++++++++++++++
-b/build_tools/install_subst                   |    1
-b/build_tools/make_defaults_h                 |    1
-b/build_tools/mkpkgconfig                     |    1
-b/channels/chan_pjsip.c                       |  214 +++-------
-b/channels/chan_sip.c                         |   32 -
-b/channels/sip/include/sip.h                  |    2
-b/configs/basic-pbx/modules.conf              |    8
-b/configs/samples/asterisk.conf.sample        |    1
-b/configs/samples/modules.conf.sample         |    9
-b/configure                                   |   11
-b/configure.ac                                |    9
-b/contrib/systemd/asterisk.service            |    7
-b/funcs/func_lock.c                           |  163 ++-----
-b/funcs/func_odbc.c                           |    1
-b/funcs/func_periodic_hook.c                  |    1
-b/include/asterisk/manager.h                  |    4
-b/include/asterisk/paths.h                    |    1
-b/main/asterisk.c                             |    4
-b/main/bridge_basic.c                         |    2
-b/main/bucket.c                               |    3
-b/main/logger.c                               |    5
-b/main/manager.c                              |    6
-b/main/manager_channels.c                     |   18
-b/main/media_cache.c                          |    1
-b/main/options.c                              |    7
-b/main/pbx_variables.c                        |    2
-b/makeopts.in                                 |    1
-b/res/res_hep_pjsip.c                         |    2
-b/res/res_http_media_cache.c                  |    1
-b/res/res_musiconhold.c                       |   21 -
-b/res/res_odbc.c                              |    1
-b/res/res_pjproject.c                         |    2
-b/res/res_pjsip.c                             |    2
-b/res/res_pjsip/pjsip_options.c               |    2
-b/res/res_pjsip_diversion.c                   |   11
-b/res/res_pjsip_dlg_options.c                 |    2
-b/res/res_pjsip_nat.c                         |   10
-b/res/res_pjsip_outbound_registration.c       |  296 +++++---------
-b/res/res_pjsip_pidf_digium_body_supplement.c |    8
-b/res/res_pjsip_session.c                     |   66 +--
-b/res/res_pjsip_stir_shaken.c                 |    4
-b/res/res_pjsip_transport_websocket.c         |    2
-b/res/res_stasis_playback.c                   |    7
-b/res/res_stasis_snoop.c                      |   12
-b/res/stasis/messaging.c                      |   33 +
-doc/CHANGES-staging/hide_messaging_ami_events |   11
-58 files changed, 1826 insertions(+), 1437 deletions(-)

\ No newline at end of file diff --git a/asterisk-16.16.0-summary.txt b/asterisk-16.16.0-summary.txt deleted file mode 100644 index 2797ad340e..0000000000 --- a/asterisk-16.16.0-summary.txt +++ /dev/null @@ -1,486 +0,0 @@ - Release Summary - - asterisk-16.16.0 - - Date: 2021-01-21 - - - - ---------------------------------------------------------------------- - - Table of Contents - - 1. Summary - 2. Contributors - 3. Closed Issues - 4. Open Issues - 5. Other Changes - 6. 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-16.15.0. - - ---------------------------------------------------------------------- - - 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 - 6 Sean Bright 1 Mark Petersen 4 Alexander Traud - 4 Alexander Traud 2 Sean Bright - 3 George Joseph 2 sungtae kim - 3 Jaco Kroon 2 George Joseph - 3 Joshua C. Colp 1 Flole Systems - 2 Asterisk Development Team 1 Michael Maier - 2 Ivan Poddubnyi 1 Ivan Poddubny - 2 Sungtae Kim 1 Julien - 1 Dan Cropp 1 Jaco Kroon - 1 Kevin Harwell 1 Jean Aunis - Prescom - 1 Boris P. Korzun 1 Hendrik Wedhorn - 1 Jean Aunis 1 Robert Sutton - 1 Torrey Searle 1 Alex Hermann - 1 laszlovl 1 Alex Hermann - 1 Richard Mudgett 1 Juan Carlos Castro y Castro - 1 Nathan Bruning 1 Boris P. Korzun - 1 Pirmin Walthert 1 Alexander Greiner-Baer - 1 Stanislav 1 Alexander Traud - 1 Alexander Greiner-Baer 1 Mark Petersen - 1 Dan Cropp - 1 Nathan Bruning - 1 Mark Petersen - 1 Michael Maier - 1 Gant Liu - 1 Schneur Rosenberg - 1 Dan Cropp - 1 Stanislav Abramenkov - 1 Torrey Searle - 1 laszlovl - 1 Mikhail Ivanov - - ---------------------------------------------------------------------- - - 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_diversion - - ASTERISK-29219: res_pjsip_diversion: Crash if Tel URI contains - History-Info - Reported by: Torrey Searle - * [9196e0d1d5] Torrey Searle -- res/res_pjsip_diversion: prevent crash - on tel: uri in History-Info - - Bug - - Category: Applications/app_chanspy - - ASTERISK-28883: Spyee information ist missing in ChanSpyStop AMI Event - Reported by: Hendrik Wedhorn - * [0a23296834] Sean Bright -- app_chanspy: Spyee information missing in - ChanSpyStop AMI Event - - Category: Applications/app_mixmonitor - - ASTERISK-28947: Segmentation fault in mixmonitor_ds_destroy - Reported by: Robert Sutton - * [e96f744816] Kevin Harwell -- app_mixmonitor: cleanup datastore when - monitor thread fails to launch - - Category: Applications/app_queue - - ASTERISK-29155: app_queue: Deadlock between queues container and - individual queues - Reported by: George Joseph - * [8d8c9db618] George Joseph -- app_queue: Fix deadlock between update - and show queues - - Category: Bridges/bridge_simple - - ASTERISK-29161: Incorrect setup of recall channels - Reported by: Boris P. Korzun - * [89d3de37ca] Boris P. Korzun -- bridge_basic: Fixed setup of recall - channels - - Category: Channels/chan_pjsip - - ASTERISK-29240: chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN - instead of a channel variable - Reported by: Ivan Poddubny - * [97afc9055f] Ivan Poddubnyi -- chan_pjsip: Assign SIPDOMAIN after - creating a channel - ASTERISK-27902: chan_pjsip isn't updating hangupcause on 4XX responses - Reported by: George Joseph - * [17fa9c93d0] Ivan Poddubnyi -- chan_pjsip: Stop queueing control - frames twice on outgoing channels - ASTERISK-28016: PJSIP sends duplicate 183 Progress responses - Reported by: Alex Hermann - * [17fa9c93d0] Ivan Poddubnyi -- chan_pjsip: Stop queueing control - frames twice on outgoing channels - ASTERISK-28185: chan_pjsip: Subsequent same responses are not stopped - Reported by: Julien - * [17fa9c93d0] Ivan Poddubnyi -- chan_pjsip: Stop queueing control - frames twice on outgoing channels - ASTERISK-29230: pjsip: Asterisk goes crazy and massively spams logfile if - registration can't be send - Reported by: Michael Maier - * [7ed20b9d3b] George Joseph -- Revert - "res_pjsip_outbound_registration.c: Use our own scheduler and other - stuff" - ASTERISK-29201: Crash occurs when Transfer and execute Hangup before the - Transfer result - Reported by: Dan Cropp - * [e127a57761] Dan Cropp -- chan_pjsip: Incorporate channel reference - count into transfer_refer(). - ASTERISK-29022: Crash when manipulating PJSIP invite dlg ref counts - Reported by: Sean Bright - * [ea744ca7c2] Joshua C. Colp -- pjsip: Match lifetime of INVITE session - to our session. - - Category: Channels/chan_sip/CodecHandling - - ASTERISK-29238: chan_sip: SDP: Offers without any enabled stream are - accepted. - Reported by: Alexander Traud - * [34b0960310] Alexander Traud -- chan_sip: SDP: Sidestep stream parsing - when its media is disabled. - ASTERISK-29237: chan_sip: SDP: m=video is parsed even when disabled. - Reported by: Alexander Traud - * [34b0960310] Alexander Traud -- chan_sip: SDP: Sidestep stream parsing - when its media is disabled. - - Category: Channels/chan_sip/SRTP - - ASTERISK-29222: chan_sip: Hold/Resume an sRTP call on a video enabled - user-agent. - Reported by: Alexander Traud - * [34b0960310] Alexander Traud -- chan_sip: SDP: Sidestep stream parsing - when its media is disabled. - - Category: Channels/chan_sip/TCP-TLS - - ASTERISK-28798: [patch] chan_sip: TCP/TLS client without server. - Reported by: Alexander Traud - * [f667c5a781] Alexander Traud -- chan_sip: Remove unused - sip_socket->port. - - Category: Channels/chan_sip/Video - - ASTERISK-29238: chan_sip: SDP: Offers without any enabled stream are - accepted. - Reported by: Alexander Traud - * [34b0960310] Alexander Traud -- chan_sip: SDP: Sidestep stream parsing - when its media is disabled. - ASTERISK-29237: chan_sip: SDP: m=video is parsed even when disabled. - Reported by: Alexander Traud - * [34b0960310] Alexander Traud -- chan_sip: SDP: Sidestep stream parsing - when its media is disabled. - - Category: Core/Logging - - ASTERISK-29209: Debug messages printed by scope trace might be missing - newlines - Reported by: Alexander Traud - * [5a2867efa9] George Joseph -- logger.c: Automatically add a newline to - formats that don't have one - - Category: Functions/func_lock - - ASTERISK-29217: LOCK() can grant the same lock to multiple channels - spuriously - Reported by: Jaco Kroon - * [32e36144c7] Jaco Kroon -- func_lock: fix multiple-channel-grant - problems. - - Category: General - - ASTERISK-29148: AST_MODULE_INFO no, MODULEINFO depend - Reported by: Alexander Traud - * [4c79bc19d1] Alexander Traud -- loader: Sync load- and build-time - deps. - - Category: Resources/res_ari_channels - - ASTERISK-29188: null media causing the Asterisk crash - Reported by: sungtae kim - * [a47e6965b3] Sungtae Kim -- res_ari: Fix wrong media uri handle for - channel play - - Category: Resources/res_http_media_cache - - ASTERISK-29173: Media cache URL requests allow infinite redirects - Reported by: Sean Bright - * [0c185c9e21] Sean Bright -- res_http_media_cache.c: Set reasonable - number of redirects - - Category: Resources/res_musiconhold - - ASTERISK-29211: res_musiconhold: Segfault on realtime music on hold - without entries - Reported by: Nathan Bruning - * [bb46595799] Nathan Bruning -- res_musiconhold: Don't crash when - real-time doesn't return any entries - - Category: Resources/res_pjsip - - ASTERISK-29165: res_pjsip: malformed header Accept-Encoding in OPTIONS - response - Reported by: Alexander Greiner-Baer - * [a8f6238cc8] Alexander Greiner-Baer -- res_pjsip: set Accept-Encoding - to identity in OPTIONS response - - Category: Resources/res_pjsip_diversion - - ASTERISK-29191: tel: URI in Diversion header causes crash - Reported by: Mikhail Ivanov - * [9196e0d1d5] Torrey Searle -- res/res_pjsip_diversion: prevent crash - on tel: uri in History-Info - - Category: Resources/res_pjsip_outbound_registration - - ASTERISK-29231: pjsip: SIGSEGV in CLI if no trunk is registered - Reported by: Michael Maier - * [7ed20b9d3b] George Joseph -- Revert - "res_pjsip_outbound_registration.c: Use our own scheduler and other - stuff" - - Category: Resources/res_pjsip_session - - ASTERISK-29240: chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN - instead of a channel variable - Reported by: Ivan Poddubny - * [97afc9055f] Ivan Poddubnyi -- chan_pjsip: Assign SIPDOMAIN after - creating a channel - - Category: Resources/res_stasis - - ASTERISK-29229: Stasis/messaging: text messages not dispatched to all - subscribers when using generic subscription - Reported by: Jean Aunis - Prescom - * [45e1d89135] Jean Aunis -- Stasis/messaging: tech subscriptions - conflict with endpoint subscriptions. - - Category: Resources/res_stir_shaken - - ASTERISK-29175: res_pjsip_stir_shaken: Fix module description - Reported by: Stanislav Abramenkov - * [159522003a] Stanislav -- res_pjsip_stir_shaken: Fix module - description - - Category: pjproject/pjsip - - ASTERISK-29191: tel: URI in Diversion header causes crash - Reported by: Mikhail Ivanov - * [9196e0d1d5] Torrey Searle -- res/res_pjsip_diversion: prevent crash - on tel: uri in History-Info - ASTERISK-29024: pjsip: Route Header in Cancel request incorrectly set - Reported by: Flole Systems - * [11def974a8] Pirmin Walthert -- res_pjsip_nat.c: Create deep copies of - strings when appropriate - - Improvement - - Category: Applications/app_voicemail/NewFeature - - ASTERISK-29118: VoiceMail() should have an option to play greetings as - Early Media - Reported by: Juan Carlos Castro y Castro - * [15566494f9] Joshua C. Colp -- voicemail: add option 'e' to play - greetings as early media - - Category: Channels/chan_pjsip - - ASTERISK-28549: Two repeated 183 - Reported by: Gant Liu - * [17fa9c93d0] Ivan Poddubnyi -- chan_pjsip: Stop queueing control - frames twice on outgoing channels - - Category: Contrib/General - - ASTERISK-29216: contrib: systemd asterisk service for centos8 or other - newer linux versions - Reported by: Mark Petersen - * [7df88c98d0] Jaco Kroon -- contrib/systemd: Added note on common - issues with systemd and asterisk - - Category: Resources/res_http_media_cache - - ASTERISK-29143: res_http_media_cache: HTTP media cache stored hardcoded in - /tmp - Reported by: laszlovl - * [8d2558209b] laszlovl -- Introduce astcachedir, to be used for - temporary bucket files - - Category: Resources/res_pjsip_session - - ASTERISK-28549: Two repeated 183 - Reported by: Gant Liu - * [17fa9c93d0] Ivan Poddubnyi -- chan_pjsip: Stop queueing control - frames twice on outgoing channels - - ---------------------------------------------------------------------- - - 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: Applications/app_voicemail/ODBC - - ASTERISK-28992: app_voicemail: Deadlock in ODBC when retrieving file - Reported by: Schneur Rosenberg - * [2b7af3eb27] Sean Bright -- app_voicemail: Prevent deadlocks when out - of ODBC database connections - - Category: Resources/res_pjsip_session - - ASTERISK-29215: res_pjsip_session: NULL active_media_state topology caused - asterisk crash - Reported by: sungtae kim - * [ab3f57d88f] Sungtae Kim -- res_pjsip_session: Fixed NULL active media - topology handle - - ---------------------------------------------------------------------- - - 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 | - |------------+-------------+---------------------------------------------| - | | Asterisk | | - | e08d866884 | Development | Update for 16.16.0-rc1 | - | | Team | | - |------------+-------------+---------------------------------------------| - | | Asterisk | | - | 6056818467 | Development | Update CHANGES and UPGRADE.txt for 16.16.0 | - | | Team | | - |------------+-------------+---------------------------------------------| - | aca435dfe7 | Jaco Kroon | pbx_lua: Add LUA_VERSIONS environment | - | | | variable to ./configure. | - |------------+-------------+---------------------------------------------| - | 4c5bffb217 | Sean Bright | asterisk: Export additional manager | - | | | functions | - |------------+-------------+---------------------------------------------| - | 89cf7899be | Richard | res_pjsip_session.c: Fix compiler warnings. | - | | Mudgett | | - |------------+-------------+---------------------------------------------| - | 7e4bb4ed11 | Joshua C. | res_pjsip_pidf_digium_body_supplement: | - | | Colp | Support Sangoma user agent. | - |------------+-------------+---------------------------------------------| - | ddbf3a7f73 | Sean Bright | media_cache: Fix reference leak with bucket | - | | | file metadata | - |------------+-------------+---------------------------------------------| - | a360150ee0 | Sean Bright | CHANGES: Remove already applied CHANGES | - | | | update | - |------------+-------------+---------------------------------------------| - | d1a78e047d | Alexander | modules.conf: Align the comments for more | - | | Traud | conclusiveness. | - +------------------------------------------------------------------------+ - - ---------------------------------------------------------------------- - - 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-16.15.0-summary.html | 213 ---------- - asterisk-16.15.0-summary.txt | 543 -------------------------- - b/.version | 2 - b/CHANGES | 18 - b/ChangeLog | 507 ++++++++++++++++++++++++ - b/Makefile | 6 - b/apps/app_chanspy.c | 6 - b/apps/app_mixmonitor.c | 23 + - b/apps/app_queue.c | 245 ++++++----- - b/apps/app_voicemail.c | 36 + - b/asterisk-16.16.0-rc1-summary.html | 164 +++++++ - b/asterisk-16.16.0-rc1-summary.txt | 492 +++++++++++++++++++++++ - b/build_tools/install_subst | 1 - b/build_tools/make_defaults_h | 1 - b/build_tools/mkpkgconfig | 1 - b/channels/chan_pjsip.c | 214 +++------- - b/channels/chan_sip.c | 32 - - b/channels/sip/include/sip.h | 2 - b/configs/basic-pbx/modules.conf | 8 - b/configs/samples/asterisk.conf.sample | 1 - b/configs/samples/modules.conf.sample | 9 - b/configure | 11 - b/configure.ac | 9 - b/contrib/systemd/asterisk.service | 7 - b/funcs/func_lock.c | 163 ++----- - b/funcs/func_odbc.c | 1 - b/funcs/func_periodic_hook.c | 1 - b/include/asterisk/manager.h | 4 - b/include/asterisk/paths.h | 1 - b/main/asterisk.c | 4 - b/main/bridge_basic.c | 2 - b/main/bucket.c | 3 - b/main/logger.c | 5 - b/main/manager.c | 6 - b/main/manager_channels.c | 18 - b/main/media_cache.c | 1 - b/main/options.c | 7 - b/main/pbx_variables.c | 2 - b/makeopts.in | 1 - b/res/res_hep_pjsip.c | 2 - b/res/res_http_media_cache.c | 1 - b/res/res_musiconhold.c | 21 - - b/res/res_odbc.c | 1 - b/res/res_pjproject.c | 2 - b/res/res_pjsip.c | 2 - b/res/res_pjsip/pjsip_options.c | 2 - b/res/res_pjsip_diversion.c | 11 - b/res/res_pjsip_dlg_options.c | 2 - b/res/res_pjsip_nat.c | 10 - b/res/res_pjsip_outbound_registration.c | 296 +++++--------- - b/res/res_pjsip_pidf_digium_body_supplement.c | 8 - b/res/res_pjsip_session.c | 66 +-- - b/res/res_pjsip_stir_shaken.c | 4 - b/res/res_pjsip_transport_websocket.c | 2 - b/res/res_stasis_playback.c | 7 - b/res/res_stasis_snoop.c | 12 - b/res/stasis/messaging.c | 33 + - doc/CHANGES-staging/hide_messaging_ami_events | 11 - 58 files changed, 1826 insertions(+), 1437 deletions(-) diff --git a/asterisk-16.16.1-summary.html b/asterisk-16.16.1-summary.html new file mode 100644 index 0000000000..27ccdab909 --- /dev/null +++ b/asterisk-16.16.1-summary.html @@ -0,0 +1,34 @@ +Release Summary - asterisk-16.16.1

Release Summary

asterisk-16.16.1

Date: 2021-02-18

<asteriskteam@digium.com>

Table of Contents

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

Summary

[Back to Top]

This release has been made to address one or more security vulnerabilities that have been identified. A security advisory document has been published for each vulnerability that includes additional information. Users of versions of Asterisk that are affected are strongly encouraged to review the advisories and determine what action they should take to protect their systems from these issues.

Security Advisories:

The data in this summary reflects changes that have been made since the previous release, asterisk-16.16.0.

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
1 Ivan Poddubnyi
1 Sean Bright
1 Kevin Harwell
1 Alexander Traud
1 Joshua C. Colp
1 Mauri de Souza Meneguzzo (3CPlus)
1 Ivan Poddubny
1 Ivan Poddubny
1 Edvin Vidmar
1 Alexander Traud
1 Gregory Massel
1 Alexander Traud

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_srtp

ASTERISK-29260: sRTP Replay Protection ignored; even tears down long calls
Reported by: Alexander Traud
    +
  • [3f4dfd5c02] 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
    +
  • [17561b5e64] Ivan Poddubnyi -- res_pjsip_diversion: Fix adding more than one histinfo to Supported
    • +

Bug

Category: Resources/res_pjsip

ASTERISK-29196: res_pjsip: Segmentation fault
Reported by: Mauri de Souza Meneguzzo (3CPlus)
    +
  • [321632b02e] Joshua C. Colp -- pjsip: Make modify_local_offer2 tolerate previous failed SDP.
    • +

Category: Resources/res_pjsip_session

ASTERISK-29203: res_pjsip_t38: Crash when changing state
Reported by: Gregory Massel
    +
  • [a5619097cd] Kevin Harwell -- AST-2021-002: Remote crash possible when negotiating T.38
    • +

Category: Resources/res_pjsip_t38

ASTERISK-29203: res_pjsip_t38: Crash when changing state
Reported by: Gregory Massel
    +
  • [a5619097cd] Kevin Harwell -- AST-2021-002: Remote crash possible when negotiating T.38
    • +

Category: Resources/res_rtp_asterisk

ASTERISK-29205: res_rtp_asterisk: Asterisk crashes when making hold/unhold from webrtc client
Reported by: Edvin Vidmar
    +
  • [4cea145aa9] Sean Bright -- res_rtp_asterisk.c: Fix signed mismatch that leads to overflow
    • +

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.

configs/samples/rtp.conf.sample                                         |   12 +++++++
+doc/CHANGES-staging/srtp_replay_protection.txt                          |    9 +++++
+doc/UPGRADE-staging/srtp_replay_protection.txt                          |    9 +++++
+res/res_pjsip_diversion.c                                               |   14 ++++++++
+res/res_pjsip_outbound_registration.c                                   |   12 +++++++
+res/res_pjsip_path.c                                                    |   12 +++++++
+res/res_pjsip_session.c                                                 |    9 +++++
+res/res_pjsip_t38.c                                                     |    9 +++++
+res/res_rtp_asterisk.c                                                  |   16 +++++++---
+res/res_srtp.c                                                          |    5 +--
+third-party/pjproject/patches/0080-fix-sdp-neg-modify-local-offer.patch |   15 +++++++++
+11 files changed, 115 insertions(+), 7 deletions(-)

\ No newline at end of file diff --git a/asterisk-16.16.1-summary.txt b/asterisk-16.16.1-summary.txt new file mode 100644 index 0000000000..bf9ca7d64e --- /dev/null +++ b/asterisk-16.16.1-summary.txt @@ -0,0 +1,138 @@ + Release Summary + + asterisk-16.16.1 + + Date: 2021-02-18 + + + + ---------------------------------------------------------------------- + + Table of Contents + + 1. Summary + 2. Contributors + 3. Closed Issues + 4. Diffstat + + ---------------------------------------------------------------------- + + Summary + + [Back to Top] + + This release has been made to address one or more security vulnerabilities + that have been identified. A security advisory document has been published + for each vulnerability that includes additional information. Users of + versions of Asterisk that are affected are strongly encouraged to review + the advisories and determine what action they should take to protect their + systems from these issues. + + Security Advisories: + + * AST-2021-001,AST-2021-002,AST-2021-003,AST-2021-004,AST-2021-005 + + The data in this summary reflects changes that have been made since the + previous release, asterisk-16.16.0. + + ---------------------------------------------------------------------- + + 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 + 1 Ivan Poddubnyi 1 Mauri de Souza Meneguzzo (3CPlus) + 1 Sean Bright 1 Ivan Poddubny + 1 Kevin Harwell 1 Ivan Poddubny + 1 Alexander Traud 1 Edvin Vidmar + 1 Joshua C. Colp 1 Alexander Traud + 1 Gregory Massel + 1 Alexander Traud + + ---------------------------------------------------------------------- + + 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_srtp + + ASTERISK-29260: sRTP Replay Protection ignored; even tears down long calls + Reported by: Alexander Traud + * [3f4dfd5c02] 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 + * [17561b5e64] Ivan Poddubnyi -- res_pjsip_diversion: Fix adding more + than one histinfo to Supported + + Bug + + Category: Resources/res_pjsip + + ASTERISK-29196: res_pjsip: Segmentation fault + Reported by: Mauri de Souza Meneguzzo (3CPlus) + * [321632b02e] Joshua C. Colp -- pjsip: Make modify_local_offer2 + tolerate previous failed SDP. + + Category: Resources/res_pjsip_session + + ASTERISK-29203: res_pjsip_t38: Crash when changing state + Reported by: Gregory Massel + * [a5619097cd] Kevin Harwell -- AST-2021-002: Remote crash possible when + negotiating T.38 + + Category: Resources/res_pjsip_t38 + + ASTERISK-29203: res_pjsip_t38: Crash when changing state + Reported by: Gregory Massel + * [a5619097cd] Kevin Harwell -- AST-2021-002: Remote crash possible when + negotiating T.38 + + Category: Resources/res_rtp_asterisk + + ASTERISK-29205: res_rtp_asterisk: Asterisk crashes when making hold/unhold + from webrtc client + Reported by: Edvin Vidmar + * [4cea145aa9] Sean Bright -- res_rtp_asterisk.c: Fix signed mismatch + that leads to overflow + + ---------------------------------------------------------------------- + + 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. + + configs/samples/rtp.conf.sample | 12 +++++++ + doc/CHANGES-staging/srtp_replay_protection.txt | 9 +++++ + doc/UPGRADE-staging/srtp_replay_protection.txt | 9 +++++ + res/res_pjsip_diversion.c | 14 ++++++++ + res/res_pjsip_outbound_registration.c | 12 +++++++ + res/res_pjsip_path.c | 12 +++++++ + res/res_pjsip_session.c | 9 +++++ + res/res_pjsip_t38.c | 9 +++++ + res/res_rtp_asterisk.c | 16 +++++++--- + res/res_srtp.c | 5 +-- + third-party/pjproject/patches/0080-fix-sdp-neg-modify-local-offer.patch | 15 +++++++++ + 11 files changed, 115 insertions(+), 7 deletions(-)