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The Asterisk Development Team would like to announce security releases for
Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
These releases are available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk/releases
https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
The following security vulnerabilities were resolved in these versions:
* AST-2021-001: Remote crash in res_pjsip_diversion
If a registered user is tricked into dialing a
* AST-2021-002: Remote crash possible when negotiating T.38
When
* AST-2021-003: Remote attacker could prematurely tear down SRTP calls
An unauthenticated remote attacker could replay SRTP packets which could cause
an Asterisk instance configured without strict RTP validation to tear down
calls prematurely.
* AST-2021-004: An unsuspecting user could crash Asterisk with multiple
hold/unhold requests
Due to a signedness comparison mismatch, an authenticated WebRTC client could
cause a stack overflow and Asterisk crash by sending multiple hold/unhold
requests in quick succession.
* AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver
Given a scenario where an outgoing call is placed from Asterisk to a remote
SIP server it is possible for a crash to occur.
For a full list of changes in the current releases, please see the ChangeLogs:
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
The security advisories are available at:
https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
Thank you for your continued support of Asterisk!
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"tio" is a simple TTY terminal application which features a straightforward
commandline interface to easily connect to TTY devices for basic input/output.
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The problem is reported by Markus Kilbinger on port-arm mailing list.
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Changelog:
The following issues are resolved in this release:
Security bugs fixed in this release:
* [ASTERISK-29219] res_pjsip_diversion: Crash if Tel URI contains
History-Info
(Reported by Torrey Searle)
Bugs fixed in this release:
* [ASTERISK-29229] Stasis/messaging: text messages not dispatched to
all subscribers when using generic subscription
(Reported by Jean Aunis Prescom)
* [ASTERISK-29238] chan_sip: SDP: Offers without any enabled stream
are accepted.
(Reported by Alexander Traud)
* [ASTERISK-29237] chan_sip: SDP: m=video is parsed even when
disabled.
(Reported by Alexander Traud)
* [ASTERISK-29222] chan_sip: Hold/Resume an sRTP call on a video
enabled user-agent.
(Reported by Alexander Traud)
* [ASTERISK-29240] chan_pjsip: Incoming PJSIP calls set global
SIPDOMAIN instead of a channel variable
(Reported by Ivan Poddubny)
* [ASTERISK-27902] chan_pjsip isnt updating hangupcause on 4XX
responses
(Reported by George Joseph)
* [ASTERISK-28016] PJSIP sends duplicate 183 Progress responses
(Reported by Alex Hermann)
* [ASTERISK-28185] chan_pjsip: Subsequent same responses are not
stopped
(Reported by Julien)
* [ASTERISK-29230] pjsip: Asterisk goes crazy and massively spams
logfile if registration cant be send
(Reported by Michael Maier)
* [ASTERISK-29231] pjsip: SIGSEGV in CLI if no trunk is registered
(Reported by Michael Maier)
* [ASTERISK-29217] LOCK() can grant the same lock to multiple
channels spuriously
(Reported by Jaco Kroon)
* [ASTERISK-29201] Crash occurs when Transfer and execute Hangup
before the Transfer result
(Reported by Dan Cropp)
* [ASTERISK-28947] Segmentation fault in mixmonitor_ds_destroy
(Reported by Robert Sutton)
* [ASTERISK-29191] tel: URI in Diversion header causes crash
(Reported by Mikhail Ivanov)
* [ASTERISK-28883] Spyee information ist missing in ChanSpyStop AMI
Event
(Reported by Hendrik Wedhorn)
* [ASTERISK-29188] null media causing the Asterisk crash
(Reported by sungtae kim)
* [ASTERISK-29209] Debug messages printed by scope trace might be
missing newlines
(Reported by Alexander Traud)
* [ASTERISK-29024] pjsip: Route Header in Cancel request incorrectly
set
(Reported by Flole Systems)
* [ASTERISK-29211] res_musiconhold: Segfault on realtime music on
hold without entries
(Reported by Nathan Bruning)
* [ASTERISK-29022] Crash when manipulating PJSIP invite dlg ref
counts
(Reported by Sean Bright)
* [ASTERISK-29173] Media cache URL requests allow infinite redirects
(Reported by Sean Bright)
* [ASTERISK-29175] res_pjsip_stir_shaken: Fix module description
(Reported by Stanislav Abramenkov)
* [ASTERISK-29148] AST_MODULE_INFO no, MODULEINFO depend
(Reported by Alexander Traud)
* [ASTERISK-28798] chan_sip: TCP/TLS client without server.
(Reported by Alexander Traud)
* [ASTERISK-29165] res_pjsip: malformed header Accept-Encoding in
OPTIONS response
(Reported by Alexander Greiner-Baer)
* [ASTERISK-29161] Incorrect setup of recall channels
(Reported by Boris P. Korzun)
* [ASTERISK-29155] app_queue: Deadlock between queues container and
individual queues
(Reported by George Joseph)
Improvements made in this release:
* [ASTERISK-28549] Two repeated 183
(Reported by Gant Liu)
* [ASTERISK-29216] contrib: systemd asterisk service for centos8 or
other newer linux versions
(Reported by Mark Petersen)
* [ASTERISK-29143] res_http_media_cache: HTTP media cache stored
hardcoded in /tmp
(Reported by laszlovl)
* [ASTERISK-29118] VoiceMail() should have an option to play
greetings as Early Media
(Reported by Juan Carlos Castro y Castro)
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-----
The Asterisk Development Team would like to announce security
releases for Asterisk 13, 15 and 16. The available releases are
released as versions 13.28.1, 15.7.4 and 16.5.1.
These releases are available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk/releases
The following security vulnerabilities were resolved in these versions:
* AST-2019-004: Crash when negotiating for T.38 with a declined stream
When Asterisk sends a re-invite initiating T.38 faxing, and the endpoint
responds with a declined media stream a crash will then occur in Asterisk.
* AST-2019-005: Remote Crash Vulnerability in audio transcoding
When audio frames are given to the audio transcoding support in Asterisk the
number of samples are examined and as part of this a message is output to
indicate that no samples are present. A change was done to suppress this
message for a particular scenario in which the message was not relevant. This
change assumed that information about the origin of a frame will always exist
when in reality it may not.
For a full list of changes in the current releases, please see the ChangeLogs:
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.4
The security advisories are available at:
https://downloads.asterisk.org/pub/security/AST-2019-004.pdf
https://downloads.asterisk.org/pub/security/AST-2019-005.pdf
-----
The Asterisk Development Team would like to announce security
releases for Asterisk 13, 15 and 16, and Certified Asterisk 13.21.
The available releases are released as versions 13.27.1, 15.7.3,
16.4.1 and 13.21-cert4.
These releases are available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk/releases
The following security vulnerabilities were resolved in these versions:
* AST-2019-002: Remote crash vulnerability with MESSAGE messages
A specially crafted SIP in-dialog MESSAGE message can cause Asterisk to crash.
* AST-2019-003: Remote Crash Vulnerability in chan_sip channel driver
When T.38 faxing is done in Asterisk a T.38 reinvite may be sent to an
endpoint to switch it to T.38. If the endpoint responds with an improperly
formatted SDP answer including both a T.38 UDPTL stream and an audio or video
stream containing only codecs not allowed on the SIP peer or user a crash will
occur. The code incorrectly assumes that there will be at least one common
codec when T.38 is also in the SDP answer.
For a full list of changes in the current releases, please see the ChangeLogs:
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.3
The security advisories are available at:
https://downloads.asterisk.org/pub/security/AST-2019-002.pdf
https://downloads.asterisk.org/pub/security/AST-2019-003.pdf
-----
The Asterisk Development Team would like to announce security
releases for Asterisk 15 and 16. The available releases are released
as versions 15.7.2 and 16.2.1.
These releases are available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk/releases
The following security vulnerabilities were resolved in these versions:
* AST-2019-001: Remote crash vulnerability with SDP protocol violation
When Asterisk makes an outgoing call, a very specific SDP protocol violation
by the remote party can cause Asterisk to crash.
For a full list of changes in the current releases, please see the ChangeLogs:
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
The security advisory is available at:
https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-----
The Asterisk Development Team would like to announce the release
of Asterisk 15.7.1.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 15.7.1 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!
The following issue is resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-28222 - Regression: MWI polling no longer works
(Reported by abelbeck)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.7.1
-----
The Asterisk Development Team would like to announce the release
of Asterisk 15.7.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 15.7.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
Security bugs fixed in this release:
-----------------------------------
* ASTERISK-28127 - Buffer overflow for DNS SRV/NAPTR records
(Reported by Jan Hoffmann)
* ASTERISK-28013 - res_http_websocket: Crash when reading HTTP
Upgrade requests
(Reported by Sean Bright)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-28076 - bridging: Asterisk crashes when receiving an
empty realtime text frame
(Reported by Emmanuel BUU)
* ASTERISK-28084 - app_queue: QueueMemberStatus Event flooding AMI
(Reported by Andrej)
* ASTERISK-28077 - res_pjsip: improve realtime performance on
CLI 'pjsip show contacts'
(Reported by Alexei Gradinari)
* ASTERISK-27920 - app_queue: Queue member considered inuse
after immediately hanging up during dialing.
(Reported by Cao Minh Hiep)
* ASTERISK-26094 - stasis: Playing MOH to bridge with ARI does not work
(Reported by Cameron)
* ASTERISK-28065 - res_odbc: missing SQL error diagnostic
(Reported by Alexei Gradinari)
* ASTERISK-28057 - chan_sip: SipNotify via AMI behaves
differently to CLI
(Reported by Peter Katzmann)
* ASTERISK-28045 - configure script does not enforce libunbound2 version
(Reported by Samuel Galarneau)
* ASTERISK-28070 - testsuite: Sniffer assumes pjmedia will use
ports below 10000
(Reported by Joshua C. Colp)
* ASTERISK-27854 - rtp: Crash in off-nominal case where RTP
instance can't be set up
(Reported by Lei Fu)
* ASTERISK-28059 - PJSIP: Update bundled PJPROJECT to version 2.8
(Reported by Joshua C. Colp)
* ASTERISK-27121 - res_pjsip_mwi: Memory leak on reload
(Reported by Sergej Kasumovic)
* ASTERISK-28047 - chan_pjsip: Declined video stream is added
when no video codecs configured and session refresh with removed
video stream occurs
(Reported by Will)
* ASTERISK-28049 - res_pjproject build failure
(Reported by Jaco Kroon)
* ASTERISK-28034 - chan_sip unstable with TLS after asterisk
start or reloads
(Reported by David Hajek)
* ASTERISK-28029 - [patch] res_musiconhold : music on hold will
not start if previous hold just reached end of file
(Reported by Frederic LE FOLL)
* ASTERISK-28005 - channel.c: ARI ring only once
(Reported by Hajek Michal)
* ASTERISK-28032 - Realtime queuemembers are not updated during
retry phase
(Reported by lvl)
* ASTERISK-27988 - alembic: PJSIP
"mwi_subscribe_replaces_unsolicited" field is integer not boolean
(Reported by Joshua C. Colp)
* ASTERISK-28020 - res_pjsip_transport_websocket: Properly set
'received' for IPv6
(Reported by Sean Bright)
* ASTERISK-28022 - res_pjsip realtime: uri column in
ps_contacts table can be too short
(Reported by Florian Floimair)
Improvements made in this release:
-----------------------------------
* ASTERISK-28046 - Remove stale nonoptreq references
(Reported by Walter Doekes)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.7.0
-----
The Asterisk Development Team would like to announce security
releases for Asterisk 15 and 16. The available releases are released
as versions 15.6.2 and 16.0.1.
These releases are available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk/releases
The following security vulnerabilities were resolved in these versions:
There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
SRV or NAPTR response. The attacker???s request causes Asterisk to segfault
and crash.
For a full list of changes in the current releases, please see the ChangeLogs:
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
The security advisory is available at:
https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-----
The Asterisk Development Team would like to announce security
releases for Asterisk 13, 14 and 15, and Certified Asterisk 13.21.
The available releases are released as versions 13.23.1, 14.7.8,
15.6.1 and 13.21-cert3.
These releases are available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk/releases
The following security vulnerabilities were resolved in these versions:
* AST-2018-009: Remote crash vulnerability in HTTP websocket upgrade
There is a stack overflow vulnerability in the res_http_websocket.so module of
Asterisk that allows an attacker to crash Asterisk via a specially crafted
HTTP request to upgrade the connection to a websocket. The attacker???s
request causes Asterisk to run out of stack space and crash.
For a full list of changes in the current releases, please see the ChangeLogs:
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
The security advisory is available at:
https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-----
The Asterisk Development Team would like to announce the release
of Asterisk 15.6.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 15.6.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-28002 - When T.140 realtime text is negociated, a
lot of debug traces are generated
(Reported by Emmanuel BUU)
* ASTERISK-27881 - PBX calls via chan_sip TCP trunk now get
authentification error
(Reported by Ian Gilmour)
* ASTERISK-28011 - chan_sip: get_refer_info() attempted unlock
mutex 'peer' without owning it!
(Reported by Alec Davis)
* ASTERISK-27944 - res_pjsip_t38: Crash receiving 1xx responses
other than 100 before 200 for T.38 reINVITE
(Reported by Joshua Elson)
* ASTERISK-28007 - rtcp-mux is put in SDP answer regardless of offer
(Reported by Torrey Searle)
* ASTERISK-27398 - No joint capabilities with video and audio-only streams
(Reported by Benjamin Keith Ford)
* ASTERISK-27973 - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
(Reported by Valentin Safonov)
* ASTERISK-27997 - pjproject_bundled: Fix for Solaris builds.
Do not undef s_addr.
(Reported by Alexander Traud)
* ASTERISK-27999 - Wrong SRTP use status report
(Reported by Salah Ahmed)
* ASTERISK-28001 - res_pjsip_registrar: Improve performance of
inbound handling
(Reported by Joshua Colp)
* ASTERISK-27966 - pjsip: Race condition in 183 re transmission
can result in a deadlock
(Reported by Torrey Searle)
* ASTERISK-15331 - make menuselect fails due to undefined
symbols (initscr32, w32addch) in menuselect_curses.o
(Reported by Majdi Bsoul)
* ASTERISK-14935 - [regression] menuselect compilation failure
on Solaris 10
(Reported by Samuel Owens)
* ASTERISK-12382 - menuselect compilation failure on Solaris 10
/ gcc 3.4.3
(Reported by rleasure)
* ASTERISK-9107 - menuselect compilation failure on Solaris 10/gcc-4.1.1
(Reported by Bob Atkins)
* ASTERISK-27991 - BuildSystem: Enable Jansson in Solaris 11.
(Reported by Alexander Traud)
* ASTERISK-27548 - res_pjsip_endpoint_identifier_ip only
matches against "generic string" headers
(Reported by George Joseph)
* ASTERISK-27990 - res_rtp_asterisk: Requires OpenSSL in
Developer Mode.
(Reported by Alexander Traud)
* ASTERISK-27591 - Frack errors in stasis.c and memory leakage
(Reported by Siruja Maharjan)
* ASTERISK-27978 - res_pjsip: Change default transport
keepalive to preserve behavior
(Reported by Joshua Colp)
* ASTERISK-27968 - systemd: asterisk.service
(Reported by seanchann.zhou)
* ASTERISK-27880 - [patch] pjproject_bundled: Repair
./configure --with-ssl=PATH.
(Reported by Alexander Traud)
* ASTERISK-27810 - BASIC-RETRANS: Implement receive
(Reported by Benjamin Keith Ford)
* ASTERISK-27972 - res_sorcery_config: Allow object name based matching
(Reported by Joshua Colp)
* ASTERISK-25548 - stasis: Improve message type "Use of before
init/after destruction" error
(Reported by Joshua Colp)
* ASTERISK-27967 - srtp: rejecting short sdes lifetimes
incompatible with obihai ATAs
(Reported by Nick French)
* ASTERISK-27961 - res_pjsip: Spurious ERROR logging when
printing headers in sip_msg
(Reported by Nick French)
* ASTERISK-27563 - pjsip modules always get -O2 even when
DONT_OPTIMIZE is set
(Reported by George Joseph)
* ASTERISK-27957 - PJSIP proposes ICE candidates on answer even
if not in offer
(Reported by Torrey Searle)
* ASTERISK-27347 - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
(Reported by Alexander Traud)
* ASTERISK-27938 - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
(Reported by Alexander Traud)
* ASTERISK-27955 - res_pjsip_session: sdp group:BUNDLE
attribute truncated
(Reported by Kevin Harwell)
* ASTERISK-27956 - res_pjsip_pubsub: segfault in function publish_expire
(Reported by Alexei Gradinari)
* ASTERISK-27949 - res_pjsip_rfc3326: A lot of endpoints do not
correctly handle two Reason headers
(Reported by Ross Beer)
* ASTERISK-27763 - res_pjsip_session: Initial INVITE with
audio+fax results in 488 instead of declining stream
(Reported by Thiago Coutinho)
* ASTERISK-27657 - res_pjsip_t38: ATA fails with hangupcause
58(Bearer capability not available)
(Reported by Jared Hull)
* ASTERISK-27080 - res_pjsip_t38: Slow T.38 re-invite rejection
if remote leg has T.38 disabled
(Reported by Torrey Searle)
* ASTERISK-26686 - res_pjsip: Lock inversion in transport management
(Reported by Ross Beer)
* ASTERISK-27939 - [patch] bridge_softmix_binaural: Enable
FFTW3 in Solaris 11.
(Reported by Alexander Traud)
Improvements made in this release:
-----------------------------------
* ASTERISK-28006 - PJSIP: Missing
"party=calling"/"party=called" in Remote-Party-ID
(Reported by Eric Dantie)
* ASTERISK-27995 - pjproject_bundled: Find shared libraries in
root --with-ssl=PATH.
(Reported by Alexander Traud)
* ASTERISK-27993 - pjsip_wizard example gives wrong info about
unsupported SRV records
(Reported by Jonathan Harris)
* ASTERISK-27970 - res_rtp_asterisk: T.140 packets containing
backspace or end of line are merged with regular text and it
causes some UA to break
(Reported by Emmanuel BUU)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-----
The Asterisk Development Team would like to announce the release
of Asterisk 15.5.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 15.5.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
Security bugs fixed in this release:
-----------------------------------
* ASTERISK-27818 - Username bruteforce is possible when using
ACL with PJSIP
(Reported by John)
* ASTERISK-27807 - iostreams: Potential DoS when client
connection closed prematurely
(Reported by Sean Bright)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-27783 - res_pjsip_pubsub: apparent crash on shutdown
(Reported by Kevin Harwell)
* ASTERISK-27870 - app_confbridge: Conference bridge and
announcer channels are not removed if conference is ended as
soon as it starts
(Reported by Robert Mordec)
* ASTERISK-27943 - AMI: Action SendText needs to use the correct thread.
(Reported by Richard Mudgett)
* ASTERISK-27942 - res_pjsip_messaging doesn't accept
application/* content-types.
(Reported by George Joseph)
* ASTERISK-27909 - cdr: Deadlock with submit_scheduled_batch
and submit_unscheduled_batch
(Reported by Denis Lebedev)
* ASTERISK-27936 - res_pjsip_session doesn't update media when
a 200 comes in with a different port than a 183
(Reported by George Joseph)
* ASTERISK-26987 - pbx_dundi: Asterisk crashes when unloading
module pbx_dundi.so with dundi peers
(Reported by Kirsty Tyerman)
* ASTERISK-27933 - [patch] uuid: Enable UUID in Solaris 11.
(Reported by Alexander Traud)
* ASTERISK-27625 - channels: CHECK_BLOCKING is ineffective
(Reported by Corey Farrell)
* ASTERISK-27931 - [patch] BuildSystem: Enable ./configure in Solaris 11.
(Reported by Alexander Traud)
* ASTERISK-27926 - [patch] bootstrap.sh: find -maxdepth is not
POSIX compatible.
(Reported by Alexander Traud)
* ASTERISK-27903 - menuselect: GCC 8: restrict-qualified
parameter passed and aliased.
(Reported by Alexander Traud)
* ASTERISK-27914 - [patch] tests/test_utils: Repair ./configure
--with-ssl=PATH.
(Reported by Alexander Traud)
* ASTERISK-27705 - chan_iax2: Stops listening for traffic
(Reported by Kirsty Tyerman)
* ASTERISK-27908 - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
(Reported by Alexander Traud)
* ASTERISK-27905 - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
(Reported by Alexander Traud)
* ASTERISK-27888 - SQL fetch error on query which return 0 columns
(Reported by Alexei Gradinari)
* ASTERISK-27902 - chan_pjsip isn't updating hangupcause on 4XX responses
(Reported by George Joseph)
* ASTERISK-27901 - [patch] ooh323c: GCC 8: output truncated
before terminating nul.
(Reported by Alexander Traud)
* ASTERISK-27872 - res_pjsip: Modified qualify_frequency
doesn't effect until pjsip reload
(Reported by Alexei Gradinari)
* ASTERISK-27094 - res_fax: Deadlock when using Local channels
and fax gateway
(Reported by David Brillert)
* ASTERISK-27848 - rtp: DTMF Breaks With telephony-event/16000
(Reported by Dominic)
* ASTERISK-25261 - Manager events for MeetMe have incorrectly
documented key name 'Usernum' - should be 'User'
(Reported by Francois Blackburn)
* ASTERISK-27878 - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
(Reported by Alexander Traud)
* ASTERISK-27876 - [patch] tcptls: Allow OpenSSL configured with no-dh.
(Reported by Alexander Traud)
* ASTERISK-27874 - [patch] tcptls: Allow OpenSSL 1.1.x
configured with enable-ssl3-method no-deprecated.
(Reported by Alexander Traud)
* ASTERISK-27845 - Codec-Change Re-INVITE during DTMF can cause
marker bit error
(Reported by Torrey Searle)
* ASTERISK-27831 - res_rtp_asterisk: Add support for
abs-send-time RTP extension
(Reported by Joshua Colp)
* ASTERISK-27863 - config/ast_destroy_realtime_fields:
successful DELETE is treated as failed
(Reported by Alexei Gradinari)
* ASTERISK-27865 - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
(Reported by Alexander Traud)
* ASTERISK-27760 - Asterisk ODBC Voicemail Prompt storage fails
with recent MariaDB version.
(Reported by Nic Colledge)
* ASTERISK-27853 - Incorrect error reported when
leaving/retrieving a ODBC voicemail
(Reported by Nic Colledge)
* ASTERISK-27726 - chan_mobile: presents incorrect inbound
Caller-ID names
(Reported by Brian)
* ASTERISK-27861 - [patch] res_pjsip_endpoint_identifier_ip:
Unregister the module for headers.
(Reported by Alexander Traud)
* ASTERISK-27860 - [patch] res_pjsip: Register
pjsip_transport_management not externally but internally.
(Reported by Alexander Traud)
* ASTERISK-27852 - cli: "manager show settings" mislabels HTTP
timeout as being minutes.
(Reported by Corey Farrell)
* ASTERISK-27824 - Fix issues exposed by GCC 8
(Reported by George Joseph)
* ASTERISK-27850 - [patch] rtp_engine: Allow Media Formats with
add_static_payload(-1) on egress again.
(Reported by Alexander Traud)
* ASTERISK-27811 - [patch] sip_to_pjsip: Enable python3 compatibility.
(Reported by Alexander Traud)
* ASTERISK-27841 - digest over for manager (ami) over http
fails on too long uris
(Reported by Jaco Kroon)
* ASTERISK-26570 - Macro allows an infinite loop of dialplan
inclusion resulting in a crash
(Reported by Tzafrir Cohen)
* ASTERISK-27801 - Asterisk got stuck while enabling "ari set
debug all on"
(Reported by shaurya jain)
* ASTERISK-27795 - chan_sip: one way / no audio with srtp
(Reported by Florian Kaiser)
* ASTERISK-27800 - One way audio when calling from Asterisk(sip
trunk) to another number where both are connected to a SBC using
TLS+SRTP
(Reported by Artur Pires)
* ASTERISK-26806 - pjsip_options: rework to make more efficient
(Reported by Kevin Harwell)
* ASTERISK-27814 - translate: interpolated frames are not
passed through
(Reported by Kevin Harwell)
* ASTERISK-27812 - When the ooh323 debug is on there is no
ringing signal to incoming calls via H323 trunk.
(Reported by Dimos)
* ASTERISK-26893 - No "alert" or "progress" in chan_ooh323 if
debug is enabled only on the module
(Reported by Marco Giordani)
* ASTERISK-27639 - [patch] BuildSystem: Enable IMAP storage on
FreeBSD and DragonFly BSD.
(Reported by Alexander Traud)
* ASTERISK-27804 - bridge_softmix / app_confbridge: Add support
for combining REMB reports
(Reported by Joshua Colp)
* ASTERISK-27418 - app_confbridge: "core show profile bridge"
does not output "sfu" when video_mode is sfu
(Reported by Carlos Chavez)
* ASTERISK-27808 - [patch] chan_vpb: Avoid GNU old-style field
designator extension.
(Reported by Alexander Traud)
Improvements made in this release:
-----------------------------------
* ASTERISK-27929 - [patch] BuildSystem: Enable autotools in Solaris 11.
(Reported by Alexander Traud)
* ASTERISK-27752 - Ten seconds of silence after mp3 playback
(Reported by Sam Wierema)
* ASTERISK-27910 - [patch] res_rtp_asterisk: Allow OpenSSL
configured with no-deprecated.
(Reported by Alexander Traud)
* ASTERISK-27906 - [patch] res_crypto: Allow OpenSSL configured
with no-deprecated.
(Reported by Alexander Traud)
* ASTERISK-27877 - app_confbridge: Add talking indicator for
ConfBridgeList AMI response
(Reported by William McCall)
* ASTERISK-27873 - documentation: Error on wiki description of
Asterisk 13 "MeetmeMute" event
(Reported by Alessandro Polidori)
* ASTERISK-27846 - ast_coredumper: Fix OUTPUT directory
(Reported by Ted G)
* ASTERISK-27867 - [patch] libasteriskssl: Allow OpenSSL 1.0.2
configured with no-deprecated.
(Reported by Alexander Traud)
* ASTERISK-27796 - res_hep: Allow create_address to resolve a
provided hostname
(Reported by Sebastian Gutierrez)
* ASTERISK-27820 - [patch] Add DragonFly BSD.
(Reported by Alexander Traud)
* ASTERISK-27793 - cppcheck identifies redundant "if"
(Reported by Ilya Shipitsin)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
|
|
1.1.0 (2020-12-25)
* Fix build problem on Ruby 3.
|
|
|
|
|
|
0.4.4
Re-org of README, to put the most insteresting parts near the top.
Added Linux makefile targets and Windows powershell scripts to automate bootstrapping a development environment, and automate the process of testing wheels before they are uploaded to PyPI.
Use stdlib unittest.mock where available
Travis CI now also builds on arm64
Demo06 demonstrates existing cursor positioning feature
Fix OSC regex & handling to prevent hang or crash
Document enterprise support by Tidelift
|
|
|
|
|
|
upstream changes: security fixes and bug fixes
|
|
-----
The Asterisk Development Team would like to announce security releases for
Asterisk 13, 16, 17 and 18. The available releases are released as versions
13.38.1, 16.15.1, 17.9.1 and 18.1.1.
The following security vulnerabilities were resolved in these versions:
* AST-2020-003: Remote crash in res_pjsip_diversion
A crash can occur in Asterisk when a SIP message is received that has a
History-Info header, which contains a tel-uri.
* AST-2020-004: Remote crash in res_pjsip_diversion
A crash can occur in Asterisk when a SIP 181 response is received that has a
Diversion header, which contains a tel-uri.
For a full list of changes in the current releases, please see the ChangeLogs:
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
The security advisories are available at:
https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-----
The Asterisk Development Team would like to announce the release
of Asterisk 13.38.0.
The release of Asterisk 13.38.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
Security bugs fixed in this release:
-----------------------------------
* ASTERISK-29057 - pjsip: Crash on call rejection during high load
(Reported by Sandro Gauci)
Improvements made in this release:
-----------------------------------
* ASTERISK-29056 - Increase reg_server column size for
ps_contacts table realtime
(Reported by sungtae kim)
* ASTERISK-29055 - Create a Bridge with video_single mode
(Reported by sungtae kim)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-29013 - res_pjsip: Asterisk doesn't stop sending
invites (with auth) on 407 replies
(Reported by Sebastian Damm)
* ASTERISK-29108 - resource_endpoints.c : Memory leak if
endpoint not found
(Reported by Jean Aunis - Prescom)
* ASTERISK-29097 - res_pjsip_config_wizard: Crash when freeing
string when failing to add extension
(Reported by Vieri)
* ASTERISK-26424 - app_voicemail: Undocumented behavior from VMSayName
(Reported by Eric Smith)
* ASTERISK-29051 - res_pjsip_sdp_rtp: Does not set correct
values on RTP instance when "auto" DTMF is used
(Reported by Sebastian Damm)
* ASTERISK-28311 - dsp: ast_dsp_silence_noise_with_energy wrong
judgment of frame format
(Reported by ?????????)
* ASTERISK-24329 - Music On Hold announcement cuts intro of
music the first time it is played
(Reported by Thomas Frederiksen)
* ASTERISK-29081 - res_stasis: Add compare function for bridges
moh container
(Reported by Hajek Michal)
* ASTERISK-29085 - func_curl: Segmentation fault when using
CURL after setting httpheader CURLOPT
(Reported by P??ter Juh??sz)
* ASTERISK-28416 - Unable to get rtp codec payload code for slin
(Reported by Brian J. Murrell)
New Features made in this release:
-----------------------------------
* ASTERISK-29027 - Implement support for History-Info
(Reported by Torrey Searle)
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.38.0
-----
The Asterisk Development Team would like to announce security
releases for Asterisk 13, 16, 17 and 18, and Certified Asterisk
16.8. The available releases are released as versions 13.37.1,
16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
The following security vulnerabilities were resolved in these versions:
* AST-2020-001: Remote crash in res_pjsip_session
Upon receiving a new SIP Invite, Asterisk did not return the created dialog
locked or referenced.
* AST-2020-002: Outbound INVITE loop on challenge with different nonce.
If Asterisk is challenged on an outbound INVITE and the nonce is changed in
each response, Asterisk will continually send INVITEs in a loop. This causes
Asterisk to consume more and more memory since the transaction will never
terminate (even if the call is hung up), ultimately leading to a restart or
shutdown of Asterisk. Outbound authentication must be configured on the
endpoint for this to occur.
For a full list of changes in the current releases, please see the ChangeLogs:
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
The security advisories are available at:
https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-----
The Asterisk Development Team would like to announce the release
of Asterisk 13.37.0.
The release of Asterisk 13.37.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-25665 - Duplicate logging in queue log for EXITEMPTY events
(Reported by Ove Aursand)
* ASTERISK-29043 - app_queue: Leave empty sometimes not
recorded as abandoned
(Reported by Kfir Itzhak)
* ASTERISK-29042 - res_parking: Parker UUID is no longer copied
(Reported by Misha Vodsedalek)
* ASTERISK-29029 - Voicemail "pollmailboxes"-option not
working, bug in function handle_subscribe
(Reported by Karsten Wemheuer)
* ASTERISK-28878 - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16
(Reported by Joseph Ades)
* ASTERISK-29046 - pbx: Deadlock when doing a reload, while
simultaneously doing an ExtensionState on a pattern match hint
that ends up adding an extension
(Reported by Ramarajan)
* ASTERISK-29040 - res_speech: Assertion on format
(Reported by Nickolay V. Shmyrev)
* ASTERISK-29001 - chan_pjsip does not process or forward 181 responses
(Reported by Torrey Searle)
* ASTERISK-27273 - app_voicemail: When a voicemail is marked as
"Urgent", it is not sent by email/processed by the mailcmd command
(Reported by Leandro Dardini)
* ASTERISK-29033 - res_pjsip_session: Aggressively terminates
session on failed re-INVITE
(Reported by Joshua C. Colp)
* ASTERISK-28974 - res_rtp_asterisk: T.140 messages have
appended RTP string to each message block.
(Reported by Thomas Johnson)
Improvements made in this release:
-----------------------------------
* ASTERISK-29010 - Allow disabling of FollowMe prompt
(Reported by Dennis)
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.37.0
-----
The Asterisk Development Team would like to announce the release
of Asterisk 13.36.0.
The release of Asterisk 13.36.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-29042 - res_parking: Parker UUID is no longer copied
(Reported by Misha Vodsedalek)
* ASTERISK-29029 - Voicemail "pollmailboxes"-option not
working, bug in function handle_subscribe
(Reported by Karsten Wemheuer)
* ASTERISK-29046 - pbx: Deadlock when doing a reload, while
simultaneously doing an ExtensionState on a pattern match hint
that ends up adding an extension
(Reported by Ramarajan)
* ASTERISK-29011 - chan_sip: ToHost property not cleared on reload
(Reported by Dennis)
* ASTERISK-28987 - BridgeCreated ARI event shows wrong
video_mode info
(Reported by sungtae kim)
* ASTERISK-28927 - Asterisk crash in music on hold
(Reported by David Cunningham)
* ASTERISK-28973 - Malformed IP address in SDP of 2nd SIP timer
triggered INVITE when NAT is active (UDP transport with
external_media_address)
(Reported by Michael Neuhauser)
* ASTERISK-28995 - res_pjsip_registrar: Expires on statically
configured contacts is not correct
(Reported by tootai)
* ASTERISK-28978 - acl: named_acl rule misconfiguration results
in segfault on reading rule from realtime
(Reported by Andrew Yager)
* ASTERISK-28975 - res_http_websocket: Text payload data
doesn't necessary include trailing zero
(Reported by Nickolay V. Shmyrev)
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.36.0
-----
The Asterisk Development Team would like to announce the release
of Asterisk 13.35.0.
The release of Asterisk 13.35.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-28878 - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16
(Reported by Joseph Ades)
* ASTERISK-28965 - res_pjsip: Apply outbound proxy to static
contacts on AOR
(Reported by Joshua C. Colp)
* ASTERISK-28930 - ./configure --without-ssl build failure
(Reported by Jaco Kroon)
* ASTERISK-28957 - chan_sip: chan_sip does not process 400
response to an INVITE.
(Reported by Frederic LE FOLL)
* ASTERISK-28888 - res_corosync: causes asterisk crash in huge
distributed environment.
(Reported by Universit?? di Bologna - CESIA VoIP)
* ASTERISK-28955 - "setvar" doesn't work properly in
dahdi-channels.conf
(Reported by Marin Odrljin)
* ASTERISK-28942 - res_sorcery_memory_cache: Individual object
expiration behaves unexpectedly with full backend caching
(Reported by Joshua C. Colp)
* ASTERISK-28952 - Queue wrapuptime sometimes not respected
(based on stale lastcall time)
(Reported by Walter Doekes)
* ASTERISK-28950 - Stale code in app_queue to check untouched channel
(Reported by Walter Doekes)
* ASTERISK-28644 - Stale comment in app_queue about ring_entry exception
(Reported by Walter Doekes)
* ASTERISK-28923 - T.38 Segfaults in chan_pjsip_queryoption
(Reported by Yury Kirsanov)
* ASTERISK-28936 - res_pjsip: crash when dialing non-sip uri
(Reported by Walter Doekes)
* ASTERISK-28900 - res_fax: Double frame free when gateway in
use with off-nominal format usage
(Reported by Gregory Massel)
* ASTERISK-28929 - pjproject_bundled: Honor --without-pjproject.
(Reported by Alexander Traud)
* ASTERISK-28932 - res_pjsip_logger writing too big packets
(Reported by nappsoft)
* ASTERISK-28885 - res_rtp_asterisk: Simultaneous termination
and ICE complete can cause crash
(Reported by Josep B)
* ASTERISK-28921 - Wrong return value check for fwrite when
writing to pcap file
(Reported by nappsoft)
Improvements made in this release:
-----------------------------------
* ASTERISK-28959 - res_pjsip: Added option for disable rport
parameter set
(Reported by sungtae kim)
* ASTERISK-28958 - Continue reading string when ping received
by websocket
(Reported by Nickolay V. Shmyrev)
* ASTERISK-28945 - AMI SendText - add Content-Type parameter
(Reported by Kevin Harwell)
* ASTERISK-28949 - res_http_websocket: Add masking to websocket client
(Reported by Moises Silva)
* ASTERISK-28899 - Upgrade Asterisk to bundled pjproject 2.10
(Reported by Kevin Harwell)
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.35.0
-----
The Asterisk Development Team would like to announce the release
of Asterisk 13.34.0.
The release of Asterisk 13.34.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-28932 - res_pjsip_logger writing too big packets
(Reported by nappsoft)
* ASTERISK-28921 - Wrong return value check for fwrite when
writing to pcap file
(Reported by nappsoft)
* ASTERISK-28794 - res_pjsip: Crash when escaping during URI printing
(Reported by nappsoft)
* ASTERISK-28884 - x-ast-orig-host not filtered out from
request URI and To header
(Reported by nappsoft)
* ASTERISK-28898 - bridge_softmix: Conference bridge not
passing silent rtp packets
(Reported by Jonathan Hunter)
* ASTERISK-28904 - RTP ICE leaks the memory
(Reported by sungtae kim)
* ASTERISK-28854 - SIGSEGV when pjsip show history encounters
IPV6 address
(Reported by Roger James)
* ASTERISK-28797 - [patch] tcptls: Fix notice when TLS is
enabled but not configured.
(Reported by Alexander Traud)
* ASTERISK-28804 - [patch] app_osplookup.c: Avoid a format
truncation.
(Reported by Alexander Traud)
* ASTERISK-28776 - Non async-signal-safe syscalls used after
fork before exec
(Reported by nappsoft)
* ASTERISK-28829 - app_queue: leaking stasis subscription when
Redirecting call
(Reported by lvl)
* ASTERISK-25844 - app_queue: Ghost channels in "core show
channels" output
(Reported by Etienne Lessard)
* ASTERISK-22920 - Crash while Forwarding from TLS extension
with CHANNEL args secure_bridge_media and
secure_bridge_signaling
(Reported by Shlomi Gutman)
* ASTERISK-28859 - pjsip: Increase maximum candidate count
(Reported by Joshua C. Colp)
* ASTERISK-28852 - Unprotected access to nochecksums variable,
causes build failures
(Reported by Guido Falsi)
Improvements made in this release:
-----------------------------------
* ASTERISK-28895 - res_pjsip_logger: Add tons'o'functionality
(Reported by Joshua C. Colp)
* ASTERISK-28879 - pjproject has race conditions in it's build system
(Reported by Guido Falsi)
* ASTERISK-28866 - third-party/pjproject/configure.m4 contains bashisms
(Reported by Guido Falsi)
* ASTERISK-28832 - chan_mobile creates PCMA streams that make
some VoIP clients crash or not render received audio
(Reported by Peter Turczak)
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.34.0
-----
The Asterisk Development Team would like to announce the release
of Asterisk 13.33.0.
The release of Asterisk 13.33.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
Improvements made in this release:
-----------------------------------
* ASTERISK-28813 - func_volume: Allow decimal numbers as
parameter to improve granularity
(Reported by Jean Aunis - Prescom)
* ASTERISK-27946 - dial (API): Storage of dialed target uses
AST_MAX_EXTENSION when it shouldn't
(Reported by Joshua Elson)
* ASTERISK-28782 - Add support for Content-Disposition header
in multi-part INVITES
(Reported by Torrey Searle)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-28852 - Unprotected access to nochecksums variable,
causes build failures
(Reported by Guido Falsi)
* ASTERISK-28847 - ARI channels cuts the endpoint string over
80 characters
(Reported by sungtae kim)
* ASTERISK-28835 - IPv6 addresses in SDP incorrectly formatted
(Reported by Daniel Heckl)
* ASTERISK-28372 - Asterisk REPLY Wrong Contact header port (TCP)
(Reported by Anton Satskiy)
* ASTERISK-24428 - Document that Asterisk will use the default
SIP ports (5060 for TCP, 5061 for TLS) if the extern option
variants aren't used
(Reported by sstream)
* ASTERISK-28838 - AST_MODULE_INFO requires, MODULEINFO does
not mention
(Reported by Alexander Traud)
* ASTERISK-28837 - pjproject_bundled: Honor
--without-pjproject.
(Reported by Alexander Traud)
* ASTERISK-27195 - chan_sip: only sets ToS bits on UDP socket,
ignoring TCP and TLS sockets
(Reported by Joshua Roys)
* ASTERISK-28812 - First DTMF is not get
(Reported by Bernard Merindol)
* ASTERISK-28758 - pjsip startup errors when using "with-ssl"
configure option
(Reported by Patrick Wakano)
* ASTERISK-28824 - BuildSystem: Search for Python/C API when
possibly needed only.
(Reported by Alexander Traud)
* ASTERISK-27717 - [patch] BuildSystem: In NetBSD, the Python
Programming Language is python-2.7.
(Reported by Alexander Traud)
* ASTERISK-28798 - [patch] chan_sip: TCP/TLS client without server.
(Reported by Alexander Traud)
* ASTERISK-28817 - chan_pjsip: constant DTMF tone if RTP is not
setup yet
(Reported by Kevin Harwell)
* ASTERISK-28816 - [patch] BuildSystem: Remove doc/tex and
doc/pdf leftovers.
(Reported by Alexander Traud)
* ASTERISK-28818 - [patch] BuildSystem: Allow space in path.
(Reported by Alexander Traud)
* ASTERISK-28801 - [patch] stasis: Avoid always true warnings
with clang.
(Reported by Alexander Traud)
* ASTERISK-28796 - func_channel: cannot read fields exten,
context, userfield, channame from dialplan
(Reported by S??bastien Duthil)
* ASTERISK-28803 - [patch] chan_unistim: Avoid tautological
warnings with clang.
(Reported by Alexander Traud)
* ASTERISK-28808 - [patch] test_stasis: Avoid always true
warning with clang.
(Reported by Alexander Traud)
* ASTERISK-28056 - res_pjsip: Incorrect endpoint status after
endpoint synchronization for a specific AOR
(Reported by Jason Hord)
* ASTERISK-28789 - test_utils: incorrectly printing error
'declined to load'
(Reported by Alexander Traud)
* ASTERISK-28788 - func_aes: incorrectly printing error
'declined to load'
(Reported by Alexander Traud)
* ASTERISK-16676 - DAHDIRAS fails to properly initiate pppd
unless asterisk is running as root
(Reported by Jaco Kroon)
* ASTERISK-21205 - [patch] dundi_read_result crash due to
negative number
(Reported by Jaco Kroon)
* ASTERISK-28743 - Asterisk is crashing if the 200 OK with SDP
(Reported by sungtae kim)
* ASTERISK-28774 - chan_pjsip's rtptimeout is erroneously
triggered during direct-media (native_rtp) bridge
(Reported by Michael Neuhauser)
* ASTERISK-20325 - Comments in configs/func_odbc.conf.sample
are not consistent with examples. Missing examples.
(Reported by Olivier Krief)
* ASTERISK-28780 - app_mixmonitor: Memory leak due to race
condition between AMI MixMonitor and hangup
(Reported by Joshua C. Colp)
* ASTERISK-28773 - Incorrect Sender SSRC in RTCP when p2p rtp
bridge is active
(Reported by Torrey Searle)
* ASTERISK-28759 - A non negotiated rtp frame causes call
disconnection when there is a SSRC change
(Reported by Paulo Vicentini)
* ASTERISK-26711 - func_enum: ENUM code wrong case
(Reported by Vitold)
* ASTERISK-23407 - Fix the FSF address in the headers of lots
of pjproject files
(Reported by Jared Smith)
* ASTERISK-28769 - DTLS Handshake Fails to Occur if ice_support
is enabled but not used
(Reported by Torrey Searle)
* ASTERISK-19460 - [patch] Function TXTCIDNAME never actually
makes DNS calls and always returns an empty string
(Reported by George Joseph)
New Features made in this release:
-----------------------------------
* ASTERISK-6863 - [patch] allow Asterisk to set high ToS bits
as non-root on Linux
(Reported by Matt Addison)
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.33.0
-----
The Asterisk Development Team would like to announce the release
of Asterisk 13.32.0.
The release of Asterisk 13.32.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-28766 - PJSIP blind transfer not completed after
using Proceeding()
(Reported by lvl)
* ASTERISK-28685 - check_expr2: linking (when hardening) and
cross-compiling troubles
(Reported by Sebastian Kemper)
* ASTERISK-28755 - SIP/Stasis: SIP headers not transmitted in
the "variables" field
(Reported by Jean Aunis - Prescom)
* ASTERISK-28754 - ASTERISK-28738 Causes Audio Issue After Hold
(Reported by Ross Beer)
* ASTERISK-28716 - ICE: pjnath shouldn't wait for ICE to
complete before allowing sending
(Reported by Benjamin Keith Ford)
* ASTERISK-28697 - res_pjsip: Named ACL does not update on
reload if changed
(Reported by Timothy Vanderaerden)
* ASTERISK-28738 - Incorrect state machine used when
MOH_PASSTHRU is used
(Reported by Torrey Searle)
* ASTERISK-28735 - Realtime MoH Unknown format '' -- defaulting
to SLIN
(Reported by Ross Beer)
* ASTERISK-26955 - pjsip: SIP Packets with Via "received="
Containing IPv6 Address Delimited by "[]" Rejected
(Reported by Peter Sokolov)
* ASTERISK-28718 - chan_sip: Returns 403 if RTP ports are
depleted, should return 503
(Reported by Walter Doekes)
* ASTERISK-28719 - Cannot remove defaultrule from queue using
realtime queues
(Reported by EDV O-TON)
* ASTERISK-28714 - REGRESSION: Feature
subscription_persistence_recreate (ASTERISK-27759) Causes Segfaults
(Reported by Ross Beer)
* ASTERISK-26082 - res_pjsip_messaging: MessageSend
Content-Type can't be changed
(Reported by Alex)
* ASTERISK-28423 - ARI causes STASIS Deadlock
(Reported by Ross Beer)
* ASTERISK-28679 - stasis application is destroyed after its creation
(Reported by Francois Blackburn)
* ASTERISK-25421 - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in
spite of the error when sending
(Reported by Dmitriy Serov)
* ASTERISK-28139 - RTP Stream Incorrect Payload Type Causes
Asterisk To Drop Calls
(Reported by Paul Brooks)
* ASTERISK-28686 - chan_sip strictrtp=yes fails when media
source is changed: no audio
(Reported by Walter Doekes)
Improvements made in this release:
-----------------------------------
* ASTERISK-28750 - TLS/SSL Key too small error
(Reported by Martin Zeh)
* ASTERISK-24798 - Documentation - Clarify That Format Is Set
By File Name Extension In MixMonitor
(Reported by xrobau)
* ASTERISK-28726 - install_prereq script uses the interactive
mode when installing aptitude
(Reported by Sylvain Afchain)
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.32.0
-----
The Asterisk Development Team would like to announce the release
of Asterisk 13.31.0.
The release of Asterisk 13.31.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
New Features made in this release:
-----------------------------------
* ASTERISK-17491 - CURLOPT() needs a "followlocation" parameter
/ "maxredirs" doesn't do anything
(Reported by candrews)
* ASTERISK-28639 - res_pjsip_endpoint_identifier_ip: Add
ability to match on source port
(Reported by Sean Bright)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-28679 - stasis application is destroyed after its creation
(Reported by Francois Blackburn)
* ASTERISK-28423 - ARI causes STASIS Deadlock
(Reported by Ross Beer)
* ASTERISK-28714 - REGRESSION: Feature
subscription_persistence_recreate (ASTERISK-27759) Causes Segfaults
(Reported by Ross Beer)
* ASTERISK-28677 - CDR billsec is always 0 for transferred calls
(Reported by Maciej Michno)
* ASTERISK-28706 - silk 24hHz doesn't show up in 'core show
translation' output
(Reported by Sean Bright)
* ASTERISK-24484 - Update documentation for statsd module -
usage requirements unclear
(Reported by Dan Jenkins)
* ASTERISK-28702 - chan_dahdi: holding a channel via flash to
dialtone times out after 0:16:40
(Reported by Andrew Siplas)
* ASTERISK-28695 - core: minmemfree watermark uses free RAM,
not available RAM
(Reported by Kevin Flyn)
* ASTERISK-28693 - chan_sip: SIP MESSAGE beginning with a
whitespace appears empty in the dialplan
(Reported by Frank Matano)
* ASTERISK-23739 - [patch]Segfault forwarding voicemail with
ODBC storage enabled and realtime voicemail_data is used
(Reported by Stas Kobzar)
* ASTERISK-27622 - empty voicemail.conf required for ARA
(realtime) voicemail to leave message
(Reported by Jim Van Meggelen)
* ASTERISK-28349 - Pause reason not reported in QueueMember AMI event
(Reported by Niksa Baldun)
* ASTERISK-21794 - CLI command 'realtime update2' syntax
failure when using according to usage help
(Reported by Cedric BASSAGET)
* ASTERISK-25429 - res_pjsip_endpoint_identifier_ip: Document
support for hostnames
(Reported by Joshua C. Colp)
* ASTERISK-27775 - res_pjsip_notify: Multiple Event headers can
be present instead of just one
(Reported by AvayaXAsterisk)
* ASTERISK-28682 - app_record: Lack of `beep` audio file causes
application to return error and hangup
(Reported by Corey Farrell)
* ASTERISK-28507 - Wiki docs missing for MessageWaiting
(Reported by David M. Lee)
* ASTERISK-27759 - res_pjsip_pubsub: Subscription persistence
does not preserve XML <dialog-info> version number
(Reported by Bryan Nelson)
* ASTERISK-28605 - chan_dahdi: Deadlock in Hangup Scenarios
with concurrent command pri show span X
(Reported by Dirk Wendland)
* ASTERISK-28633 - stasis bridge topic leak
(Reported by Joeran Vinzens)
* ASTERISK-28492 - pjsip reload not reloading wizard
endpoint/pickup_group endpoint/call_group
(Reported by Jean-Denis Girard)
* ASTERISK-27243 - contrib: valgrind.supp doesn't suppress what
it's supposed to due to invalid syntax
(Reported by Richard Kenner)
* ASTERISK-28497 - func_odbc: truncating Unicode string on readsql
(Reported by Boris P. Korzun)
* ASTERISK-28647 - chan_sip: RTP frames not transmitted after
emitting a COLP
(Reported by Jean Aunis - Prescom)
* ASTERISK-28667 - Asterisk ignores parsing of config files if
a Byte order mark is present
(Reported by Robin Leffmann)
* ASTERISK-28664 - "trustrpid" is misspelled in
sip_to_pjsip.py
(Reported by Pascal Cadotte Michaud)
* ASTERISK-28663 - jansson: Support old versions
(Reported by Joshua C. Colp)
* ASTERISK-28636 - app_chanisavail+cdr: ChanIsAvail sometimes
fails to deactivate CDR.
(Reported by Frederic LE FOLL)
* ASTERISK-28604 - app_meetme, chan_ooh323 and cdr_mysql don't
build on 17.0.0
(Reported by George Joseph)
* ASTERISK-28660 - res_fax: wrap Asterisk initiated negotiation
with config option
(Reported by Kevin Harwell)
* ASTERISK-28628 - Debian 10.2: Warning when app_voicemail is compiling
(Reported by Stanislav Abramenkov)
* ASTERISK-28626 - Missing arguments in PJSIP_CONTACT function
documentation
(Reported by Pascal Cadotte Michaud)
* ASTERISK-28651 - chan_sip logs errors on tx to non-existent
TCP connections
(Reported by Jaco Kroon)
* ASTERISK-28502 - chan_pjsip incorrectly re-writes REGISTER
200 Response Contact
(Reported by Ross Beer)
Improvements made in this release:
-----------------------------------
* ASTERISK-28710 - Should be able to disable the /httpstatus
URI in the built-in HTTP server
(Reported by Sean Bright)
* ASTERISK-28638 - Simplify dialplan for Dial, Page, and ChanIsAvail
(Reported by cmaj)
* ASTERISK-28673 - GET FULL VARIABLE documentation clarification
(Reported by Jonathan Harris)
* ASTERISK-28658 - app_confbridge: Add support for setting
maximum sample rate
(Reported by Joshua C. Colp)
-----
The Asterisk Development Team would like to announce the release
of Asterisk 13.30.0.
The release of Asterisk 13.30.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
Security bugs fixed in this release:
-----------------------------------
* ASTERISK-28589 - chan_sip: Depending on configuration an
INVITE can alter Addr of a peer
(Reported by Andrey V. T.)
* ASTERISK-28580 - Bypass SYSTEM write permission in manager
action allows system commands execution
(Reported by Eliel Sarda½½ons)
Improvements made in this release:
-----------------------------------
* ASTERISK-28602 - res_pjsip_outbound_registration: Maximum
retries reached
(Reported by Daniel)
* ASTERISK-28586 - Typo in README-SERIOUSLY.bestpractices.md
(Reported by Sam Banks)
* ASTERISK-22192 - [patch] Allow voicemail forwards with ODBC
backend when format differs from attachfmt column
(Reported by cmaj)
* ASTERISK-28567 - Problem with ASTERISK-20207: Asterisk should
clear out any .lock files in the voice mail directory on startup.
(Reported by Michael)
* ASTERISK-28542 - [patch] add the ability for asterisk to
generate on-hold re-invites
(Reported by Torrey Searle)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-28663 - jansson: Support old versions
(Reported by Joshua C. Colp)
* ASTERISK-28604 - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
(Reported by George Joseph)
* ASTERISK-28641 - res_pjsip Segfaults when realtime
configuration to an AOR points to a not existent AOR
(Reported by Ross Beer)
* ASTERISK-28644 - Stale comment in app_queue about ring_entry exception
(Reported by Walter Doekes)
* ASTERISK-28637 - chan_sip+native_bridge_rtp: directmedia
compatibility check failure when negociated ptime is not default ptime.
(Reported by Frederic LE FOLL)
* ASTERISK-28445 - res_pjsip_session: ast_json_vpack: Invalid
UTF-8 string on hangup when TEST_FRAMEWORK enabled
(Reported by Bernhard Schmidt)
* ASTERISK-28631 - res_parking: Doesn't park when parkee and
parker are the same
(Reported by Ross Beer)
* ASTERISK-28612 - res_pjsip_t38: crash on reinvite with zero
port and no c= line
(Reported by Salah Ahmed)
* ASTERISK-28621 - Enforce T.38 error correction mode at 200 ok received
(Reported by Salah Ahmed)
* ASTERISK-28615 - chan_dahdi: PRI span status may stay "Down,
Active" after a short alarm
(Reported by Frederic LE FOLL)
* ASTERISK-28616 - parking: Deadlock when multi call parking
(Reported by Joshua C. Colp)
* ASTERISK-28423 - ARI causes STASIS Deadlock
(Reported by Ross Beer)
* ASTERISK-28608 - app_amd: Use time calculation to calculate timeout
(Reported by Michael Cargile)
* ASTERISK-28576 - res_rtp_asterisk: ICE Completion Crash when
sent packet length doesn't match
(Reported by Joshua Elson)
* ASTERISK-28618 - bridge_softmix: hold not cleared when
joining a softmix bridge
(Reported by Kevin Harwell)
* ASTERISK-26481 - FILE function grabs garbage along with read
data when target line has no newline
(Reported by Jonathan Harris)
* ASTERISK-28572 - Memory leaks in res_calendar_exchange and
res_calendar_icalendar
(Reported by Yoooooo Ha)
* ASTERISK-28585 - ari/resource_events: Crash in event session cleanup
(Reported by Kevin Harwell)
* ASTERISK-28590 - utils.c throws repeated warnings;
"pthread_attr_setstacksize: Invalid argument"
(Reported by Speed Dial Dave)
* ASTERISK-28578 - race condition on pjsip channelstats command
(Reported by Salah Ahmed)
* ASTERISK-28571 - cdr_pgsql: accesses obsolete (and finally
removed) column
(Reported by Christoph Moench-Tegeder)
* ASTERISK-28575 - MWI Send Notify Crash on 16.6
(Reported by Joshua Elson)
* ASTERISK-28574 - pjproject fails to build on 16.6.0, works on 16.5
(Reported by Niklas Larsson)
* ASTERISK-28561 - Asterisk Deadlocks
(Reported by Aheliotech)
* ASTERISK-28086 - chan_pjsip: Crash when initiating PlayDTMF over AMI
(Reported by Jeremiah Gadd)
* ASTERISK-28552 - res_pjsip_mwi: Frack during unload on
unsolicited_mwi container
(Reported by Kevin Harwell)
* ASTERISK-28566 - CDR backend unload problem during active call(s)
(Reported by Marian Piater)
* ASTERISK-28544 - Wrong contact representation in ipv6 mode
(Reported by J½½rgen H)
* ASTERISK-28534 - Segmentation fault when there is no priority
for an extension
(Reported by Timothy Vanderaerden)
* ASTERISK-28463 - res_pjsip_path: Crash when invalid contact
is configured
(Reported by Juan Martin)
* ASTERISK-28521 - pjsip: Memory Leak
(Reported by Mark)
* ASTERISK-28523 - Asterisk 16.5.0 Memory leak
(Reported by Cyril Rami½½re)
* ASTERISK-28538 - chan_pjsip: Deadlock on fax detection
(Reported by Joshua C. Colp)
* ASTERISK-28536 - Asterisk release candidates fail to build on FreeBSD
(Reported by Guido Falsi)
* ASTERISK-23756 - setvar directive when used in template and a
child of said template, results in duplicate variable names
(Reported by Michael Goryainov)
New Features made in this release:
-----------------------------------
* ASTERISK-28614 - app_senddtmf: Allow "receiving" DTMF with
PlayDTMF instead of only "sending"
(Reported by lvl)
* ASTERISK-28613 - func_curl: CURLOPT cannot set Content-Type header
(Reported by Martin Tomec)
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.30.0
-----
The Asterisk Development Team would like to announce security
releases for Asterisk 13, 16 and 17, and Certified Asterisk 13.21.
The available releases are released as versions 13.29.2, 16.6.2,
17.0.1 and 13.21-cert5.
The following security vulnerabilities were resolved in these versions:
* AST-2019-006: SIP request can change address of a SIP peer.
A SIP request can be sent to Asterisk that can change a SIP peer½½½s IP
address. A REGISTER does not need to occur, and calls can be hijacked as a
result. The only thing that needs to be known is the peer½½½s name;
authentication details such as passwords do not need to be known. This
vulnerability is only exploitable when the ½½½nat½½½ option is set to the
default, or ½½½auto_force_rport½½½.
* AST-2019-007: AMI user could execute system commands.
A remote authenticated Asterisk Manager Interface (AMI) user without
½½½system½½½ authorization could use a specially crafted ½½½Originate½½½ AMI
request to execute arbitrary system commands.
* AST-2019-008: Re-invite with T.38 and malformed SDP causes crash.
If Asterisk receives a re-invite initiating T.38 faxing and has a port of 0
and no c line in the SDP, a crash will occur.
For a full list of changes in the current releases, please see the ChangeLogs:
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
The security advisories are available at:
https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-----
The Asterisk Development Team would like to announce the release
of Asterisk 13.29.1.
The release of Asterisk 13.29.1 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-28574 - pjproject fails to build on 16.6.0, works on 16.5
(Reported by Niklas Larsson)
* ASTERISK-28575 - MWI Send Notify Crash on 16.6
(Reported by Joshua Elson)
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.29.1
-----
The Asterisk Development Team would like to announce the release
of Asterisk 13.29.0.
The release of Asterisk 13.29.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-28521 - pjsip: Memory Leak
(Reported by Mark)
* ASTERISK-28523 - Asterisk 16.5.0 Memory leak
(Reported by Cyril Rami½½re)
* ASTERISK-28538 - chan_pjsip: Deadlock on fax detection
(Reported by Joshua C. Colp)
* ASTERISK-28536 - Asterisk release candidates fail to build on FreeBSD
(Reported by Guido Falsi)
* ASTERISK-28527 - ChanIsAvail() creates a CDR if
unanswered=yes is set in cdr.conf
(Reported by Frederic LE FOLL)
* ASTERISK-28525 - chan_dahdi: set CHANNEL(hangupsource) when a
PRI channel hangs up
(Reported by Frederic LE FOLL)
* ASTERISK-28511 - codec_resample: Bad sound quality when up
sampling from SLIN16 to SLIN32
(Reported by Ruddy G)
* ASTERISK-28499 - translate: Crash when frame does not have a
"src" field set
(Reported by Gregory Massel)
* ASTERISK-25592 - chan_unistim: Clang Warning: variable sized
type not at end of a struct
(Reported by Alexander Traud)
* ASTERISK-28488 - pjsip mwi: n+1 sip notify's sent on re-register
(Reported by Chris Savinovich)
* ASTERISK-28509 - PJSIP cnonce generated on Linux contains 36
characters, NEC only supports up to 32 characters
(Reported by Dan Cropp)
* ASTERISK-28505 - app_voicemail/IMAP: segfault in
leave_voicemail because not checking mailstream
(Reported by Alexei Gradinari)
* ASTERISK-28487 - compile menuselect on gentoo
(Reported by Kilburn)
* ASTERISK-28472 - Asterisk occasionally passes a NULL as
srtp->session to srtp_protect/unprotect causing SEGV
(Reported by Jonas Swiatek)
* ASTERISK-28498 - cel / cdr: Event times may be incorrect
(Reported by Joshua C. Colp)
* ASTERISK-28483 - packet lost on UDPTL wrap around
(Reported by Torrey Searle)
* ASTERISK-28480 - json integer overflow in ssrc and timestamp
(Reported by Salah Ahmed)
* ASTERISK-28228 - res_pjsip: pjsip show contacts prints double entries
(Reported by Ian Jones)
* ASTERISK-28477 - Crash when not specifying "dbfile" in
res_config_sqlite3.conf
(Reported by Dennis)
* ASTERISK-28478 - Crash performing "core reload" with modified
res_config_sqlite3.conf
(Reported by Dennis)
* ASTERISK-28282 - AST_SCHED_REPLACE_UNREF causes wait-on-self
deadlocks (in chan_sip)
(Reported by Walter Doekes)
New Features made in this release:
-----------------------------------
* ASTERISK-17808 - [patch] Unregister a realtime moh class
(Reported by Byron Clark)
* ASTERISK-28489 - Channel variable SIPFROMDOMAIN for
chan_pjsip to setup From header URI domain
(Reported by Stas Kobzar)
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.29.0
-----
The Asterisk Development Team would like to announce security
releases for Asterisk 13, 15 and 16. The available releases are
released as versions 13.28.1, 15.7.4 and 16.5.1.
The following security vulnerabilities were resolved in these versions:
* AST-2019-004: Crash when negotiating for T.38 with a declined stream
When Asterisk sends a re-invite initiating T.38 faxing, and the endpoint
responds with a declined media stream a crash will then occur in Asterisk.
* AST-2019-005: Remote Crash Vulnerability in audio transcoding
When audio frames are given to the audio transcoding support in Asterisk the
number of samples are examined and as part of this a message is output to
indicate that no samples are present. A change was done to suppress this
message for a particular scenario in which the message was not relevant. This
change assumed that information about the origin of a frame will always exist
when in reality it may not.
For a full list of changes in the current releases, please see the ChangeLogs:
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.28.1
The security advisories are available at:
https://downloads.asterisk.org/pub/security/AST-2019-004.pdf
https://downloads.asterisk.org/pub/security/AST-2019-005.pdf
-----
The Asterisk Development Team would like to announce the release
of Asterisk 13.28.0
The release of Asterisk 13.28.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
Security bugs fixed in this release:
-----------------------------------
* ASTERISK-28447 - res_pjsip_messaging: In-dialog MESSAGE with
no body causes crash
(Reported by Gil Richard)
* ASTERISK-28465 - Broken SDP can cause a segfault in a T.38 reINVITE
(Reported by Francesco Castellano)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-28457 - [patch] Fix crash in chan_dahdi on 32-bit
systems caused by ASTERISK-28317
(Reported by abelbeck)
* ASTERISK-26006 - Show offending IP for TLS setup failures in logs
(Reported by Oleksandr Natalenko)
* ASTERISK-28444 - chan_pjsip: Peer IP for SSL handshake errors not logged
(Reported by Bernhard Schmidt)
* ASTERISK-28460 - res_pjsip_sdp_rtp: Fix ICE candidate leak
with specific usage
(Reported by Joshua C. Colp)
* ASTERISK-28018 - IP Fragmentation happening instead of DTLS
fragmentation on handshake server hello certificate
(Reported by vijay kumar)
* ASTERISK-25371 - Crash in hangup at chan_pjsip.c:1749 when
Asterisk attempts to generate hangup event
(Reported by Abhay Gupta)
* ASTERISK-28435 - cdr_pgsql: Unix socket doesn't work
(Reported by Dmitry Svyatogorov)
* ASTERISK-27981 - res_fax: Fax session leak with fax gatewaying
(Reported by pasandev)
* ASTERISK-28419 - app_amd: Does not work with silence suppression
(Reported by Nasir Iqbal)
* ASTERISK-28427 - new mwi.h include missing from some dahdi
source files, causes build failure
(Reported by Guido Falsi)
* ASTERISK-27994 - PJSIP: Early media ringback not indicated
after Progress()
(Reported by Gregory Massel)
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.28.0
-----
The Asterisk Development Team would like to announce security
releases for Asterisk 13, 15 and 16, and Certified Asterisk 13.21.
The available releases are released as versions 13.27.1, 15.7.3,
16.4.1 and 13.21-cert4.
The following security vulnerabilities were resolved in these versions:
* AST-2019-002: Remote crash vulnerability with MESSAGE messages
A specially crafted SIP in-dialog MESSAGE message can cause Asterisk to crash.
* AST-2019-003: Remote Crash Vulnerability in chan_sip channel driver
When T.38 faxing is done in Asterisk a T.38 reinvite may be sent to an
endpoint to switch it to T.38. If the endpoint responds with an improperly
formatted SDP answer including both a T.38 UDPTL stream and an audio or video
stream containing only codecs not allowed on the SIP peer or user a crash will
occur. The code incorrectly assumes that there will be at least one common
codec when T.38 is also in the SDP answer.
For a full list of changes in the current releases, please see the ChangeLogs:
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.27.1
The security advisories are available at:
https://downloads.asterisk.org/pub/security/AST-2019-002.pdf
https://downloads.asterisk.org/pub/security/AST-2019-003.pdf
-----
The Asterisk Development Team would like to announce the release
of Asterisk 13.27.0.
The release of Asterisk 13.27.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
New Features made in this release:
-----------------------------------
* ASTERISK-28375 - res_pjsip: New configuration setting to
allow disabling norefersub
(Reported by Dan Cropp)
* ASTERISK-28320 - Added ARI resource
/ari/channels/{channelid}/rtp_statistics
(Reported by sungtae kim)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-28427 - new mwi.h include missing from some dahdi
source files, causes build failure
(Reported by Guido Falsi)
* ASTERISK-28412 - GCC 9 catches more string formatting issues
(Reported by George Joseph)
* ASTERISK-28392 - The no-partial-inlining flag isn't passed to
the bundled pjproject or jansson builds
(Reported by George Joseph)
* ASTERISK-28402 - res_pjsip_registrar: SEGV in
registrar_find_contact
(Reported by Ross Beer)
* ASTERISK-28143 - app_amd: Infinite loop on silent calls
(Reported by Abhay Gupta)
* ASTERISK-28353 - stasis: Crash at shutdown when statistics enabled
(Reported by Joshua C. Colp)
* ASTERISK-28374 - latest asterisk unconditionally launch gcc
--version, even if the compiler is different
(Reported by Guido Falsi)
* ASTERISK-28391 - res_indications: Crash requesting
autocomplete on indications cli command
(Reported by Lucas Mendes)
* ASTERISK-27935 - app_voicemail: emailbody per user can't
contain commas
(Reported by S½½bastien Duthil)
* ASTERISK-17695 - 1.8.3.2 extenpatternmatchnew=yes cannot find
extensions with '-' in them
(Reported by test011)
* ASTERISK-17799 - AEL reload causes loss of control in a macro
(Reported by Kirill Katsnelson)
* ASTERISK-18593 - AEL for loops use Macro app and pipe delimiter
(Reported by Luke-Jr)
* ASTERISK-14939 - AEL parsers does not find existing label
(Reported by klaus3000)
* ASTERISK-20182 - Parsing a label beginning with a numeric
character in all Goto/GotoIf/GotoIfTime application causes
unexpected behavior
(Reported by Janu)
* ASTERISK-28348 - Failed to initialize OOH323 endpoint-OOH323 Disabled
(Reported by Dmitry Shubin)
* ASTERISK-28371 - chan_pjsip: DTMF Mode auto_info fallback
lead to both inband and info
(Reported by Salah Ahmed)
* ASTERISK-28362 - strtok_r() makes gcc compile warning
(Reported by sungtae kim)
Improvements made in this release:
-----------------------------------
* ASTERISK-28363 - Millisecond-resolution call stats including
PDD in channel variables
(Reported by Antoni Goldstein)
* ASTERISK-20207 - Asterisk should clear out any .lock files in
the voice mail directory on startup.
(Reported by Steven Wheeler)
* ASTERISK-28111 - build: CHANGES/UPGRADE are irritating to
work with.
(Reported by Corey Farrell)
* ASTERISK-28343 - Added app_name, app_data to channel type
(Reported by sungtae kim)
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.27.0
-----
The Asterisk Development Team would like to announce the release
of Asterisk 13.26.0.
The release of Asterisk 13.26.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
New Features made in this release:
-----------------------------------
* ASTERISK-28267 - res_stasis: Add ability to switch
applications
(Reported by Benjamin Keith Ford)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-20986 - QUEUE_MEMBER 's description is inaccurate
(Reported by Olivier Krief)
* ASTERISK-28350 - manager: Stasis backed up due to locking
(Reported by Joshua C. Colp)
* ASTERISK-25792 - chan_sip: qualifygap bounds checking
(Reported by Paul Sandys)
* ASTERISK-28341 - res_config_odbc eliminates empty custom (½½½@½½½
prefix) variables
(Reported by Alexei Gradinari)
* ASTERISK-28333 - StasisEnd event makes wrong timestamp value
(Reported by sungtae kim)
* ASTERISK-28306 - res_pjsip_mwi: MWI NOTIFY occasionally takes
minutes to be sent
(Reported by Jared Hull)
* ASTERISK-27964 - app_queue: ring_entry accesses nativeformats
without channel lock or reference
(Reported by Francisco Seratti)
* ASTERISK-28314 - ARI: API changed but "apiVersion" in
rest-api\resources.json did not
(Reported by Stefan Repke)
* ASTERISK-28335 - stasis: Make topic and maybe subscription
names unique and more useful
(Reported by Joshua C. Colp)
* ASTERISK-28321 - res_rtp_asterisk: Fixing possible divide by
zero for rtcp stat calculation
(Reported by sungtae kim)
* ASTERISK-28332 - Variable ALTCONF ignored when service is
used in Debian
(Reported by Cirillo Ferreira)
* ASTERISK-28322 - chan_pjsip: Add option to allow ignoring of
183 without SDP
(Reported by Torrey Searle)
* ASTERISK-28328 - MeetMe global non-admin mute is muting
admins that subsequently join
(Reported by Philip Mott)
* ASTERISK-28168 - app_queue: Adding a blank entry into sql
queue_members crashes asterisk.
(Reported by Michael)
* ASTERISK-28323 - pjsip: sip.conf to pjsip.conf conversion script fails
(Reported by Guido Weckwerth)
* ASTERISK-28272 - The basic-pbx config samples don't produce a
running asterisk
(Reported by George Joseph)
* ASTERISK-28312 - res_pjsip_diversion: Corrupted SIP Diversion
field after handling a 302 redirect
(Reported by Alex Odrov)
* ASTERISK-24173 - File menuselect/menuselect_gtk.c has no
license header
(Reported by Jeremy Lain½½)
* ASTERISK-28166 - app_voicemail: Asterisk unresponsive after
changing voicemail password with ODBC
(Reported by Michael)
* ASTERISK-28309 - res_pjsip: Wrong Contact and Via fields with
multiple UDP interfaces
(Reported by Nikolay shakin)
* ASTERISK-27992 - PJSIP: Adding `sends_registrations = yes` to
pjsip_wizard.conf causes crash
(Reported by Jonathan Harris)
* ASTERISK-28213 - res_pjsip: Threads pile up needlessly when
AOR is blocked
(Reported by Ross Beer)
* ASTERISK-28301 - Allow voicemail boxes to be subscribed to
with a presence event package
(Reported by George Joseph)
* ASTERISK-28303 - res_rtp_asterisk: Interaction between
smoother and DTMF can cause out of order timestamps
(Reported by Torrey Searle)
* ASTERISK-28302 - ARI: "Error destroying mutex" when listing
all ARI applications
(Reported by Stefan Repke)
* ASTERISK-28300 - AST_PBX_MAX_STACK is too low for some
applications
(Reported by George Joseph)
* ASTERISK-28106 - Astricon Feedback: Unable to filter ARI
events when GETting causes overload of events
(Reported by George Joseph)
* ASTERISK-28284 - switching between native_bridge and
simple_bridge can cause one way audio
(Reported by Torrey Searle)
* ASTERISK-28288 - Resources (udptl fd) leaking for T.38 calls
(Reported by Paulo Vicentini)
* ASTERISK-28251 - CI: Fix CI so it reverifies commit message changes
(Reported by George Joseph)
* ASTERISK-28277 - database: Add some basic logging
(Reported by Joshua C. Colp)
* ASTERISK-28181 - ari: Originating overwrites channel start time
(Reported by sungtae kim)
Improvements made in this release:
-----------------------------------
* ASTERISK-28326 - ari: Added timestamp for some ari events.
(Reported by sungtae kim)
* ASTERISK-28317 - Add logical group at DAHDIChannel event and
create "dahdi_group" at CHANNEL function
(Reported by Cirillo Ferreira)
* ASTERISK-28279 - Added creation timestamp for bridge
(Reported by sungtae kim)
* ASTERISK-28292 - Changed to show all channel stats including
wrong media
(Reported by sungtae kim)
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.26.0
-----
The Asterisk Development Team would like to announce the release
of Asterisk 13.25.0.
The release of Asterisk 13.25.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-28288 - Resources (udptl fd) leaking for T.38 calls
(Reported by Paulo Vicentini)
* ASTERISK-28213 - res_pjsip: Threads pile up needlessly when
AOR is blocked
(Reported by Ross Beer)
* ASTERISK-28271 - Opensuse Leap 15 --with-jannson-bundled will not compile
(Reported by David Wilcox)
* ASTERISK-28104 - AstriCon Feedback: Automatically create a 1
line dialplan context for stasis apps
(Reported by George Joseph)
* ASTERISK-28238 - PJSIP realtime. getcontext not working with DUNDI
(Reported by Ray)
* ASTERISK-28173 - Deadlock in chan_sip handling subscribe
request during res_parking reload
(Reported by Giuseppe Sucameli)
* ASTERISK-28263 - codec_opus: errors setting max_playback_rate
and bitrate to "sdp"
(Reported by Gianluca Merlo)
* ASTERISK-28250 - build: Cross-compilation fails for target
arm-linux-gnueabihf
(Reported by Jean Aunis - Prescom)
* ASTERISK-28156 - Race condition involving session->media
(res_pjsip_session) leads to crash.
(Reported by Paulo Vicentini)
* ASTERISK-28257 - res_http_websocket: PING / PONG opcodes
break data reception
(Reported by Jeremy Lain½½)
* ASTERISK-28252 - HangupHandler manager events are never thrown
(Reported by Gerald Schnabel)
* ASTERISK-28231 - res_http_websocket: Not responding to
Connection Close Frame (opcode 8)
(Reported by Jeremy Lain½½)
* ASTERISK-28249 - res_monitor: Segfault with
Monitor(wav,file,i)
(Reported by Valentin Vidi½½)
* ASTERISK-28244 - stasis: Filter messages at publishing to AMI/ARI
(Reported by Joshua C. Colp)
* ASTERISK-28197 - stasis: ast_endpoint struct holds the
channel_ids of channels past destruction in certain cases
(Reported by Mohit Dhiman)
* ASTERISK-28232 - core: RAII using clang use-after-scope issue
(Reported by Diederik de Groot)
* ASTERISK-28225 - app_voicemail: Channel variable
VM_MESSAGEFILE not updated correctly if message marked "urgent"
(Reported by boatright)
* ASTERISK-28212 - stasis: Statistics broke ABI under developer mode
(Reported by Joshua C. Colp)
* ASTERISK-28222 - Regression: MWI polling no longer works
(Reported by abelbeck)
* ASTERISK-28221 - Bug in ast_coredumper
(Reported by Andrew Nagy)
* ASTERISK-28162 - [patch] need to reset DTMF last sequence
number and timestamp on RTP renegotiation
(Reported by Alexei Gradinari)
* ASTERISK-28215 - app_voicemail: Leaving voicemail sometimes
doesn't trigger NOTIFYs
(Reported by George Joseph)
* ASTERISK-27959 - [patch] Asterisk 15.4.1 h264 fmtp
negotiation problem
(Reported by David Kuehling)
* ASTERISK-28117 - stasis: Add statistics for usage when in
developer mode
(Reported by Joshua C. Colp)
* ASTERISK-28201 - [patch] confbridge: no announce to the
marked users when they join an empty conference
(Reported by Alexei Gradinari)
* ASTERISK-28194 - chan_sip: Leak using contact ACL
(Reported by Giuseppe Sucameli)
* ASTERISK-28186 - stasis: Filter messages at publishing based
on to_* presence
(Reported by Joshua C. Colp)
* ASTERISK-27095 - chan_pjsip: When connected_line_method is
set to invite, we're not trying UPDATE
(Reported by George Joseph)
* ASTERISK-28182 - chan_pjsip: When connected_line_method is
set to invite, asterisk is not trying UPDATE
(Reported by nappsoft)
Improvements made in this release:
-----------------------------------
* ASTERISK-28246 - Support skipping on the g726 format
(Reported by Eyal Hasson)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.25.0
-----
The Asterisk Development Team would like to announce the release
of Asterisk 13.24.1.
The release of Asterisk 13.24.1 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!
The following issue is resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-28222 - Regression: MWI polling no longer works
(Reported by abelbeck)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.24.1
-----
The Asterisk Development Team would like to announce the release
of Asterisk 13.24.0.
The release of Asterisk 13.24.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
Security bugs fixed in this release:
-----------------------------------
* ASTERISK-28013 - res_http_websocket: Crash when reading HTTP
Upgrade requests
(Reported by Sean Bright)
New Features made in this release:
-----------------------------------
* ASTERISK-28087 - add flag to allow CALLERID(num) to be placed
in Contact header in chan_pjsip
(Reported by Torrey Searle)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-28125 - app_queue: Revert broken queue channel
reference patch
(Reported by lvl)
* ASTERISK-28151 - app_voicemail: MWI fails with
mailboxes=##@device instead of mailboxes=##@default
(Reported by Ronald Raikes)
* ASTERISK-28157 - Asterisk crashes when the res_pjsip_* modules unload
(Reported by sungtae kim)
* ASTERISK-28159 - SIGABRT caused by stack corruption in
hashkeys_read when no matching keys present
(Reported by Michael Walton)
* ASTERISK-28140 - repeated segmentation faults
(Reported by Eyal Hasson)
* ASTERISK-28103 - stasis: Filter messages at publishing to
reduce work done
(Reported by Joshua C. Colp)
* ASTERISK-28129 - Incorrect Behavior for rewrite_contact when
Re-Invite omits routset
(Reported by Torrey Searle)
* ASTERISK-28158 - Some conditions prevent running of el_end,
break the terminal.
(Reported by Corey Farrell)
* ASTERISK-28162 - [patch] need to reset DTMF last sequence
number and timestamp on voice packet with marker bit
(Reported by Alexei Gradinari)
* ASTERISK-28110 - rtp: Incorrect Packetization
(Reported by Robert Cripps)
* ASTERISK-28146 - pbx_config: Only the first [globals] section
is processed.
(Reported by Corey Farrell)
* ASTERISK-28150 - Formatting error in documentation
(Reported by Scott Griepentrog)
* ASTERISK-28081 - chan_sip: Asterisk 12+ chan_sip doesn't
report AST_CEL_PICKUP in handle_invite_replaces
(Reported by Luit van Drongelen)
* ASTERISK-28137 - res_pjsip_notify: improve realtime
performance on CLI completion on the endpoint
(Reported by Alexei Gradinari)
* ASTERISK-27980 - Caller ID cannot be changed on Attended
Transfer before dialing out
(Reported by Alexei Gradinari)
* ASTERISK-28089 - function ast_sendtext() create RTP realtime
packets with a trailing null byte in the payload
(Reported by Emmanuel BUU)
* ASTERISK-28076 - bridging: Asterisk crashes when receiving an
empty realtime text frame
(Reported by Emmanuel BUU)
* ASTERISK-28084 - app_queue: QueueMemberStatus Event flooding AMI
(Reported by Andrej)
* ASTERISK-28077 - res_pjsip: improve realtime performance on
CLI 'pjsip show contacts'
(Reported by Alexei Gradinari)
* ASTERISK-26094 - stasis: Playing MOH to bridge with ARI does not work
(Reported by Cameron)
* ASTERISK-27920 - app_queue: Queue member considered inuse
after immediately hanging up during dialing.
(Reported by Cao Minh Hiep)
* ASTERISK-28070 - testsuite: Sniffer assumes pjmedia will use
ports below 10000
(Reported by Joshua C. Colp)
* ASTERISK-28065 - res_odbc: missing SQL error diagnostic
(Reported by Alexei Gradinari)
* ASTERISK-27121 - res_pjsip_mwi: Memory leak on reload
(Reported by Sergej Kasumovic)
* ASTERISK-28059 - PJSIP: Update bundled PJPROJECT to version 2.8
(Reported by Joshua C. Colp)
* ASTERISK-28057 - chan_sip: SipNotify via AMI behaves differently to CLI
(Reported by Peter Katzmann)
* ASTERISK-28049 - res_pjproject build failure
(Reported by Jaco Kroon)
* ASTERISK-28029 - [patch] res_musiconhold : music on hold will
not start if previous hold just reached end of file
(Reported by Frederic LE FOLL)
* ASTERISK-28032 - Realtime queuemembers are not updated during retry phase
(Reported by lvl)
* ASTERISK-27988 - alembic: PJSIP
"mwi_subscribe_replaces_unsolicited" field is integer not boolean
(Reported by Joshua C. Colp)
* ASTERISK-28020 - res_pjsip_transport_websocket: Properly set
'received' for IPv6
(Reported by Sean Bright)
Improvements made in this release:
-----------------------------------
* ASTERISK-28144 - [patch] New function PJSIP_PARSE_URI to
parse an URI and return a specified part of the URI
(Reported by Alexei Gradinari)
* ASTERISK-28136 - Allow the sip_to_pjsip script to be used in a pipe
(Reported by Pascal Cadotte Michaud)
* ASTERISK-28046 - Remove stale nonoptreq references
(Reported by Walter Doekes)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.24.0
-----
The Asterisk Development Team would like to announce security
releases for Asterisk 13, 14 and 15, and Certified Asterisk 13.21.
The available releases are released as versions 13.23.1, 14.7.8,
15.6.1 and 13.21-cert3.
These releases are available for immediate download at
The following security vulnerabilities were resolved in these versions:
* AST-2018-009: Remote crash vulnerability in HTTP websocket upgrade
There is a stack overflow vulnerability in the res_http_websocket.so module of
Asterisk that allows an attacker to crash Asterisk via a specially crafted
HTTP request to upgrade the connection to a websocket. The attacker½½½s
request causes Asterisk to run out of stack space and crash.
For a full list of changes in the current releases, please see the ChangeLogs:
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
The security advisory is available at:
https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-----
The Asterisk Development Team would like to announce the release
of Asterisk 13.23.0.
The release of Asterisk 13.23.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-27881 - PBX calls via chan_sip TCP trunk now get
authentification error
(Reported by Ian Gilmour)
* ASTERISK-28022 - res_pjsip realtime: uri column in
ps_contacts table can be too short
(Reported by Florian Floimair)
* ASTERISK-28011 - chan_sip: get_refer_info() attempted unlock
mutex 'peer' without owning it!
(Reported by Alec Davis)
* ASTERISK-28002 - When T.140 realtime text is negociated, a
lot of debug traces are generated
(Reported by Emmanuel BUU)
* ASTERISK-27973 - app_queue: QUEUESTATUS = CONTINUE instead
LEAVEEMPTY
(Reported by Valentin Safonov)
* ASTERISK-28007 - rtcp-mux is put in SDP answer regardless of offer
(Reported by Torrey Searle)
* ASTERISK-27997 - pjproject_bundled: Fix for Solaris builds.
Do not undef s_addr.
(Reported by Alexander Traud)
* ASTERISK-28001 - res_pjsip_registrar: Improve performance of
inbound handling
(Reported by Joshua Colp)
* ASTERISK-27999 - Wrong SRTP use status report
(Reported by Salah Ahmed)
* ASTERISK-27966 - pjsip: Race condition in 183 re transmission
can result in a deadlock
(Reported by Torrey Searle)
* ASTERISK-15331 - make menuselect fails due to undefined
symbols (initscr32, w32addch) in menuselect_curses.o
(Reported by Majdi Bsoul)
* ASTERISK-14935 - [regression] menuselect compilation failure on Solaris 10
(Reported by Samuel Owens)
* ASTERISK-12382 - menuselect compilation failure on Solaris 10 / gcc 3.4.3
(Reported by rleasure)
* ASTERISK-9107 - menuselect compilation failure on Solaris 10/gcc-4.1.1
(Reported by Bob Atkins)
* ASTERISK-27991 - BuildSystem: Enable Jansson in Solaris 11.
(Reported by Alexander Traud)
* ASTERISK-27548 - res_pjsip_endpoint_identifier_ip only
matches against "generic string" headers
(Reported by George Joseph)
* ASTERISK-27990 - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
(Reported by Alexander Traud)
* ASTERISK-27591 - Frack errors in stasis.c and memory leakage
(Reported by Siruja Maharjan)
* ASTERISK-27978 - res_pjsip: Change default transport
keepalive to preserve behavior
(Reported by Joshua Colp)
* ASTERISK-27957 - PJSIP proposes ICE candidates on answer even
if not in offer
(Reported by Torrey Searle)
* ASTERISK-27880 - [patch] pjproject_bundled: Repair
./configure --with-ssl=PATH.
(Reported by Alexander Traud)
* ASTERISK-25548 - stasis: Improve message type "Use of before
init/after destruction" error
(Reported by Joshua Colp)
* ASTERISK-27972 - res_sorcery_config: Allow object name based matching
(Reported by Joshua Colp)
* ASTERISK-27967 - srtp: rejecting short sdes lifetimes
incompatible with obihai ATAs
(Reported by Nick French)
* ASTERISK-27961 - res_pjsip: Spurious ERROR logging when
printing headers in sip_msg
(Reported by Nick French)
* ASTERISK-27563 - pjsip modules always get -O2 even when
DONT_OPTIMIZE is set
(Reported by George Joseph)
* ASTERISK-27347 - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
(Reported by Alexander Traud)
* ASTERISK-27938 - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
(Reported by Alexander Traud)
* ASTERISK-27956 - res_pjsip_pubsub: segfault in function publish_expire
(Reported by Alexei Gradinari)
* ASTERISK-27949 - res_pjsip_rfc3326: A lot of endpoints do not
correctly handle two Reason headers
(Reported by Ross Beer)
* ASTERISK-27763 - res_pjsip_session: Initial INVITE with
audio+fax results in 488 instead of declining stream
(Reported by Thiago Coutinho)
* ASTERISK-27657 - res_pjsip_t38: ATA fails with hangupcause
58(Bearer capability not available)
(Reported by Jared Hull)
* ASTERISK-27080 - res_pjsip_t38: Slow T.38 re-invite rejection
if remote leg has T.38 disabled
(Reported by Torrey Searle)
* ASTERISK-26686 - res_pjsip: Lock inversion in transport management
(Reported by Ross Beer)
* ASTERISK-27944 - res_pjsip_t38: Crash receiving 1xx responses
other than 100 before 200 for T.38 reINVITE
(Reported by Joshua Elson)
Improvements made in this release:
-----------------------------------
* ASTERISK-28006 - PJSIP: Missing
"party=calling"/"party=called" in Remote-Party-ID
(Reported by Eric Dantie)
* ASTERISK-27995 - pjproject_bundled: Find shared libraries in
root --with-ssl=PATH.
(Reported by Alexander Traud)
* ASTERISK-27993 - pjsip_wizard example gives wrong info about
unsupported SRV records
(Reported by Jonathan Harris)
* ASTERISK-27970 - res_rtp_asterisk: T.140 packets containing
backspace or end of line are merged with regular text and it
causes some UA to break
(Reported by Emmanuel BUU)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.23.0
-----
The Asterisk Development Team would like to announce the release
of Asterisk 13.22.0.
The release of Asterisk 13.22.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
Security bugs fixed in this release:
-----------------------------------
* ASTERISK-27818 - Username bruteforce is possible when using
ACL with PJSIP
(Reported by John)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-27783 - res_pjsip_pubsub: apparent crash on shutdown
(Reported by Kevin Harwell)
* ASTERISK-27870 - app_confbridge: Conference bridge and
announcer channels are not removed if conference is ended as
soon as it starts
(Reported by Robert Mordec)
* ASTERISK-27909 - cdr: Deadlock with submit_scheduled_batch
and submit_unscheduled_batch
(Reported by Denis Lebedev)
* ASTERISK-26987 - pbx_dundi: Asterisk crashes when unloading
module pbx_dundi.so with dundi peers
(Reported by Kirsty Tyerman)
* ASTERISK-27943 - AMI: Action SendText needs to use the
correct thread.
(Reported by Richard Mudgett)
* ASTERISK-27942 - res_pjsip_messaging doesn't accept
application/* content-types.
(Reported by George Joseph)
* ASTERISK-27936 - res_pjsip_session doesn't update media when
a 200 comes in with a different port than a 183
(Reported by George Joseph)
* ASTERISK-27933 - [patch] uuid: Enable UUID in Solaris 11.
(Reported by Alexander Traud)
* ASTERISK-27625 - channels: CHECK_BLOCKING is ineffective
(Reported by Corey Farrell)
* ASTERISK-27931 - [patch] BuildSystem: Enable ./configure in Solaris 11.
(Reported by Alexander Traud)
* ASTERISK-27926 - [patch] bootstrap.sh: find -maxdepth is not
POSIX compatible.
(Reported by Alexander Traud)
* ASTERISK-27903 - menuselect: GCC 8: restrict-qualified
parameter passed and aliased.
(Reported by Alexander Traud)
* ASTERISK-27914 - [patch] tests/test_utils: Repair ./configure
--with-ssl=PATH.
(Reported by Alexander Traud)
* ASTERISK-27705 - chan_iax2: Stops listening for traffic
(Reported by Kirsty Tyerman)
* ASTERISK-27908 - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
(Reported by Alexander Traud)
* ASTERISK-27905 - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
(Reported by Alexander Traud)
* ASTERISK-27888 - SQL fetch error on query which return 0 columns
(Reported by Alexei Gradinari)
* ASTERISK-27902 - chan_pjsip isn't updating hangupcause on 4XX responses
(Reported by George Joseph)
* ASTERISK-27901 - [patch] ooh323c: GCC 8: output truncated
before terminating nul.
(Reported by Alexander Traud)
* ASTERISK-27094 - res_fax: Deadlock when using Local channels
and fax gateway
(Reported by David Brillert)
* ASTERISK-25261 - Manager events for MeetMe have incorrectly
documented key name 'Usernum' - should be 'User'
(Reported by Francois Blackburn)
* ASTERISK-27878 - [patch] tcptls.h: Repair ./configure
--with-ssl=PATH.
(Reported by Alexander Traud)
* ASTERISK-27872 - res_pjsip: Modified qualify_frequency
doesn't effect until pjsip reload
(Reported by Alexei Gradinari)
* ASTERISK-27876 - [patch] tcptls: Allow OpenSSL configured with no-dh.
(Reported by Alexander Traud)
* ASTERISK-27874 - [patch] tcptls: Allow OpenSSL 1.1.x
configured with enable-ssl3-method no-deprecated.
(Reported by Alexander Traud)
* ASTERISK-27845 - Codec-Change Re-INVITE during DTMF can cause
marker bit error
(Reported by Torrey Searle)
* ASTERISK-27863 - config/ast_destroy_realtime_fields:
successful DELETE is treated as failed
(Reported by Alexei Gradinari)
* ASTERISK-27865 - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
(Reported by Alexander Traud)
* ASTERISK-27853 - Incorrect error reported when
leaving/retrieving a ODBC voicemail
(Reported by Nic Colledge)
* ASTERISK-27726 - chan_mobile: presents incorrect inbound Caller-ID names
(Reported by Brian)
* ASTERISK-27861 - [patch] res_pjsip_endpoint_identifier_ip:
Unregister the module for headers.
(Reported by Alexander Traud)
* ASTERISK-27860 - [patch] res_pjsip: Register
pjsip_transport_management not externally but internally.
(Reported by Alexander Traud)
* ASTERISK-27760 - Asterisk ODBC Voicemail Prompt storage fails
with recent MariaDB version.
(Reported by Nic Colledge)
* ASTERISK-27852 - cli: "manager show settings" mislabels HTTP
timeout as being minutes.
(Reported by Corey Farrell)
* ASTERISK-27824 - Fix issues exposed by GCC 8
(Reported by George Joseph)
* ASTERISK-27811 - [patch] sip_to_pjsip: Enable python3 compatibility.
(Reported by Alexander Traud)
* ASTERISK-27841 - digest over for manager (ami) over http
fails on too long uris
(Reported by Jaco Kroon)
* ASTERISK-26570 - Macro allows an infinite loop of dialplan
inclusion resulting in a crash
(Reported by Tzafrir Cohen)
* ASTERISK-27801 - Asterisk got stuck while enabling "ari set debug all on"
(Reported by shaurya jain)
* ASTERISK-26806 - pjsip_options: rework to make more efficient
(Reported by Kevin Harwell)
* ASTERISK-27814 - translate: interpolated frames are not passed through
(Reported by Kevin Harwell)
* ASTERISK-27812 - When the ooh323 debug is on there is no
ringing signal to incoming calls via H323 trunk.
(Reported by Dimos)
* ASTERISK-26893 - No "alert" or "progress" in chan_ooh323 if
debug is enabled only on the module
(Reported by Marco Giordani)
* ASTERISK-27639 - [patch] BuildSystem: Enable IMAP storage on
FreeBSD and DragonFly BSD.
(Reported by Alexander Traud)
* ASTERISK-27808 - [patch] chan_vpb: Avoid GNU old-style field
designator extension.
(Reported by Alexander Traud)
Improvements made in this release:
-----------------------------------
* ASTERISK-27929 - [patch] BuildSystem: Enable autotools in Solaris 11.
(Reported by Alexander Traud)
* ASTERISK-27752 - Ten seconds of silence after mp3 playback
(Reported by Sam Wierema)
* ASTERISK-27910 - [patch] res_rtp_asterisk: Allow OpenSSL
configured with no-deprecated.
(Reported by Alexander Traud)
* ASTERISK-27906 - [patch] res_crypto: Allow OpenSSL configured
with no-deprecated.
(Reported by Alexander Traud)
* ASTERISK-27877 - app_confbridge: Add talking indicator for
ConfBridgeList AMI response
(Reported by William McCall)
* ASTERISK-27873 - documentation: Error on wiki description of
Asterisk 13 "MeetmeMute" event
(Reported by Alessandro Polidori)
* ASTERISK-27846 - ast_coredumper: Fix OUTPUT directory
(Reported by Ted G)
* ASTERISK-27867 - [patch] libasteriskssl: Allow OpenSSL 1.0.2
configured with no-deprecated.
(Reported by Alexander Traud)
* ASTERISK-27796 - res_hep: Allow create_address to resolve a
provided hostname
(Reported by Sebastian Gutierrez)
* ASTERISK-27820 - [patch] Add DragonFly BSD.
(Reported by Alexander Traud)
* ASTERISK-27793 - cppcheck identifies redundant "if"
(Reported by Ilya Shipitsin)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.22.0
-----
The Asterisk Development Team would like to announce security
releases for Asterisk 15, 13 and 14, and Certified Asterisk 13.18
and 13.21. The available releases are released as versions 15.4.1,
13.21.1, 14.7.7, 13.18-cert4 and 13.21-cert2.
The following security vulnerabilities were resolved in these versions:
* AST-2018-007: Infinite loop when reading iostreams
When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
sends a specially crafted message then Asterisk gets caught in an infinite
loop while trying to read the data stream. Thus rendering the system as
unusable.
* AST-2018-008: PJSIP endpoint presence disclosure when using ACL
When endpoint specific ACL rules block a SIP request they respond with a 403
forbidden. However, if an endpoint is not identified then a 401 unauthorized
response is sent. This vulnerability just discloses which requests hit a
defined endpoint. The ACL rules cannot be bypassed to gain access to the
disclosed endpoints.
For a full list of changes in the current releases, please see the ChangeLogs:
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
The security advisories are available at:
https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-----
The Asterisk Development Team would like to announce the release
of Asterisk 13.21.0.
The release of Asterisk 13.21.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
New Features made in this release:
-----------------------------------
* ASTERISK-27704 - Add cache_pools debug option to pjproject.conf
(Reported by Richard Mudgett)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-27809 - [patch] utils/pval: Add -lBlocksRuntime for
compiler clang conditionally.
(Reported by Alexander Traud)
* ASTERISK-27774 - res_musiconhold: Music on hold restarts
after every announcement
(Reported by lvl)
* ASTERISK-27782 - cdr_mysql: Missing MYSQL_PORT definition
(Reported by Evandro César Arruda)
* ASTERISK-27614 - res_pjsip_session: SDP origin does not use
resolved address
(Reported by John M.)
* ASTERISK-27740 - chan_sip: New Channel creation from new SIP
dialog with Replaces failed to be properly tracked and destroyed
(Reported by Shannon Price)
* ASTERISK-27706 - PJSIP: Deadlock shutting down subscription
TCP connection and sending subscription message.
(Reported by Ross Beer)
* ASTERISK-27435 - [patch] configure:
pjsip_evsub_set_uas_timeout not found.
(Reported by Alexander Traud)
* ASTERISK-27761 - [patch] BuildSystem: With external editline,
do not require libs for internal editline.
(Reported by Alexander Traud)
* ASTERISK-27755 - ConfBridge: raise ConfbridgeTalking when put
on hold and clear talking status
(Reported by Kevin Harwell)
* ASTERISK-27688 - res_pjsip: Crash on TCP PJSIP Transport Disconnect
(Reported by Ross Beer)
* ASTERISK-27743 - Generic PLC doesn't work if the 2 codecs on
a channel are equal
(Reported by George Joseph)
* ASTERISK-27745 - [patch] BuildSystem: Remove unused
dependency on libltdl.
(Reported by Alexander Traud)
* ASTERISK-12841 - [patch] Make format_ogg_vorbis work on OpenBSD
(Reported by Michiel van Baak)
* ASTERISK-27720 - [patch] BuildSystem: Enable Advanced Linux
Sound Architecture (ALSA) in NetBSD.
(Reported by Alexander Traud)
* ASTERISK-27741 - res_pjsip_rfc3326.c
rfc3326_use_reason_header doesn't account for more than one
'Reason' header
(Reported by Ross Beer)
* ASTERISK-27734 - [patch] BuildSystem: Enable IMAP storage on
openSUSE and Arch Linux.
(Reported by Alexander Traud)
* ASTERISK-27733 - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
(Reported by Alexander Traud)
* ASTERISK-11015 - NetBSD Build Needs RPATH set in 1.2.25
(Reported by Curt Sampson)
* ASTERISK-27641 - BuildSystem: Enable Better Backtraces in FreeBSD.
(Reported by Alexander Traud)
* ASTERISK-25586 - uuid_generate_random detection failure
(Reported by John Nemeth)
* ASTERISK-27721 - [patch] BuildSystem: Enable PortAudio in NetBSD.
(Reported by Alexander Traud)
* ASTERISK-27715 - [patch] BuildSystem: AC_PATH_PROG sets to
colon character when not found.
(Reported by Alexander Traud)
* ASTERISK-27703 - AMI Action VoicemailUsersList returns 0 MessageCount
(Reported by Sébastien Duthil)
* ASTERISK-27674 - chan_sip: RTP framing issues on outgoing calls
(Reported by Jean Aunis - Prescom)
* ASTERISK-27554 - res_pjsip_rfc3326: Order of 'Reason' headers
break many endpoints
(Reported by Ross Beer)
* ASTERISK-27718 - [patch] BuildSystem: Enable Lua in NetBSD.
(Reported by Alexander Traud)
* ASTERISK-27722 - [patch] BuildSystem: Depend not implicitly
but explicitly on external libraries.
(Reported by Alexander Traud)
* ASTERISK-27719 - [patch] res_http_post: Enable GMime in NetBSD.
(Reported by Alexander Traud)
* ASTERISK-27716 - [patch] BuildSystem: Enable autotools in NetBSD.
(Reported by Alexander Traud)
* ASTERISK-27714 - [patch] chan_unistim: NetBSD has an
incompatible struct in_pktinfo.
(Reported by Alexander Traud)
* ASTERISK-27713 - [patch] BuildSystem: Cast any intptr_t
explicitly to its proposed type.
(Reported by Alexander Traud)
* ASTERISK-27712 - [patch] BuildSystem: Detect whether
uselocale(.) is available.
(Reported by Alexander Traud)
* ASTERISK-27711 - [patch] BuildSystem: Avoid re-defining of
pthread_* on NetBSD.
(Reported by Alexander Traud)
* ASTERISK-27710 - [patch] BuildSystem: Install init scripts on
openSUSE Tumbleweed.
(Reported by Alexander Traud)
* ASTERISK-27709 - [patch] BuildSystem: Avoid == for comparison
in ./configure.
(Reported by Alexander Traud)
* ASTERISK-27610 - app_amd.so returning TOOLONG before reaching
the timeout
(Reported by Michael Cargile)
* ASTERISK-26688 - Documentation: voicemail.conf.sample shows
512 limit for emailbody field, however this is only true if
compiled with LOW_MEMORY option
(Reported by Fran Vicente)
* ASTERISK-27568 - PJSIP: Crash during SIP attended transfer.
(Reported by Bryan Walters)
* ASTERISK-27686 - [patch] install_prereq: Update FreeBSD libraries.
(Reported by Alexander Traud)
* ASTERISK-24488 - Wrong remote identity and target in dialog
package XML in NOTIFY
(Reported by Alejandro Padilla)
* ASTERISK-27646 - ICE fails with no candidate nominated
(Reported by Thomas Guebels)
* ASTERISK-27457 - chan_sip: Guests disallowed via TCP (or TLS)
if existing peer from same IP.
(Reported by Alexander Traud)
Improvements made in this release:
-----------------------------------
* ASTERISK-27697 - Enable in-dialog NOTIFY on chan_pjsip channels
(Reported by Nathan Bruning)
* ASTERISK-26540 - cdr_radius: use radcli instead of
freeradius-client
(Reported by Tzafrir Cohen)
* ASTERISK-27770 - [patch] install_prereq: Add Slackware (somehow).
(Reported by Alexander Traud)
* ASTERISK-27769 - [patch] install_prereq: Add Gentoo Linux.
(Reported by Alexander Traud)
* ASTERISK-27738 - [patch] install_prereq: Add Arch Linux.
(Reported by Alexander Traud)
* ASTERISK-27736 - [patch] install_prereq: Add SUSE.
(Reported by Alexander Traud)
* ASTERISK-26976 - libsrtp-2.x.x support
(Reported by Alex)
* ASTERISK-27728 - [patch] BuildSystem: Add NetBSD.
(Reported by Alexander Traud)
* ASTERISK-27730 - PJSIP: Update bundled PJPROJECT to version 2.7.2
(Reported by Richard Mudgett)
* ASTERISK-27729 - [patch] install_prereq: Add NetBSD.
(Reported by Alexander Traud)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.21.0
-----
The release of Asterisk 13.20.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
Security bugs fixed in this release:
-----------------------------------
* ASTERISK-27583 - Segmentation fault occurs in asterisk with
an invalid SDP fmtp attribute
(Reported by Sandro Gauci)
* ASTERISK-27582 - Segmentation fault occurs in Asterisk with
an invalid SDP media format description
(Reported by Sandro Gauci)
* ASTERISK-27618 - Crash occurs when sending a repeated number
of INVITE messages over TCP or TLS transport
(Reported by Sandro Gauci)
* ASTERISK-27640 - SUBSCRIBE message with a large Accept value
causes stack corruption
(Reported by Sandro Gauci)
New Features made in this release:
-----------------------------------
* ASTERISK-27117 - core: Add support for timelen parsing to
ast_parse_arg and ACO.
(Reported by Corey Farrell)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-27703 - AMI Action VoicemailUsersList returns 0
MessageCount
(Reported by Sébastien Duthil)
* ASTERISK-24386 - Asterisk "doc/lang/language-criteria.txt"
needs update or removal.
(Reported by Rusty Newton)
* ASTERISK-27689 - [patch] rtp_engine: Load format name / mime
type in uppercase again.
(Reported by Alexander Traud)
* ASTERISK-27679 - res_pjsip: Endpoint destruction does not
free DTLS configuration
(Reported by Mak Dee)
* ASTERISK-27684 - [patch] install_prereq: Update OpenBSD libraries.
(Reported by Alexander Traud)
* ASTERISK-27681 - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
(Reported by Alexander Traud)
* ASTERISK-27680 - [patch] res_calendar: Specialized calendars
depend on symbols of general calendar.
(Reported by Alexander Traud)
* ASTERISK-27677 - [patch] BuildSystem: Enable system provided
libedit on OpenBSD.
(Reported by Alexander Traud)
* ASTERISK-27670 - [patch] BuildSystem: Remove chan_h323 leftovers.
(Reported by Alexander Traud)
* ASTERISK-27595 - [patch] BuildSystem: Invoke ldconfig with previous paths.
(Reported by Alexander Traud)
* ASTERISK-27631 - [patch] BuildSystem: Do not warn when bash
is not installed.
(Reported by Alexander Traud)
* ASTERISK-27666 - chan_sip: Crash processing CANCEL request
(Reported by Leandro Dardini)
* ASTERISK-27584 - Internal pjproject build doesn't disable bcg729
(Reported by Stuart Henderson)
* ASTERISK-27669 - [patch] codecs: Add support for WebRTC iLBC 2.0.
(Reported by Alexander Traud)
* ASTERISK-27642 - [patch] backtrace: Avoid
-Wlogical-not-parentheses.
(Reported by Alexander Traud)
* ASTERISK-27555 - [patch] install_prereq: Update Debian/Ubuntu libraries.
(Reported by Alexander Traud)
* ASTERISK-27656 - CDR: Leaking channel snapshots allocated by
stasis_channel.c
(Reported by Kristijan Vrban)
* ASTERISK-27426 - chan_console: cannot read and write at the
same time with alsa backend
(Reported by Tzafrir Cohen)
* ASTERISK-27621 - (null) string tailing after AsyncAGIEnd AMI event
(Reported by sungtae kim)
* ASTERISK-27652 - Null pointer Crash in PJSIP MWI
(Reported by Joshua Elson)
* ASTERISK-27612 - Subscriptions Persist After Expiration and
TCP/TLS Disconnect
(Reported by Ross Beer)
* ASTERISK-27571 - res_pjsip: If SIP response is received
during shutdown a crash may occur
(Reported by Joshua Colp)
* ASTERISK-27637 - [patch] BuildSystem: Enable autotools in FreeBSD.
(Reported by Alexander Traud)
* ASTERISK-27635 - [patch] app_voicemail: Avoid always true
warnings with clang.
(Reported by Alexander Traud)
* ASTERISK-27599 - [patch] install_prereq: Update
RHEL/CentOS/Fedora libraries.
(Reported by Alexander Traud)
* ASTERISK-26563 - core: macOS devmode build fails: variable
'freeswap' set but not used
(Reported by David M. Lee)
* ASTERISK-27630 - [patch] editline: Avoid shifting a negative signed value.
(Reported by Alexander Traud)
* ASTERISK-16172 - Problems with siren14 codec; problems with
siren7 sound files.
(Reported by Steve Murphy)
* ASTERISK-16951 - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
(Reported by Stéphan Kochen)
* ASTERISK-27603 - [patch] install_prereq: Download latest Jansson.
(Reported by Alexander Traud)
* ASTERISK-27607 - [patch] res_config_mysql: Avoid the header mysql_version.h.
(Reported by Alexander Traud)
* ASTERISK-24598 - When running
./contrib/scripts/install_prereq install-unpackaged pjproject is
installed in wrong place
(Reported by PowerPBX)
* ASTERISK-27602 - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
(Reported by Alexander Traud)
* ASTERISK-27600 - [patch] BuildSystem: Allow make clean all again.
(Reported by Alexander Traud)
* ASTERISK-27598 - [patch] install_prereq: Support package manager DNF.
(Reported by Alexander Traud)
* ASTERISK-26596 - Placing call on hold temporarily locks up set
(Reported by Igor Goncharovsky)
* ASTERISK-27596 - [patch] BuildSystem: Use the detected name
for MD5 everywhere.
(Reported by Alexander Traud)
* ASTERISK-27594 - [patch] BuildSystem: Invoke install not in
GNU but POSIX style.
(Reported by Alexander Traud)
* ASTERISK-27593 - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
(Reported by Alexander Traud)
* ASTERISK-27592 - [patch] BuildSystem: Detect external library
Lua in version 5.3.
(Reported by Alexander Traud)
* ASTERISK-26832 - res_pjsip: Segfault when calling
pjsip_hdr_print_on in sip_msg.c:581
(Reported by Ross Beer)
* ASTERISK-27589 - [patch] BuildSystem: Avoid $EUID and use id -u instead.
(Reported by Alexander Traud)
* ASTERISK-27575 - menuselect : remove obsolete TRACE_FRAMES
compiler flag
(Reported by Jean Aunis - Prescom)
* ASTERISK-27576 - [patch] res_config_pgsql: Avoid typecasting
an int to unsigned char.
(Reported by Alexander Traud)
* ASTERISK-27560 - [patch] clang 5 does not know
-Wno-format-truncation
(Reported by Alexander Traud)
* ASTERISK-27578 - [patch] app_osplookup.c: Avoid a format truncation.
(Reported by Alexander Traud)
* ASTERISK-27577 - [patch] chan_ooh323: Avoid typecasting an
int to unsigned short.
(Reported by Alexander Traud)
* ASTERISK-27491 - res_pjsip_endpoint_identifier_ip only
matches against header if match by ip fails
(Reported by George Joseph)
* ASTERISK-27549 - [patch] translate: Avoid absolute value on
unsigned substraction.
(Reported by Alexander Traud)
* ASTERISK-27553 - [patch] res_curl: Avoid error message on unload.
(Reported by Alexander Traud)
* ASTERISK-27557 - [patch] clang 5.0: implicit conversion to
char changes value to negative.
(Reported by Alexander Traud)
* ASTERISK-27559 - [patch] editline: Avoid comparison between
pointer and zero character constant.
(Reported by Alexander Traud)
* ASTERISK-27558 - [patch] codec_gsm: Avoid shifting a negative signed value.
(Reported by Alexander Traud)
* ASTERISK-25329 - Asterisk configure fails on 'cannot find
ptlib-config', despite ptlib-config existing
(Reported by Rusty Newton)
* ASTERISK-27552 - [patch] chan_ooh323: Limit outgoinglimit to
positive values as intended.
(Reported by Alexander Traud)
* ASTERISK-27551 - [patch] ooh323cDriver: Fix typo in header guard.
(Reported by Alexander Traud)
* ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf.
(Reported by Alexander Traud)
* ASTERISK-27539 - 'cdr submit' fails: batch mode not enabled.
(Reported by Tzafrir Cohen)
* ASTERISK-27498 - ICE candidate parser - ICE foundation
parsing too short
(Reported by Michele Prà )
* ASTERISK-27366 - Asterisk Turkish Language Set Problem
(Reported by Halil Ä°brahim YILDIZ)
* ASTERISK-23133 - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
(Reported by Shane Mitchell)
* ASTERISK-27531 - Compiler optimizations can break module load sequence.
(Reported by abelbeck)
* ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
Contact crashes asterisk
(Reported by Ross Beer)
* ASTERISK-24198 - Typo's
(Reported by Walter Doekes)
* ASTERISK-27229 - bridge: Old channel video source not set to
NULL after unref
(Reported by Richard Kenner)
Improvements made in this release:
-----------------------------------
* ASTERISK-27683 - [patch] BuildSystem: Allow newer autotools on OpenBSD.
(Reported by Alexander Traud)
* ASTERISK-27651 - app_confbridge: Add Muted to ConfbridgeJoin
and channel snapshot headers to ConfbridgeList AMI events
(Reported by Richard Mudgett)
* ASTERISK-27647 - app_confbridge/bridge_softmix: When channel
muted report talking stopped if was talking.
(Reported by Richard Mudgett)
* ASTERISK-27084 - Reduce verbosity while loading PBX extensions.
(Reported by Ludovic Gasc (Eyepea))
* ASTERISK-24372 - [patch] Add config option to play a prompt
to the "winner" in app_followme
(Reported by Graham Mainwaring)
* ASTERISK-27461 - 3PCC patch for AMI "SIPnotify"
(Reported by Yasuhiko Kamata)
* ASTERISK-27348 - [patch]contrib/scripts: add a way to migrate
from chan_sip to chan_pjsip realtime
(Reported by Torrey Searle)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.20.0
-----
The Asterisk Development Team would like to announce security
releases for Asterisk 13, 14 and 15, and Certified Asterisk 13.18.
The available releases are released as versions 13.19.2, 14.7.6,
15.2.2 and 13.18-cert3.
The following security vulnerabilities were resolved in these versions:
* AST-2018-001: Crash when receiving unnegotiated dynamic payload
The RTP support in Asterisk maintains its own registry of dynamic codecs and
desired payload numbers. While an SDP negotiation may result in a codec using
a different payload number these desired ones are still stored internally.
When an RTP packet was received this registry would be consulted if the
payload number was not found in the negotiated SDP. This registry was
incorrectly consulted for all packets, even those which are dynamic. If the
payload number resulted in a codec of a different type than the RTP stream
(for example the payload number resulted in a video codec but the stream
carried audio) a crash could occur if no stream of that type had been
negotiated. This was due to the code incorrectly assuming that a stream of the
type would always exist.
* AST-2018-002: Crash when given an invalid SDP media format description
By crafting an SDP message with an invalid media format description Asterisk
crashes when using the pjsip channel driver because pjproject's sdp parsing
algorithm fails to catch the invalid media format description.
* AST-2018-003: Crash with an invalid SDP fmtp attribute
By crafting an SDP message body with an invalid fmtp attribute Asterisk
crashes when using the pjsip channel driver because pjproject's fmtp retrieval
function fails to check if fmtp value is empty (set empty if previously parsed
as invalid).
* AST-2018-004: Crash when receiving SUBSCRIBE request
When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
accepted formats present in the Accept headers of the request. This code did
not limit the number of headers it processed despite having a fixed limit of
32. If more than 32 Accept headers were present the code would write outside
of its memory and cause a crash.
* AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
A crash occurs when a number of authenticated INVITE messages are sent over
TCP or TLS and then the connection is suddenly closed. This issue leads to a
segmentation fault.
* AST-2018-006: WebSocket frames with 0 sized payload causes DoS
When reading a websocket, the length was not being checked. If a payload of
length 0 was read, it would result in a busy loop that waited for the
underlying connection to close.
For a full list of changes in the current releases, please see the ChangeLogs:
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
The security advisories are available at:
https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-----
The release of Asterisk 13.19.1 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!
The following issue is resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-27656 - CDR: Leaking channel snapshots allocated by
stasis_channel.c
(Reported by Kristijan Vrban)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.19.1
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Also avoid passing crazy optimization and debug flags in general, just
honor the user's CFLAGS.
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Upstream changes:
bugfixes
minor improvements
STIR/SHAKEN support
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For the Python 3.8 default switch.
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Version 3.5
Bugfixes:
- spy: ensure bytes in write()
Bugfixes (posix):
- serialposix: Fix inconsistent state after exception in open()
Bugfixes (win32):
- win32: Fix exception for composite serial number search on Windows
Bugfixes (MacOS):
- list_ports_osx: kIOMasterPortDefault no longer exported on Big Sur
- list_ports_osx: getting USB info on BigSur/AppleSilicon
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1.8.20
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I am talking to upstream about integrating patches, and about to
package an alpha in wip. This should be viewed as a soft
MAINTAINERship, but please ask me if you want to do anything
signficant to avoid duplicated effort.
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1.59 Mon Jun 15 08:17:54 CEST 2020
- Merged pull request #6 from ghciv6/fix_multi_instance_log
fixed log handling with multi instances and typo in close().
Thanks to @ghciv6 !
1.58
- Updated test suite a bit.
- Added the tests to the manifest.
- Got rid of indirect object syntax.
- Moved test.pl to the actual test suite.
- Updated $VERSION declarations according to:
http://www.dagolden.com/index.php/369/version-numbers-should-be-boring/
- Added some extra tests (xt/author, xt/release).
- Fixed some spelling.
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Changelog:
Bugs fixed in this release:
-----------------------------------
[ASTERISK-28878] -
chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16
(Reported by Joseph Ades)
[ASTERISK-28965] -
res_pjsip: Apply outbound proxy to static contacts on AOR
(Reported by Joshua C. Colp)
[ASTERISK-28930] -
./configure --without-ssl build failure
(Reported by Jaco Kroon)
[ASTERISK-28886] -
chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
(Reported by Jared Smith)
[ASTERISK-28957] -
chan_sip: chan_sip does not process 400 response to an INVITE.
(Reported by Frederic LE FOLL)
[ASTERISK-28888] -
res_corosync: causes asterisk crash in huge distributed environment.
(Reported by Università di Bologna - CESIA VoIP)
[ASTERISK-28955] -
"setvar" doesn't work properly in dahdi-channels.conf
(Reported by Marin Odrljin)
[ASTERISK-28954] -
StreamEcho() only returns 1 active stream
(Reported by Bill Kervaski)
[ASTERISK-28942] -
res_sorcery_memory_cache: Individual object expiration behaves unexpectedly with full backend caching
(Reported by Joshua C. Colp)
[ASTERISK-28953] -
res_pjsip_session: Preserve stream label
(Reported by Joshua C. Colp)
[ASTERISK-28952] -
Queue wrapuptime sometimes not respected (based on stale lastcall time)
(Reported by Walter Doekes)
[ASTERISK-28950] -
Stale code in app_queue to check untouched channel
(Reported by Walter Doekes)
[ASTERISK-28644] -
Stale comment in app_queue about ring_entry exception
(Reported by Walter Doekes)
[ASTERISK-28948] -
ARI channel create doesn't referencing the channel_id parameter
(Reported by sungtae kim)
[ASTERISK-28938] -
core_unreal / core_local: Add support for multistream and re-negotiation
(Reported by Joshua C. Colp)
[ASTERISK-28939] -
res_rtp_asterisk: Don't have send/receive buffers on non-WebRTC
(Reported by Joshua C. Colp)
[ASTERISK-28944] -
bridge_softmix: Transitioning a stream from inactive -> sendrecv/sendonly doesn't re-negotiation
(Reported by Joshua C. Colp)
[ASTERISK-28923] -
T.38 Segfaults in chan_pjsip_queryoption
(Reported by Yury Kirsanov)
[ASTERISK-28940] -
/channels/create doesn't get any parameters from the body
(Reported by sungtae kim)
[ASTERISK-28936] -
res_pjsip: crash when dialing non-sip uri
(Reported by Walter Doekes)
[ASTERISK-28900] -
res_fax: Double frame free when gateway in use with off-nominal format usage
(Reported by Gregory Massel)
[ASTERISK-28929] -
pjproject_bundled: Honor --without-pjproject.
(Reported by Alexander Traud)
[ASTERISK-28932] -
res_pjsip_logger writing too big packets
(Reported by nappsoft)
[ASTERISK-28921] -
Wrong return value check for fwrite when writing to pcap file
(Reported by nappsoft)
Improvements made in this release:
-----------------------------------
[ASTERISK-28959] -
res_pjsip: Added option for disable rport parameter set
(Reported by sungtae kim)
[ASTERISK-28958] -
Continue reading string when ping received by websocket
(Reported by Nickolay V. Shmyrev)
[ASTERISK-28945] -
AMI SendText - add Content-Type parameter
(Reported by Kevin Harwell)
[ASTERISK-28949] -
res_http_websocket: Add masking to websocket client
(Reported by Moises Silva)
[ASTERISK-28899] -
Upgrade Asterisk to bundled pjproject 2.10
(Reported by Kevin Harwell)
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Fix taken from the upstream project's 9.0.305 Alpha.01 release, noted to
be a temporary workaround. (Separately, from how I read the change log,
there has been no stable 9.0 release since 9.0.302.) Tested on Debian
9.13 (which has an older version of glibc which wouldn't reproduce the
issue) and Fedora 31 & 32.
(This issue was reported on pkgsrc-users back in July 2019 by Pierre
Dupond, and I'd provided a workaround for it in that email chain, but
I'd never actually committed anything to pkgsrc.)
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Version 2.8
Features
esptool.py image_info now prints a summary of segment memory types (IRAM, DRAM, etc) based on the address range.
esptool.py write_flash will warn if it looks like a bootloader binary is built for ESP32-S2 or another newer chip (support for flashing ESP32-S2 will be added in a future version.)
Bug Fixes
Removed ESP8266 SDK & ESP-IDF dependencies when building the flasher stub binaries. Previously the SDKs were used to include some register address macros, only. This removes any uncertainty about whether the flasher stub binary is a derived work of either SDK. The flasher stub binary itself is the same as the binary in v2.7.
Fixed minor issues running esptool automated tests on macOS.
Minor flake8 fixes including compatibility with newer flake8 versions.
ESP32 Only
Features
Support detection of new ESP32 silicon revisions
New esptool.py elf2image --min-rev X option allows creating a .bin file which only supports a minimum ESP32 silicon revision.
Bugfixes
Fix burning custom MAC with espefuse.py when 3/4 Coding Scheme is set
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Changelog:
Bugs fixed in this release:
-----------------------------------
[ASTERISK-28940] -
/channels/create doesn't get any parameters from the body
(Reported by sungtae kim)
[ASTERISK-28932] -
res_pjsip_logger writing too big packets
(Reported by nappsoft)
[ASTERISK-28921] -
Wrong return value check for fwrite when writing to pcap file
(Reported by nappsoft)
[ASTERISK-28794] -
res_pjsip: Crash when escaping during URI printing
(Reported by nappsoft)
[ASTERISK-28884] -
x-ast-orig-host not filtered out from request URI and To header
(Reported by nappsoft)
[ASTERISK-28871] -
res_pjsip_session: Unnecessary re-Invite on call answer
(Reported by Alexei Gradinari)
[ASTERISK-28903] -
res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
(Reported by Alexander Traud)
[ASTERISK-28898] -
bridge_softmix: Conference bridge not passing silent rtp packets
(Reported by Jonathan Hunter)
[ASTERISK-28892] -
res_musiconhold: Module res_musiconhold throws false warning
(Reported by Nicholas John Koch)
[ASTERISK-28904] -
RTP ICE leaks the memory
(Reported by sungtae kim)
[ASTERISK-26780] -
res_pjsip: PJSIP Registration Fails when transport=transport-udp6
(Reported by Peter Sokolov)
[ASTERISK-28854] -
SIGSEGV when pjsip show history encounters IPV6 address
(Reported by Roger James)
[ASTERISK-28804] -
[patch] app_osplookup.c: Avoid a format truncation.
(Reported by Alexander Traud)
[ASTERISK-28797] -
[patch] tcptls: Fix notice when TLS is enabled but not configured.
(Reported by Alexander Traud)
[ASTERISK-28776] -
Non async-signal-safe syscalls used after fork before exec
(Reported by nappsoft)
[ASTERISK-28870] -
streams: One memory leak and one issue cloning streams
(Reported by George Joseph)
[ASTERISK-28829] -
app_queue: leaking stasis subscription when Redirecting call
(Reported by lvl)
[ASTERISK-25844] -
app_queue: Ghost channels in "core show channels" output
(Reported by Etienne Lessard)
[ASTERISK-22920] -
Crash while Forwarding from TLS extension with CHANNEL args secure_bridge_media and secure_bridge_signaling
(Reported by Shlomi Gutman)
[ASTERISK-28859] -
pjsip: Increase maximum candidate count
(Reported by Joshua C. Colp)
[ASTERISK-28852] -
Unprotected access to nochecksums variable, causes build failures
(Reported by Guido Falsi)
[ASTERISK-28848] -
app_fax: Compile.
(Reported by Alexander Traud)
Improvements made in this release:
-----------------------------------
[ASTERISK-28895] -
res_pjsip_logger: Add tons'o'functionality
(Reported by Joshua C. Colp)
[ASTERISK-28896] -
ari: Add support for specifying variables on channel create
(Reported by Joshua C. Colp)
[ASTERISK-28879] -
pjproject has race conditions in it's build system
(Reported by Guido Falsi)
[ASTERISK-28866] -
third-party/pjproject/configure.m4 contains bashisms
(Reported by Guido Falsi)
[ASTERISK-28853] -
Missing include on FreeBSD
(Reported by Guido Falsi)
[ASTERISK-28832] -
chan_mobile creates PCMA streams that make some VoIP clients crash or not render received audio
(Reported by Peter Turczak)
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Changelog:
Version 3.2.15 (3rd June 2020)
--------------
Fix build for gcc-10 (efax/efaxlib.h, efax/efaxlib.c,
efax/Makefile.am, efax/Makefile.in).
Version 3.2.14 (6th March 2020)
--------------
Remove X11 specific code to allow the program to run better
against wayland compositors (acinclude.m4, configure.ac;
dialogs.cpp, helpfile.cpp, logger.cpp, main.cpp, mainwindow.cpp,
prog_defs.h; src/Makefile.am).
Fix label layout in settings dialog (settings.cpp).
Apply SO_REUSEADDR option when constructing sockets
(socket_server.cpp).
Deal with strict aliasing warning (efax/efaxos.c).
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This option has been removed in 2018, see ChangeLog.
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These packages are susceptible to bugs when confronted with non-ASCII
characters.
See https://gcc.gnu.org/bugzilla/show_bug.cgi?id=94182.
It takes some time to analyze and fix these individually, therefore they
are only marked as "needs work".
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revision.
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asterisk 14.7.8:
* AST-2018-009: Fix crash processing websocket HTTP Upgrade requests
The HTTP request processing in res_http_websocket allocates additional
space on the stack for various headers received during an Upgrade request.
An attacker could send a specially crafted request that causes this code
to overflow the stack, resulting in a crash.
* No longer allocate memory from the stack in a loop to parse the header
values. NOTE: There is a slight API change when using the passed in
strings as is. We now require the passed in strings to no longer have
leading or trailing whitespace. This isn't a problem as the only callers
have already done this before passing the strings to the affected
function.
asterisk 14.7.7:
* AST-2018-008: Fix enumeration of endpoints from ACL rejected addresses.
When endpoint specific ACL rules block a SIP request they respond with a
403 forbidden. However, if an endpoint is not identified then a 401
unauthorized response is sent. This vulnerability just discloses which
requests hit a defined endpoint. The ACL rules cannot be bypassed to gain
access to the disclosed endpoints.
* Made endpoint specific ACL rules now respond with a 401 unauthorized
which is the same as if an endpoint were not identified. The fix is
accomplished by replacing the found endpoint with the artificial endpoint
which always fails authentication.
asterisk 14.7.6:
* AST-2018-003: Crash with an invalid SDP fmtp attribute
pjproject's fmtp retrieval function failed to catch invalid fmtp attributes.
Because of this Asterisk would crash if given an SDP with an invalid fmtp
attribute.
When retrieving the format this patch now makes sure the fmtp attribute is
available. If not available it now returns an error status.
* AST-2018-002: Crash with an invalid SDP media format description
pjproject's media format parsing algorithm failed to catch invalid values.
Because of this Asterisk would crash if given an SDP with a invalid media
format description.
When parsing the media format description this patch now properly parses the
value and returns an error status if it can't successfully parse/convert the
value.
* AST-2018-005: res_pjsip_transport_management: Move to core
Since res_pjsip_transport_management provides several attack
mitigation features, its functionality moved to res_pjsip and
this module has been removed. This way the features will always
be available if res_pjsip is loaded.
* AST-2018-005: Fix tdata leaks when calling pjsip_endpt_send_response(2)
pjsip_distributor:
authenticate() creates a tdata and uses it to send a challenge or
failure response. When pjsip_endpt_send_response2() succeeds, it
automatically decrements the tdata ref count but when it fails, it
doesn't. Since we weren't checking for a return status, we weren't
decrementing the count ourselves on error and were therefore leaking
tdatas.
res_pjsip_session:
session_reinvite_on_rx_request wasn't decrementing the ref count
if an error happened while sending a 491 response.
pre_session_setup wasn't decrementing the ref count if
while sending an error after a pjsip_inv_verify_request failure.
res_pjsip:
ast_sip_send_response wasn't decrementing the ref count on error.
* AST-2018-005: Add a check for NULL tdata in ast_sip_failover_request
It was discovered that there are some corner cases where a pjsip tsx
might have no last_tx so calling ast_sip_failover_request with
a NULL last_tx as its tdata would cause a crash.
* AST-2018-004: Restrict the number of Accept headers in a SUBSCRIBE.
When receiving a SUBSCRIBE request the Accept headers from it are
stored locally. This operation has a fixed limit of 32 Accept headers
but this limit was not enforced. As a result it was possible for
memory outside of the allocated space to get written to resulting
in a crash.
This change enforces the limit so only 32 Accept headers are
processed.
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libsrtp 2.3.0
Major changes in this release are a fuzzer for libsrtp, NSS as optional crypto back end and cmake support for building. For more details and a complete list of changes please see the CHANGES file.
libsrtp 2.2.0
First release in the 2.2 series.
The major change with this release is that the all the code has been reformatted to be consistent and this consistency can be enforced with the include .clang-format file. This resulted in a lot of none functional changes but was considered worth it to simplify maintenance in the future. There are numerous other minor fixes, see the CHANGES file for more details.
libsrtp 2.1.0
First release in the 2.1 series.
libsrtp 2.0.0
Initial libsrtp 2.0 release.
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