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)
-----
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.pdfhttps://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.pdfhttps://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
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
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
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.
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.
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)
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.)
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
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)
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).
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".
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.
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.
Changelog:
16.10.0:
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)
Bugs fixed in this release:
-----------------------------------
[ASTERISK-28852] -
Unprotected access to nochecksums variable, causes build failures
(Reported by Guido Falsi)
[ASTERISK-28846] -
stream: Enforce formats immutability
(Reported by Joshua C. Colp)
[ASTERISK-28847] -
ARI channels cuts the endpoint string over 80 characters
(Reported by sungtae kim)
[ASTERISK-28811] -
Crash occurs when fax session switches from T.38 to audio
(Reported by Alexey Vasilyev)
[ASTERISK-28839] -
Sporadic crashes with Segmentation fault
(Reported by Joeran Vinzens)
[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-28841] -
app_confbridge: Add support for disabling text messaging for a user
(Reported by Joshua C. Colp)
[ASTERISK-28837] -
pjproject_bundled: Honor --without-pjproject.
(Reported by Alexander Traud)
[ASTERISK-28827] -
res_rtp_asterisk: Loop when receive buffer is flushed by a received packet that is also in receive buffer with NACK
(Reported by nappsoft)
[ASTERISK-27195] -
chan_sip: only sets ToS bits on UDP socket, ignoring TCP and TLS sockets
(Reported by Joshua Roys)
[ASTERISK-28826] -
res_rtp_asterisk: Duplicate seqnos being added to send buffer with NACK
(Reported by nappsoft)
[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-28819] -
[patch] bridge_softmix_binaural: Show state in menuselect.
(Reported by Alexander Traud)
[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-28796] -
func_channel: cannot read fields exten, context, userfield, channame from dialplan
(Reported by Sébastien Duthil)
[ASTERISK-28809] -
[patch] res_rtp_asterisk: Avoid absolute value on unsigned subtraction.
(Reported by Alexander Traud)
[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-28795] -
channel: write to a stream on multi-frame writes
(Reported by Kevin Harwell)
[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-28790] -
Crash during conference call using confbridge and video
(Reported by Pascal Cadotte Michaud)
[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-28784] -
res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
(Reported by Joshua C. Colp)
[ASTERISK-28743] -
Asterisk is crashing if the 200 OK with SDP
(Reported by sungtae kim)
[ASTERISK-28783] -
res_pjsip_session: Allow default non-audio streams to have reflected state
(Reported by Joshua C. Colp)
[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-28769] -
DTLS Handshake Fails to Occur if ice_support is enabled but not used
(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-19460] -
[patch] Function TXTCIDNAME never actually makes DNS calls and always returns an empty string
(Reported by George Joseph)
Improvements made in this release:
-----------------------------------
[ASTERISK-28853] -
Missing include on FreeBSD
(Reported by Guido Falsi)
[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)
[ASTERISK-28787] -
res_pjsip_session: Decide more intelligently when to add video
(Reported by Joshua C. Colp)
16.9.0:
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-28764] -
res_rtp_asterisk: Improve NACK support and seqno handling
(Reported by Joshua C. Colp)
[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-28697] -
res_pjsip: Named ACL does not update on reload if changed
(Reported by Timothy Vanderaerden)
[ASTERISK-28746] -
res_pjsip_outbound_registration keeps retrying the first entry in a SRV record set
(Reported by George Joseph)
[ASTERISK-28716] -
ICE: pjnath shouldn't wait for ICE to complete before allowing sending
(Reported by Benjamin Keith Ford)
[ASTERISK-28738] -
Incorrect state machine used when MOH_PASSTHRU is used
(Reported by Torrey Searle)
[ASTERISK-28742] -
res_rtp_asterisk: static for audio due to incomplete dtls/srtp setup
(Reported by Kevin Harwell)
[ASTERISK-28735] -
Realtime MoH Unknown format '' -- defaulting to SLIN
(Reported by Ross Beer)
[ASTERISK-28730] -
res_pjsip_session: Fix out of order session refreshes
(Reported by Joshua C. Colp)
[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-28713] -
res_stasis_playback: Error building JSON
(Reported by Sébastien Duthil)
[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-28686] -
chan_sip strictrtp=yes fails when media source is changed: no audio
(Reported by Walter Doekes)
[ASTERISK-28139] -
RTP Stream Incorrect Payload Type Causes Asterisk To Drop Calls
(Reported by Paul Brooks)
[ASTERISK-26955] -
pjsip: SIP Packets with Via "received=" Containing IPv6 Address Delimited by "[]" Rejected
(Reported by Peter Sokolov)
Improvements made in this release:
-----------------------------------
[ASTERISK-28750] -
TLS/SSL Key too small error
(Reported by Martin Zeh)
[ASTERISK-28733] -
stream: Add support for adding/removing streams during SFU/calls
(Reported by Joshua C. Colp)
[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)
16.8.0:
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-28702] -
chan_dahdi: holding a channel via flash to dialtone times out after 0:16:40
(Reported by Andrew Siplas)
[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-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 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-28562] -
SIP WSS message not processed until next frame arrives
(Reported by Robert Sutton)
[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-28604] -
app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
(Reported by George Joseph)
[ASTERISK-28659] -
res_pjsip_sdp_rtp: Bundle includes non-existent media stream if codecs create additional streams and offer does not have them
(Reported by nappsoft)
[ASTERISK-28660] -
res_fax: wrap Asterisk initiated negotiation with config option
(Reported by Kevin Harwell)
[ASTERISK-28636] -
app_chanisavail+cdr: ChanIsAvail sometimes fails to deactivate CDR.
(Reported by Frederic LE FOLL)
[ASTERISK-28626] -
Missing arguments in PJSIP_CONTACT function documentation
(Reported by Pascal Cadotte Michaud)
[ASTERISK-28609] -
Memory Leak in res_rtp_asterisk.c
(Reported by Ted G)
[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)
[ASTERISK-28625] -
Playback of local files impacted by large media cache
(Reported by Kevin Reeves)
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)
Parts are inspired by the FreeBSD port.
I could not easily find a telnetd with SSL support so I did not really test it.
Without SSL/TLS, it disconnects from NetBSD's telnetd if telnetd is run
with "-a valid" ("Authentication failed: No authentication method
available"); but "telnetd -a none" works.
The remserial program acts as a communications bridge between a
TCP/IP network port and a Linux device such as a serial port. Any
character-oriented Linux /dev device will work.
The program can also use pseudo-ttys as the device. A pseudo-tty
is like a serial port in that it has a /dev entry that can be opened
by a program that expects a serial port device, except that instead
of belonging to a physical serial device, the data can be intercepted
by another program. The remserial program uses this to connect a
network port to the "master" (programming) side of the pseudo-tty
allowing the device driver (slave) side to be used by some program
expecting a serial port. See example 3 below for details.
The program can operate as a server accepting network connections
from other machines, or as a client, connecting to remote machine
that is running the remserial program or some other program that
accepts a raw network connection. The network connection passes
data as-is, there is no control protocol over the network socket.
Multiple copies of the program can run on the same computer at the
same time assuming each is using a different network port and
device.
pkglint -r --network --only "migrate"
As a side-effect of migrating the homepages, pkglint also fixed a few
indentations in unrelated lines. These and the new homepages have been
checked manually.
pkglint --only "https instead of http" -r -F
With manual adjustments afterwards since pkglint 19.4.4 fixed a few
indentations in unrelated lines.
This mainly affects projects hosted at SourceForce, as well as
freedesktop.org, CTAN and GNU.
pkglint --only "https instead of http" -r -F
With manual adjustments afterwards since pkglint 19.4.4 fixed a few
indentations in unrelated lines.
This mainly affects projects hosted at SourceForce, as well as
freedesktop.org, CTAN and GNU.
Changes:
1.41.0
------
[-] * Documentation improvements.
[-] * Updated MySQL script to be compatible with current server versions.
[-] * Fixed SMSD operation on phones with more SMS folders.
[-] * Fixed off by one in Python example script.
[-] * Fixed PostgreSQL compilation on openSUSE.
[-] * Several compatibility fixes with recent compilers.
[-] * Improved USSD support.
[-] * Localization updates.
Changelog:
16.7.0
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 Sardañons)
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)
[ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by Florian Floimair)
Bugs fixed in this release:
-----------------------------------
[ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
[ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0 (Reported by George Joseph)
[ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream if codecs create additional streams and offer does not have them (Reported by nappsoft)
[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-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on hangup when TEST_FRAMEWORK enabled (Reported by Bernhard Schmidt)
[ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check failure when negociated ptime is not default ptime. (Reported by Frederic LE FOLL)
[ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the same (Reported by Ross Beer)
[ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received (Reported by Salah Ahmed)
[ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported by Kevin Harwell)
[ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported by Michael Cargile)
[ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a short alarm (Reported by Frederic LE FOLL)
[ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet length doesn't match (Reported by Joshua Elson)
[ASTERISK-26481] - FILE function grabs garbage along with read data when target line has no newline (Reported by Jonathan Harris)
[ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix bridge (Reported by Kevin Harwell)
[ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by Joshua C. Colp)
[ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
[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-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-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
[ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI (Reported by Jeremiah Gadd)
[ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by Jørgen 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 Ramière)
[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)
[ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
[ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI (Reported by Jeremiah Gadd)
[ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by Jørgen 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 Ramière)
[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)
[ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization (Reported by Joshua C. Colp)
16.6.0
Security bugs fixed in this release:
-----------------------------------
[ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream causes crash (Reported by Alexei Gradinari)
Bugs fixed in this release:
-----------------------------------
[ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
[ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
[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-28511] - codec_resample: Bad sound quality when up sampling from SLIN16 to SLIN32 (Reported by Ruddy G)
[ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up (Reported by Frederic LE FOLL)
[ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in cdr.conf (Reported by Frederic LE FOLL)
[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-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-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
[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-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS reflecting SIP response to transfer (Reported by Dan Cropp)
[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)
AUTOFIX: Makefile:290: Replacing "${PKGSRC_COMPILER} == \"clang\"" with "${PKGSRC_COMPILER:Mclang}".
The PKGSRC_COMPILER can be a list of chained compilers, e.g. "ccache
distcc clang". Therefore, comparing it using == or != leads to wrong
results in these cases.
Qodem is a from-scratch clone implementation of the Qmodem
communications program made popular in the days when Bulletin Board
Systems ruled the night. Qodem emulates the dialing directory and the
terminal screen features of Qmodem over both modem and Internet
connections.
OK kamil@
Asterisk is a complete PBX in software. It provides all of the
features you would expect from a PBX and more. Asterisk does voice
over IP in three protocols, and can interoperate with almost all
standards-based telephony equipment using relatively inexpensive
hardware.
Asterisk provides Voicemail services with Directory, Call Conferencing,
Interactive Voice Response, Call Queuing. It has support for
three-way calling, caller ID services, ADSI, SIP and H.323 (as both
client and gateway).
From the website:
2009-06-22 15:13:28 Version .30 released. FOP2 is born.
I have just released FOP 0.30, this version works reasonably well with
Asterisk 1.6. There are no new features. It is a maintenance and
compatiblity release.
I would also like to inform you that FOP2 is born. It is the next
generation FOP. A complete rewrite focused on the user and taking into
account all what I learned throughout the years.
Please visit http://www.fop2.com to read more about it.
FOP1 will not be discontinued. I will keep mantaining it but I won't be
adding any new features. I will fix bugs and make it work with future
asterisk releases.
Pkgsrc changes:
* Adapt to re-location to github
* Fix patching of the conserver.cf man page
* Adapt to README -> README.md change
* Enable LICENSE setting (even though there's more to it, see comment)
Upstream changes:
version 8.2.4 (March 26, 2019):
- Correct man page typo (Ed Maste <emaste@freebsd.org>)
- Remove autotools generated files from repo and create with release
- Better integration of Cirrus CI - FreeBSD, Linux, and MacOS
- Moving README to markdown
- Fix#12 - Remote infomation flags (i.e. "-x") cannot be filtered by console
- Fix#8 - defaultaccess appears broken
- Rename configure.in and use autoreconf
- Better use of version.h and letting configure build things with versions
version 8.2.3 (March 17, 2019):
- Correct 'impi' typo (Ed Maste <emaste@freebsd.org>)
- Correct argument type passed to time() (Ed Maste <emaste@freebsd.org>)
- Fix compilation without deprecated OpenSSL APIs
(Rosen Penev <rosenp@gmail.com>)
- Fix compilation without deprecated OpenSSL 1.1 APIs
(Rosen Penev <rosenp@gmail.com>)
- Fix#6 - clang "-Wstring-plus-int" warning
(Bryan Stansell <bryan@conserver.com>)
- configure.in: Add test for closefrom (Ed Maste <emaste@freebsd.org>)
- regenerate autoconf files (Ed Maste <emaste@freebsd.org>)
- Use closefrom if available (Ed Maste <emaste@freebsd.org>)
- Correct typo (Ed Maste <emaste@freebsd.org>)
- Add Cirrus-CI FreeBSD CI build config (Ed Maste <emaste@freebsd.org>)
- off by one found by Ed Maste (Bryan Stansell <bryan@conserver.com>)
version 8.2.2 (May 28, 2018):
- fixes for OpenSSL 1.1+ - patch by Eneas U de Queiroz
<cote2004-github@yahoo.com>
- adjustments to documentation after move to github
- removal of old RCS/CVS tags since we have git
wrong size, and the linker complained about ckcpro's 'dest' (which
was int vs long.)
i bumped the package version since it actually fixes real bugs on
big endian 64 bit platforms, and maybe bugs on other 64 bit.
Changes:
1.40.0
------
[+] * Added SMSD configuration option RetryTimeout.
[-] * Removed non configurable sleep after failed message send.
[+] * SMSD now tries to store whole decoded text for concatenated
messages in the first entry in database.
[-] * Improved compatibility with Sierra SL8084TR.
[+] * Added support for delivery reports stored in SR memory.
[+] * Configure CNMI parameters for AT driver.
0.4.0:
Fix2: reset LIGHT_EX colors with RESET_ALL.
Fix: ignore invalid "erase" ANSI codes.
Fix stream wrapping under PyCharm.
Added contextlib magic methods to ansitowin32.StreamWrapper.
Fix: don't cache stdio handles, since they might be closed/changed by fd redirection. This fixes an issue with pytest.
Drop support for EOL Python 2.5, 2.6, 3.1, 3.2 and 3.3, and add 3.6.
AST-2017-005, AST-2017-006, and AST-2017-008. There was no release
announcement as only security patches were issued. I just found
this update while looking to see what updates I was missing for
more recent versions of Asterisk. The Asterisk 11.x series was
declared end-of-life on Oct. 25th, 2017, so there will not be any
more updates to this package (other then PKGREVISION bumps for
dependencies) before it gets deleted. There is a reasonable chance
that there are unpatched vulnerabilities in this package. Anybody
still using it should upgrade a newer version as soon as possibble.
----- AST-2017-2005 -----
Description The "strictrtp" option in rtp.conf enables a feature of the
RTP stack that learns the source address of media for a
session and drops any packets that do not originate from
the expected address. This option is enabled by default in
Asterisk 11 and above.
The "nat" and "rtp_symmetric" options for chan_sip and
chan_pjsip respectively enable symmetric RTP support in the
RTP stack. This uses the source address of incoming media
as the target address of any sent media. This option is not
enabled by default but is commonly enabled to handle
devices behind NAT.
A change was made to the strict RTP support in the RTP
stack to better tolerate late media when a reinvite occurs.
When combined with the symmetric RTP support this
introduced an avenue where media could be hijacked. Instead
of only learning a new address when expected the new code
allowed a new source address to be learned at all times.
If a flood of RTP traffic was received the strict RTP
support would allow the new address to provide media and
with symmetric RTP enabled outgoing traffic would be sent
to this new address, allowing the media to be hijacked.
Provided the attacker continued to send traffic they would
continue to receive traffic as well.
Resolution The RTP stack will now only learn a new source address if it
has been told to expect the address to change. The RTCP
support has now also been updated to drop RTCP reports that
are not regarding the RTP session currently in progress. The
strict RTP learning progress has also been improved to guard
against a flood of RTP packets attempting to take over the
media stream.
----- AST-2017-006 -----
Description The app_minivm module has an "externnotify" program
configuration option that is executed by the MinivmNotify
dialplan application. The application uses the caller-id
name and number as part of a built string passed to the OS
shell for interpretation and execution. Since the caller-id
name and number can come from an untrusted source, a
crafted caller-id name or number allows an arbitrary shell
command injection.
Resolution Patched Asterisk's app_minivm module to use a different
system call that passes argument strings in an array instead
of having the OS shell determine the application parameter
boundaries.
----- AST-2017-008 -----
Description This is a follow up advisory to AST-2017-005.
Insufficient RTCP packet validation could allow reading
stale buffer contents and when combined with the "nat" and
"symmetric_rtp" options allow redirecting where Asterisk
sends the next RTCP report.
The RTP stream qualification to learn the source address of
media always accepted the first RTP packet as the new
source and allowed what AST-2017-005 was mitigating. The
intent was to qualify a series of packets before accepting
the new source address.
Resolution The RTP/RTCP stack will now validate RTCP packets before
processing them. Packets failing validation are discarded.
RTP stream qualification now requires the intended series of
packets from the same address without seeing packets from a
different source address to accept a new source address.
Asterisk is a complete PBX in software. It provides all of the
features you would expect from a PBX and more. Asterisk does voice
over IP in three protocols, and can interoperate with almost all
standards-based telephony equipment using relatively inexpensive
hardware.
Asterisk provides Voicemail services with Directory, Call Conferencing,
Interactive Voice Response, Call Queuing. It has support for
three-way calling, caller ID services, ADSI, SIP and H.323 (as both
client and gateway).
This is a standard version. It is scheduled to go to security
fixes only on October 3th, 2018, and EOL on October 3th, 2019.
See here for more information about Asterisk versions:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
pkgsrc-users@ a few weeks ago. This package is ancient and has
been EOL for a couple of years. It likely has numerous security
issues. Also, the PKGNAME will conflict with the upcoming Asterisk
18.* in a couple of years times. There were no objections.
Performing substitutions during post-patch breaks tools such as mkpatches,
making it very difficult to regenerate correct patches after making changes,
and often leading to substituted string replacements being committed.
pkgsrc changes:
- Indent a DEPENDS (suggested by `pkglint -Wall')
Changes:
1.39.0
------
* Fixed answering call in AT module.
* Improved support for Huawei E392 and E3131.
* Fixed compatibility of binaries with Windows XP and 2003.
* Improved support for ZTE MF667.
* Updated list of GSM networks and countries.
