minor features
pkgsrc changes:
- new version of core sounds
- add options for SNMP and PostgreSQL from Mike Bowie in PR/49661
and by popular demand
- add back support for menuselect personalization as that's how I was
doing menuselect non-interactively
- XXX need to look at a better way of doing this
- disable PJSIP for now as it doesn't work well on NetBSD from Mike Bowie
Since I added an option for PostgreSQL I also looked at adding an
option for directly using MySQL. Turns out that all the MySQL
modules are in the addons directory and are marked as being
deprecated. So I didn't bother. While investigating this, I also
noted that all the pgsql modules are marked as "extended" support.
This basically means that it is supported by the community, but
there is no one person listed as being responsible who would take
the lead for maintaining them. This basically means that they are
unsupported / low priority. See
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States .
Also with the pgsql modules, there is no way to do a database query
from the dialplan. Thus it is recommended to use the unixodbc
option as the modules are supported and offer the most functionality.
-----
The Asterisk Development Team has announced the release of Asterisk 11.19.0.
The release of Asterisk 11.19.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-25250 - chan_sip - Despite the channel being answered,
caller on a call established via Local channel continues to hear
ringback (Reported by Etienne Lessard)
* ASTERISK-25247 - choppy audio when spying on a g722 channel,
chan_sip or chan_pjsip (Reported by hristo)
* ASTERISK-24853 - Documentation claims chan_sip outbound
registrations support WS or WSS as valid transports (not true)
(Reported by PSDK)
* ASTERISK-25257 - [patch]channels/sig_pri.h -> sig_pri_span ->
force_restart_unavailable_chans in wrong scope (Reported by
Patric Marschall)
* ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
(Reported by Rusty Newton)
* ASTERISK-22805 - res_rtp_asterisk: Crash when calling
BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
(Reported by Dmitry Burilov)
* ASTERISK-24550 - res_rtp_asterisk: Crash in
ast_rtp_on_ice_complete during DTLS handshake (Reported by
Osaulenko Alexander)
* ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
Badalian Vyacheslav)
* ASTERISK-24832 - [patch]DTLS-crashes within openssl (Reported
by Stefan Engström)
* ASTERISK-25127 - DTLS crashes following "Unable to cancel
schedule ID" in dtls_srtp_check_pending (Reported by Dade
Brandon)
* ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
INVITE early Replace code (Reported by Walter Doekes)
* ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
(Reported by Walter Doekes)
* ASTERISK-25219 - [patch]Source and destination overlap in memcpy
in rtp_engine.c (Reported by Walter Doekes)
* ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
(Reported by Walter Doekes)
* ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
Bad file descriptor" (Reported by Barry Chern)
* ASTERISK-25202 - Hints extension state broken between 13.3.2 and
13.4 (Reported by cervajs)
* ASTERISK-25154 - [patch]fromtag may need to be updated after
successful call dialog match (Reported by Damian Ivereigh)
* ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
(Reported by Gregory Massel)
* ASTERISK-25094 - PBX core: Investigate thread safety issues
(Reported by Corey Farrell)
* ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
but asterisk doesn't detect it. (Reported by ibercom)
* ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
| adpcm | ipc10} (Reported by Badalian Vyacheslav)
* ASTERISK-25100 - asterisk coredump if host has an IPv6 address
that end with ::80 (Reported by Mark Petersen)
Improvements made in this release:
-----------------------------------
* ASTERISK-25040 - pbx: Improve performance of reloads by making
hint destruction more performant (Reported by Matt Jordan)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
Thank you for your continued support of Asterisk!
-----
The Asterisk Development Team has announced the release of Asterisk 11.18.0.
The release of Asterisk 11.18.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-25112 - Logger: Configuration settings are not reset to
default during reload. (Reported by Corey Farrell)
* ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
or more digits (Reported by Makoto Dei)
* ASTERISK-24944 - main/audiohook.c change prevents G722 call
recording (Reported by Ronald Raikes)
* ASTERISK-25083 - Message.c: Message channel becomes saturated
with frames leading to spammy log messages (Reported by Jonathan
Rose)
* ASTERISK-25041 - [patch]Broken column type checking in
res_config_mysql addon (Reported by Alexandre Fournier)
* ASTERISK-21893 - Segfault after call hangup, in
ast_channel_hangupcause_set, at channel_internal_api.c (Reported
by Alexandr Gordeev)
* ASTERISK-25074 - Regression: Recent clang-related change broke
cross compiling of Asterisk (Reported by Sebastian Kemper)
* ASTERISK-25042 - asterisk.conf options override command-line
options. (Reported by Corey Farrell)
* ASTERISK-24442 - Outgoing call files don't work properly when
set in the future (Reported by tootai)
* ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
ignore ISDN RESTART requests. (Reported by Richard Mudgett)
* ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
contain waiting time (Reported by Etienne Lessard)
* ASTERISK-22708 - res_odbc.conf negative_connection_cache option
not respected, failover between DSNs doesn't work (Reported by
JoshE)
* ASTERISK-25028 - Build System: Unneeded defines in
asterisk/buildopts.h (Reported by Corey Farrell)
* ASTERISK-19608 - Asterisk-1.8.x starts rejecting calls with
cause code 44 after some time. (Reported by Denis Alberto
Martinez)
* ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
(Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
(Reported by Steve Davies)
* ASTERISK-22790 - check_modem_rate() may return incorrect rate
for V.27 (Reported by not here)
* ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
to minrate=2400, then res_fax refuse to load (Reported by David
Brillert)
* ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
which is disallowed in res_fax's check_modem_rate (Reported by
Matt Jordan)
* ASTERISK-24916 - Increasing memory usage when multiple reinvite
during call (Reported by Christophe Osuna)
* ASTERISK-19538 - Asterisk segfaults on sippeers realtime
redundancy (Reported by Alex)
* ASTERISK-24749 - ConfBridge: Wrong language on playing
conf-hasjoin and conf-hasleft when played to bridge (Reported by
Philippe Bolduc)
* ASTERISK-24991 - Check for ao2_alloc failure in
__ast_channel_internal_alloc (Reported by Corey Farrell)
* ASTERISK-24895 - After hangup on the side of the ISDN network no
HangupRequest event comes for the dahdi channel. (Reported by
Andrew Zherdin)
* ASTERISK-24774 - Segfault in ast_context_destroy with
extensions.ael and extensions.conf (Reported by Corey Farrell)
* ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
to Fail (Reported by Ashley Sanders)
* ASTERISK-24959 - [patch]CLI command cdr show pgsql status
(Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-24954 - Git migration: Asterisk version numbers are
incompatible with the Test Suite (Reported by Matt Jordan)
* ASTERISK-21777 - Asterisk tries to transcode video instead of
audio (Reported by Nick Ruggles)
* ASTERISK-24380 - core: Native formats are set to h264 with
certain audio/video codec configuration, resulting in path
translation WARNINGs (Reported by Matt Jordan)
* ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
into account (Reported by Frederic Van Espen)
* ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
short (Reported by Y Ateya)
* ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
(Reported by Vadim)
* ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
byte prefix bug (Reported by Matt Jordan)
* ASTERISK-21211 - chan_iax2 - unprotected access of
iaxs[peer->callno] potentially results in segfault (Reported by
Jaco Kroon)
* ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
(Reported by Christoph Timm)
* ASTERISK-24942 - Voicemail API: message is deleted when
destination mailbox is at maxmsg (Reported by Scott Griepentrog)
* ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
(Reported by Jeffrey C. Ollie)
* ASTERISK-21854 - Long Asterisk-version strings display
improperly in the 'Connected to ...' line upon remote console
connection (Reported by klaus3000)
* ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
detection in ast_malloc (Reported by Timo Teräs)
* ASTERISK-24142 - CCSS: crash during shutdown due to device
lookup in destroyed container (Reported by David Brillert)
* ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
core restart now (Reported by Peter Katzmann)
* ASTERISK-24805 - [patch] - ASAN: Race condition
(heap-use-after-free) on asterisk closing (Reported by Badalian
Vyacheslav)
* ASTERISK-24881 - ast_register_atexit should only be used when
absolutely needed (Reported by Corey Farrell)
* ASTERISK-24864 - app_confbridge: file playback blocks dtmf
(Reported by Kevin Harwell)
* ASTERISK-14233 - [patch] Buddies are always auto-registered when
processing the roster (Reported by Simon Arlott)
* ASTERISK-24780 - [patch] - Buddies are always auto-registered
when processing the roster (Reported by Simon Arlott)
Improvements made in this release:
-----------------------------------
* ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
Hjelm)
* ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
(Reported by Alexander Traud)
* ASTERISK-24917 - [patch] clang compilation warnings (Reported by
Diederik de Groot)
* ASTERISK-25040 - pbx: Improve performance of reloads by making
hint destruction more performant (Reported by Matt Jordan)
* ASTERISK-24965 - cel_pgsql - log_error string references CDR
instead of CEL (Reported by Rodrigo Ramirez Norambuena)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
Thank you for your continued support of Asterisk!
pkgsrc changes:
- adapt to upstream support for clang
- more comprehensive sweep for 64-bit time_t related stuff
- XXX pjsip has its own time related stuff that is 32-bit only
-----
The Asterisk Development Team has announced security releases for Certified
Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
The release of these versions resolves the following security vulnerability:
* AST-2015-003: TLS Certificate Common name NULL byte exploit
When Asterisk registers to a SIP TLS device and and verifies the server,
Asterisk will accept signed certificates that match a common name other than
the one Asterisk is expecting if the signed certificate has a common name
containing a null byte after the portion of the common name that Asterisk
expected. This potentially allows for a man in the middle attack.
For more information about the details of this vulnerability, please read
security advisory AST-2015-003, which was released at the same time as this
announcement.
For a full list of changes in the current releases, please see the ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
The security advisory is available at:
* http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
Thank you for your continued support of Asterisk!
-----
The Asterisk Development Team has announced the release of Asterisk 11.17.0.
The release of Asterisk 11.17.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
New Features made in this release:
-----------------------------------
* ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
(Reported by Dwayne Hubbard)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
res_odbc (Reported by ibercom)
* ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
with replaces (Reported by Eelco Brolman)
* ASTERISK-24479 - Enable REF_DEBUG for module references
(Reported by Corey Farrell)
* ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
fully disconnect underlying socket, leading to events being
dropped with no additional information (Reported by Matt Jordan)
* ASTERISK-24772 - ODBC error in realtime sippeers when device
unregisters under MariaDB (Reported by Richard Miller)
* ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
(Reported by Corey Farrell)
* ASTERISK-24799 - [patch] make fails with undefined reference to
SSLv3_client_method (Reported by Alexander Traud)
* ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
for playing back messages stored in IMAP - play_message: No
origtime (Reported by Graham Barnett)
* ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
OSX with 64 bit integers (Reported by Corey Farrell)
* ASTERISK-24796 - Codecs and bucket schema's prevent module
unload (Reported by Corey Farrell)
* ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
(Reported by Ashley Sanders)
* ASTERISK-24797 - bridge_softmix: G.729 codec license held
(Reported by Kevin Harwell)
* ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
thread ID being passed to pthread_kill (Reported by JoshE)
* ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
fail (Reported by Terry Wilson)
* ASTERISK-23214 - chan_sip WARNING message 'We are requesting
SRTP for audio, but they responded without it' is ambiguous and
wrong in some cases (Reported by Rusty Newton)
* ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
error response and BYE are sent to the caller (Reported by
Makoto Dei)
* ASTERISK-18105 - most of asterisk modules are unbuildable in
cygwin environment (Reported by feyfre)
* ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
* ASTERISK-24838 - chan_sip: Locking inversion occurs when
building a peer causes a peer poke during request handling
(Reported by Richard Mudgett)
* ASTERISK-24825 - Caller ID not recognized using
Centrex/Distinctive dialing (Reported by Richard Mudgett)
* ASTERISK-24739 - [patch] - Out of files -- call fails --
numerous files with inodes from under /usr/share/zoneinfo,
mostly posixrules (Reported by Ed Hynan)
* ASTERISK-23390 - NewExten Event with application AGI shows up
before and after AGI runs (Reported by Benjamin Keith Ford)
* ASTERISK-24786 - [patch] - Asterisk terminates when playing a
voicemail stored in LDAP (Reported by Graham Barnett)
* ASTERISK-24808 - res_config_odbc: Improper escaping of
backslashes occurs with MySQL (Reported by Javier Acosta)
* ASTERISK-20850 - [patch]Nested functions aren't portable.
Adapting RAII_VAR to use clang/llvm blocks to get the
same/similar functionality. (Reported by Diederik de Groot)
* ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
by Frank DiGennaro)
* ASTERISK-21038 - Bad command completion of "core set debug
channel" (Reported by Richard Kenner)
* ASTERISK-18708 - func_curl hangs channel under load (Reported by
Dave Cabot)
* ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
Atis Lezdins)
* ASTERISK-24876 - Investigate reference leaks from
tests/channels/local/local_optimize_away (Reported by Corey
Farrell)
* ASTERISK-24817 - init_logger_chain: unreachable code block
(Reported by Corey Farrell)
* ASTERISK-24880 - [patch]Compilation under OpenBSD (Reported by
snuffy)
* ASTERISK-24879 - [patch]Compilation fails due to 64bit time
under OpenBSD (Reported by snuffy)
Improvements made in this release:
-----------------------------------
* ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
Couldn't find mailbox %s in context (Reported by Graham Barnett)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
Thank you for your continued support of Asterisk!
-----
The Asterisk Development Team has announced the release of Asterisk 11.16.0.
The release of Asterisk 11.16.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
from JSSIP (Reported by Badalian Vyacheslav)
* ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
enabled (Reported by Richard Mudgett)
* ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
enabled (Reported by Andreas Steinmetz)
* ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
casts char to unsigned int (Reported by Walter Doekes)
* ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
level - 'Remote address is null, most likely RTP has been
stopped' (Reported by Rusty Newton)
* ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
on startup (Reported by Richard Kenner)
* ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
destination when 'sendrpid=yes' (in proxy environment) (Reported
by Karsten Wemheuer)
* ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
(Reported by Kristian Høgh)
* ASTERISK-20744 - [patch] Security event logging does not work
over syslog (Reported by Michael Keuter)
* ASTERISK-23850 - Park Application does not respect Return
Context Priority (Reported by Andrew Nagy)
* ASTERISK-23991 - [patch]asterisk.pc file contains a small error
in the CFlags returned (Reported by Diederik de Groot)
* ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
voicemail is not deleted after review, hangup (Reported by LEI
FU)
* ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
32-bit packages on 64-bit hosts (Reported by Ben Klang)
* ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
m() option does not queue an MWI event (Reported by Gareth
Palmer)
* ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
column comparison for 'defaultuser' (Reported by
HZMI8gkCvPpom0tM)
* ASTERISK-24719 - ConfBridge recording channels get stuck when
recording started/stopped more than once (Reported by Richard
Mudgett)
* ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
by Kevin Harwell)
* ASTERISK-24728 - tcptls: Bad file descriptor error when
reloading chan_sip (Reported by Kevin Harwell)
* ASTERISK-24676 - Security Vulnerability: URL request injection
in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
* ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
versions (Reported by Jared Biel)
* ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
Stephan Eisvogel)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0
Thank you for your continued support of Asterisk!
pkgsrc change: adapt to splitting up of speex
The Asterisk Development Team has announced security releases for Certified
Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10,
11.15.1, 12.8.1, and 13.1.1.
The release of these versions resolves the following security vulnerabilities:
* AST-2015-001: File descriptor leak when incompatible codecs are offered
Asterisk may be configured to only allow specific audio or
video codecs to be used when communicating with a
particular endpoint. When an endpoint sends an SDP offer
that only lists codecs not allowed by Asterisk, the offer
is rejected. However, in this case, RTP ports that are
allocated in the process are not reclaimed.
This issue only affects the PJSIP channel driver in
Asterisk. Users of the chan_sip channel driver are not
affected.
* AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability
CVE-2014-8150 reported an HTTP request injection
vulnerability in libcURL. Asterisk uses libcURL in its
func_curl.so module (the CURL() dialplan function), as well
as its res_config_curl.so (cURL realtime backend) modules.
Since Asterisk may be configured to allow for user-supplied
URLs to be passed to libcURL, it is possible that an
attacker could use Asterisk as an attack vector to inject
unauthorized HTTP requests if the version of libcURL
installed on the Asterisk server is affected by
CVE-2014-8150.
For more information about the details of these vulnerabilities, please read
security advisory AST-2015-001 and AST-2015-002, which were released at the same
time as this announcement.
For a full list of changes in the current releases, please see the ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.2http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.15.1
The security advisories are available at:
* http://downloads.asterisk.org/pub/security/AST-2015-001.pdf
* http://downloads.asterisk.org/pub/security/AST-2015-002.pdf
Thank you for your continued support of Asterisk!
The Asterisk Development Team has announced the release of Asterisk 11.15.0.
The release of Asterisk 11.15.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-20127 - [Regression] Config.c config_text_file_load()
unescapes semicolons ("\;" -> ";") turning them into comments
(corruption) on rewrite of a config file (Reported by George
Joseph)
* ASTERISK-24307 - Unintentional memory retention in stringfields
(Reported by Etienne Lessard)
* ASTERISK-24492 - main/file.c: ast_filestream sometimes causes
extra calls to ast_module_unref (Reported by Corey Farrell)
* ASTERISK-24504 - chan_console: Fix reference leaks to pvt
(Reported by Corey Farrell)
* ASTERISK-24468 - Incoming UCS2 encoded SMS truncated if SMS
length exceeds 50 (roughly) national symbols (Reported by
Dmitriy Bubnov)
* ASTERISK-24500 - Regression introduced in chan_mgcp by SVN
revision r227276 (Reported by Xavier Hienne)
* ASTERISK-20402 - Unable to cancel (features.conf) attended
transfer (Reported by Matt Riddell)
* ASTERISK-24505 - manager: http connections leak references
(Reported by Corey Farrell)
* ASTERISK-24502 - Build fails when dev-mode, dont optimize and
coverage are enabled (Reported by Corey Farrell)
* ASTERISK-24444 - PBX: Crash when generating extension for
pattern matching hint (Reported by Leandro Dardini)
* ASTERISK-24522 - ConfBridge: delay occurs between kicking all
endmarked users when last marked user leaves (Reported by Matt
Jordan)
* ASTERISK-15242 - transmit_refer leaks sip_refer structures
(Reported by David Woolley)
* ASTERISK-24440 - Call leak in Confbridge (Reported by Ben Klang)
* ASTERISK-24469 - Security Vulnerability: Mixed IPv4/IPv6 ACLs
allow blocked addresses through (Reported by Matt Jordan)
* ASTERISK-24516 - [patch]Asterisk segfaults when playing back
voicemail under high concurrency with an IMAP backend (Reported
by David Duncan Ross Palmer)
* ASTERISK-24572 - [patch]App_meetme is loaded without its
defaults when the configuration file is missing (Reported by
Nuno Borges)
* ASTERISK-24573 - [patch]Out of sync conversation recording when
divided in multiple recordings (Reported by Nuno Borges)
Improvements made in this release:
-----------------------------------
* ASTERISK-24283 - [patch]Microseconds precision in the eventtime
column in the cel_odbc module (Reported by Etienne Lessard)
* ASTERISK-24530 - [patch] app_record stripping 1/4 second from
recordings (Reported by Ben Smithurst)
* ASTERISK-24577 - Speed up loopback switches by avoiding unneeded
lookups (Reported by Birger "WIMPy" Harzenetter)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.15.0
Thank you for your continued support of Asterisk!
The Asterisk Development Team has announced security releases for
Certified Asterisk 11.6 and Asterisk 11, 12, and 13. The available
security releases are released as versions 11.6-cert9, 11.14.2,
12.7.2, and 13.0.2.
The release of these versions resolves the following security vulnerability:
* AST-2014-019: Remote Crash Vulnerability in WebSocket Server
When handling a WebSocket frame the res_http_websocket module
dynamically changes the size of the memory used to allow the
provided payload to fit. If a payload length of zero was received
the code would incorrectly attempt to resize to zero. This
operation would succeed and end up freeing the memory but be
treated as a failure. When the session was subsequently torn down
this memory would get freed yet again causing a crash.
For more information about the details of this vulnerability, please read
security advisory AST-2014-019, which was released at the same time as this
announcement.
For a full list of changes in the current releases, please see the Change Logs:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.14.2
The security advisory is available at:
* http://downloads.asterisk.org/pub/security/AST-2014-019.pdf
Thank you for your continued support of Asterisk!
The Asterisk Development Team has announced security releases for Certified
Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
security releases are released as versions 1.8.28-cert3, 11.6-cert8, 1.8.32.1,
11.14.1, 12.7.1, and 13.0.1.
The release of these versions resolves the following security vulnerabilities:
* AST-2014-012: Unauthorized access in the presence of ACLs with mixed IP
address families
Many modules in Asterisk that service incoming IP traffic have ACL options
("permit" and "deny") that can be used to whitelist or blacklist address
ranges. A bug has been discovered where the address family of incoming
packets is only compared to the IP address family of the first entry in the
list of access control rules. If the source IP address for an incoming
packet is not of the same address as the first ACL entry, that packet
bypasses all ACL rules.
* AST-2014-018: Permission Escalation through DB dialplan function
The DB dialplan function when executed from an external protocol, such as AMI,
could result in a privilege escalation. Users with a lower class authorization
in AMI can access the internal Asterisk database without the required SYSTEM
class authorization.
In addition, the release of 11.6-cert8 and 11.14.1 resolves the following
security vulnerability:
* AST-2014-014: High call load with ConfBridge can result in resource exhaustion
The ConfBridge application uses an internal bridging API to implement
conference bridges. This internal API uses a state model for channels within
the conference bridge and transitions between states as different things
occur. Unload load it is possible for some state transitions to be delayed
causing the channel to transition from being hung up to waiting for media. As
the channel has been hung up remotely no further media will arrive and the
channel will stay within ConfBridge indefinitely.
In addition, the release of 11.6-cert8, 11.14.1, 12.7.1, and 13.0.1 resolves
the following security vulnerability:
* AST-2014-017: Permission Escalation via ConfBridge dialplan function and
AMI ConfbridgeStartRecord Action
The CONFBRIDGE dialplan function when executed from an external protocol (such
as AMI) can result in a privilege escalation as certain options within that
function can affect the underlying system. Additionally, the AMI
ConfbridgeStartRecord action has options that would allow modification of the
underlying system, and does not require SYSTEM class authorization in AMI.
For a full list of changes in the current releases, please see the ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.14.1
The security advisories are available at:
* http://downloads.asterisk.org/pub/security/AST-2014-012.pdf
* http://downloads.asterisk.org/pub/security/AST-2014-014.pdf
* http://downloads.asterisk.org/pub/security/AST-2014-017.pdf
* http://downloads.asterisk.org/pub/security/AST-2014-018.pdf
Thank you for your continued support of Asterisk!
The Asterisk Development Team has announced the release of Asterisk 11.14.0.
The release of Asterisk 11.14.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-24348 - Built-in editline tab complete segfault with
MALLOC_DEBUG (Reported by Walter Doekes)
* ASTERISK-24335 - [PATCH] Asterisk incorrectly responds 503 to
INVITE retransmissions of rejected calls (Reported by Torrey
Searle)
* ASTERISK-23768 - [patch] Asterisk man page contains a (new)
unquoted minus sign (Reported by Jeremy Lainé)
* ASTERISK-24357 - [fax] Out of bounds error in update_modem_bits
(Reported by Jeremy Lainé)
* ASTERISK-20567 - bashism in autosupport (Reported by Tzafrir
Cohen)
* ASTERISK-22945 - [patch] Memory leaks in chan_sip.c with
realtime peers (Reported by ibercom)
* ASTERISK-24384 - chan_motif: format capabilities leak on module
load error (Reported by Corey Farrell)
* ASTERISK-24385 - chan_sip: process_sdp leaks on an error path
(Reported by Corey Farrell)
* ASTERISK-24378 - Release AMI connections on shutdown (Reported
by Corey Farrell)
* ASTERISK-24354 - AMI sendMessage closes AMI connection on error
(Reported by Peter Katzmann)
* ASTERISK-24390 - astobj2: REF_DEBUG reports false leaks with
ao2_callback with OBJ_MULTIPLE (Reported by Corey Farrell)
* ASTERISK-24326 - res_rtp_asterisk: ICE-TCP candidates are
incorrectly attempted (Reported by Joshua Colp)
* ASTERISK-24011 - [patch]safe_asterisk tries to set ulimit -n too
high on linux systems with lots of RAM (Reported by Michael
Myles)
* ASTERISK-24383 - res_rtp_asterisk: Crash if no candidates
received for component (Reported by Kevin Harwell)
* ASTERISK-20784 - Failure to receive an ACK to a SIP Re-INVITE
results in a SIP channel leak (Reported by NITESH BANSAL)
* ASTERISK-15879 - [patch] Failure to receive an ACK to a SIP
Re-INVITE results in a SIP channel leak (Reported by Torrey
Searle)
* ASTERISK-24406 - Some caller ID strings are parsed differently
since 11.13.0 (Reported by Etienne Lessard)
* ASTERISK-24325 - res_calendar_ews: cannot be used with neon 0.30
(Reported by Tzafrir Cohen)
* ASTERISK-13797 - [patch] relax badshell tilde test (Reported by
Tzafrir Cohen)
* ASTERISK-22791 - asterisk sends Re-INVITE after receiving a BYE
(Reported by Paolo Compagnini)
* ASTERISK-18923 - res_fax_spandsp usage counter is wrong
(Reported by Grigoriy Puzankin)
* ASTERISK-24392 - res_fax: fax gateway sessions leak (Reported by
Corey Farrell)
* ASTERISK-24393 - rtptimeout=0 doesn't disable rtptimeout
(Reported by Dmitry Melekhov)
* ASTERISK-23846 - Unistim multilines. Loss of voice after second
call drops (on a second line). (Reported by Rustam Khankishyiev)
* ASTERISK-24063 - [patch]Asterisk does not respect outbound proxy
when sending qualify requests (Reported by Damian Ivereigh)
* ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of
SSLv3, security fix POODLE (CVE-2014-3566) (Reported by
abelbeck)
* ASTERISK-24436 - Missing header in res/res_srtp.c when compiling
against libsrtp-1.5.0 (Reported by Patrick Laimbock)
* ASTERISK-24454 - app_queue: ao2_iterator not destroyed, causing
leak (Reported by Corey Farrell)
* ASTERISK-24430 - missing letter "p" in word response in
OriginateResponse event documentation (Reported by Dafi Ni)
* ASTERISK-24457 - res_fax: fax gateway frames leak (Reported by
Corey Farrell)
* ASTERISK-21721 - SIP Failed to parse multiple Supported: headers
(Reported by Olle Johansson)
* ASTERISK-24304 - asterisk crashing randomly because of unistim
channel (Reported by dhanapathy sathya)
* ASTERISK-24190 - IMAP voicemail causes segfault (Reported by
Nick Adams)
* ASTERISK-24466 - app_queue: fix a couple leaks to struct
call_queue (Reported by Corey Farrell)
* ASTERISK-24432 - Install refcounter.py when REF_DEBUG is enabled
(Reported by Corey Farrell)
* ASTERISK-24476 - main/app.c / app_voicemail: ast_writestream
leaks (Reported by Corey Farrell)
* ASTERISK-24307 - Unintentional memory retention in stringfields
(Reported by Etienne Lessard)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.14.0
Thank you for your continued support of Asterisk!
The Asterisk Development Team has announced the release of Asterisk 11.13.0.
The release of Asterisk 11.13.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-24032 - Gentoo compilation emits warning:
"_FORTIFY_SOURCE" redefined (Reported by Kilburn)
* ASTERISK-24225 - Dial option z is broken (Reported by
dimitripietro)
* ASTERISK-24178 - [patch]fromdomainport used even if not set
(Reported by Elazar Broad)
* ASTERISK-22252 - res_musiconhold cleanup - REF_DEBUG reload
warnings and ref leaks (Reported by Walter Doekes)
* ASTERISK-23997 - chan_sip: port incorrectly incremented for RTCP
ICE candidates in SDP answer (Reported by Badalian Vyacheslav)
* ASTERISK-24019 - When a Music On Hold stream starts it restarts
at beginning of file. (Reported by Jason Richards)
* ASTERISK-23767 - [patch] Dynamic IAX2 registration stops trying
if ever not able to resolve (Reported by David Herselman)
* ASTERISK-24211 - testsuite: Fix the dial_LS_options test
(Reported by Matt Jordan)
* ASTERISK-24249 - SIP debugs do not stop (Reported by Avinash
Mohod)
* ASTERISK-23577 - res_rtp_asterisk: Crash in
ast_rtp_on_turn_rtp_state when RTP instance is NULL (Reported by
Jay Jideliov)
* ASTERISK-23634 - With TURN Asterisk crashes on multiple (7-10)
concurrent WebRTC (avpg/encryption/icesupport) calls (Reported
by Roman Skvirsky)
* ASTERISK-24301 - Security: Out of call MESSAGE requests
processed via Message channel driver can crash Asterisk
(Reported by Matt Jordan)
Improvements made in this release:
-----------------------------------
* ASTERISK-24171 - [patch] Provide a manpage for the aelparse
utility (Reported by Jeremy Lainé)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.13.0
Thank you for your continued support of Asterisk!
The Asterisk Development Team has announced security releases for Certified
Asterisk 11.6 and Asterisk 11 and 12. The available security releases are
released as versions 11.6-cert6, 11.12.1, and 12.5.1.
Please note that the release of these versions resolves the following security
vulnerability:
* AST-2014-010: Remote Crash when Handling Out of Call Message in Certain
Dialplan Configurations
Note that the crash described in AST-2014-010 can be worked around through
dialplan configuration. Given the likelihood of the issue, an advisory was
deemed to be warranted.
For more information about the details of these vulnerabilities, please read
security advisories AST-2014-009 and AST-2014-010, which were released at the
same time as this announcement.
For a full list of changes in the current releases, please see the ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.12.1
The security advisories are available at:
* http://downloads.asterisk.org/pub/security/AST-2014-010.pdf
Thank you for your continued support of Asterisk!
The Asterisk Development Team has announced the release of Asterisk 11.12.0.
The release of Asterisk 11.12.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-23911 - URIENCODE/URIDECODE: WARNING about passing an
empty string is a bit over zealous (Reported by Matt Jordan)
* ASTERISK-23985 - PresenceState Action response does not contain
ActionID; duplicates Message Header (Reported by Matt Jordan)
* ASTERISK-23814 - No call started after peer dialed (Reported by
Igor Goncharovsky)
* ASTERISK-24087 - [patch]chan_sip: sip_subscribe_mwi_destroy
should not call sip_destroy (Reported by Corey Farrell)
* ASTERISK-23818 - PBX_Lua: after asterisk startup module is
loaded, but dialplan not available (Reported by Dennis Guse)
* ASTERISK-18345 - [patch] sips connection dropped by asterisk
with a large INVITE (Reported by Stephane Chazelas)
* ASTERISK-23508 - Memory Corruption in
__ast_string_field_ptr_build_va (Reported by Arnd Schmitter)
Improvements made in this release:
-----------------------------------
* ASTERISK-21178 - Improve documentation for manager command
Getvar, Setvar (Reported by Rusty Newton)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.12.0
Thank you for your continued support of Asterisk!
pkgsrc change: MAKE_JOBS_SAFE=NO from joerg@
The Asterisk Development Team has announced the release of Asterisk 11.11.0.
The release of Asterisk 11.11.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-22551 - Session timer : UAS (Asterisk) starts counting
at Invite, UAC starts counting at 200 OK. (Reported by i2045)
* ASTERISK-23792 - Mutex left locked in chan_unistim.c (Reported
by Peter Whisker)
* ASTERISK-23582 - [patch]Inconsistent column length in *odbc
(Reported by Walter Doekes)
* ASTERISK-23803 - AMI action UpdateConfig EmptyCat clears all
categories but the requested one (Reported by zvision)
* ASTERISK-23035 - ConfBridge with name longer than max (32 chars)
results in several bridges with same conf_name (Reported by
Iñaki Cívico)
* ASTERISK-23824 - ConfBridge: Users cannot be muted via CLI or
AMI when waiting to enter a conference (Reported by Matt Jordan)
* ASTERISK-23683 - #includes - wildcard character in a path more
than one directory deep - results in no config parsing on module
reload (Reported by tootai)
* ASTERISK-23827 - autoservice thread doesn't exit at shutdown
(Reported by Corey Farrell)
* ASTERISK-23609 - Security: AMI action MixMonitor allows
arbitrary programs to be run (Reported by Corey Farrell)
* ASTERISK-23673 - Security: DOS by consuming the number of
allowed HTTP connections. (Reported by Richard Mudgett)
* ASTERISK-23246 - DEBUG messages in sdp_crypto.c display despite
a DEBUG level of zero (Reported by Rusty Newton)
* ASTERISK-23766 - [patch] Specify timeout for database write in
SQLite (Reported by Igor Goncharovsky)
* ASTERISK-23844 - Load of pbx_lua fails on sample extensions.lua
with Lua 5.2 or greater due to addition of goto statement
(Reported by Rusty Newton)
* ASTERISK-23818 - PBX_Lua: after asterisk startup module is
loaded, but dialplan not available (Reported by Dennis Guse)
* ASTERISK-23834 - res_rtp_asterisk debug message gives wrong
length if ICE (Reported by Richard Kenner)
* ASTERISK-23790 - [patch] - SIP From headers longer than 256
characters result in dropped call and 'No closing bracket'
warnings. (Reported by uniken1)
* ASTERISK-23917 - res_http_websocket: Delay in client processing
large streams of data causes disconnect and stuck socket
(Reported by Matt Jordan)
* ASTERISK-23908 - [patch]When using FEC error correction,
asterisk tries considers negative sequence numbers as missing
(Reported by Torrey Searle)
* ASTERISK-23921 - refcounter.py uses excessive ram for large refs
files (Reported by Corey Farrell)
* ASTERISK-23948 - REF_DEBUG fails to record ao2_ref against
objects that were already freed (Reported by Corey Farrell)
* ASTERISK-23916 - [patch]SIP/SDP fmtp line may include whitespace
between attributes (Reported by Alexander Traud)
* ASTERISK-23984 - Infinite loop possible in ast_careful_fwrite()
(Reported by Steve Davies)
* ASTERISK-23897 - [patch]Change in SETUP ACK handling (checking
PI) in revision 413765 breaks working environments (Reported by
Pavel Troller)
Improvements made in this release:
-----------------------------------
* ASTERISK-23492 - Add option to safe_asterisk to disable
backgrounding (Reported by Walter Doekes)
* ASTERISK-22961 - [patch] DTLS-SRTP not working with SHA-256
(Reported by Jay Jideliov)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.11.0
Thank you for your continued support of Asterisk!
with general bug fixes. The security issues fixed are: AST-2014-001,
AST-2014-002, AST-2014-006, and AST-2014-007.
-----
The Asterisk Development Team has announced security releases for
Certified Asterisk 1.8.15, 11.6, and Asterisk 1.8, 11, and 12. The
available security releases are released as versions 1.8.15-cert7,
11.6-cert4, 1.8.28.2, 11.10.2, and 12.3.2.
These releases resolve security vulnerabilities that were previously
fixed in 1.8.15-cert6, 11.6-cert3, 1.8.28.1, 11.10.1, and 12.3.1.
Unfortunately, the fix for AST-2014-007 inadvertently introduced
a regression in Asterisk's TCP and TLS handling that prevented
Asterisk from sending data over these transports. This regression
and the security vulnerabilities have been fixed in the versions
specified in this release announcement.
Please note that the release of these versions resolves the following security
vulnerabilities:
* AST-2014-006: Permission Escalation via Asterisk Manager User Unauthorized
Shell Access
* AST-2014-007: Denial of Service via Exhaustion of Allowed Concurrent HTTP
Connections
For more information about the details of these vulnerabilities,
please read security advisories AST-2014-005, AST-2014-006,
AST-2014-007, and AST-2014-008, which were released with the previous
versions that addressed these vulnerabilities.
For a full list of changes in the current releases, please see the ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.10.2
The security advisories are available at:
* http://downloads.asterisk.org/pub/security/AST-2014-006.pdf
* http://downloads.asterisk.org/pub/security/AST-2014-007.pdf
Thank you for your continued support of Asterisk!
-----
The Asterisk Development Team has announced security releases for
Certified Asterisk 1.8.15, 11.6, and Asterisk 1.8, 11, and 12. The
available security releases are released as versions 1.8.15-cert6,
11.6-cert3, 1.8.28.1, 11.10.1, and 12.3.1.
The release of these versions resolves the following issue:
* AST-2014-007: Denial of Service via Exhaustion of Allowed Concurrent HTTP
Connections
Establishing a TCP or TLS connection to the configured HTTP or HTTPS port
respectively in http.conf and then not sending or completing a HTTP request
will tie up a HTTP session. By doing this repeatedly until the maximum number
of open HTTP sessions is reached, legitimate requests are blocked.
Additionally, the release of 11.6-cert3, 11.10.1, and 12.3.1 resolves the
following issue:
* AST-2014-006: Permission Escalation via Asterisk Manager User Unauthorized
Shell Access
Manager users can execute arbitrary shell commands with the MixMonitor manager
action. Asterisk does not require system class authorization for a manager
user to use the MixMonitor action, so any manager user who is permitted to use
manager commands can potentially execute shell commands as the user executing
the Asterisk process.
These issues and their resolutions are described in the security advisories.
For more information about the details of these vulnerabilities,
please read security advisories AST-2014-005, AST-2014-006,
AST-2014-007, and AST-2014-008, which were released at the same
time as this announcement.
For a full list of changes in the current releases, please see the ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.10.1
The security advisories are available at:
* http://downloads.asterisk.org/pub/security/AST-2014-006.pdf
* http://downloads.asterisk.org/pub/security/AST-2014-007.pdf
Thank you for your continued support of Asterisk!
-----
The Asterisk Development Team has announced the release of Asterisk 11.10.0.
The release of Asterisk 11.10.0 resolves several issues reported
by the community and would have not been possible without your
participation. Thank you!
The following are the issues resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-23547 - [patch] app_queue removing callers from queue
when reloading (Reported by Italo Rossi)
* ASTERISK-23559 - app_voicemail fails to load after fix to
dialplan functions (Reported by Corey Farrell)
* ASTERISK-22846 - testsuite: masquerade super test fails on all
branches (still) (Reported by Matt Jordan)
* ASTERISK-23545 - Confbridge talker detection settings
configuration load bug (Reported by John Knott)
* ASTERISK-23546 - CB_ADD_LEN does not do what you'd think
(Reported by Walter Doekes)
* ASTERISK-23620 - Code path in app_stack fails to unlock list
(Reported by Bradley Watkins)
* ASTERISK-23616 - Big memory leak in logger.c (Reported by
ibercom)
* ASTERISK-23576 - Build failure on SmartOS / Illumos / SunOS
(Reported by Sebastian Wiedenroth)
* ASTERISK-23550 - Newer sound sets don't show up in menuselect
(Reported by Rusty Newton)
* ASTERISK-18331 - app_sms failure (Reported by David Woodhouse)
* ASTERISK-19465 - P-Asserted-Identity Privacy (Reported by
Krzysztof Chmielewski)
* ASTERISK-23605 - res_http_websocket: Race condition in shutting
down websocket causes crash (Reported by Matt Jordan)
* ASTERISK-23707 - Realtime Contacts: Apparent mismatch between
PGSQL database state and Asterisk state (Reported by Mark
Michelson)
* ASTERISK-23381 - [patch]ChanSpy- Barge only works on the initial
'spy', if the spied-on channel makes a new call, unable to
barge. (Reported by Robert Moss)
* ASTERISK-23665 - Wrong mime type for codec H263-1998 (h263+)
(Reported by Guillaume Maudoux)
* ASTERISK-23664 - Incorrect H264 specification in SDP. (Reported
by Guillaume Maudoux)
* ASTERISK-22977 - chan_sip+CEL: missing ANSWER and PICKUP event
for INVITE/w/replaces pickup (Reported by Walter Doekes)
* ASTERISK-23709 - Regression in Dahdi/Analog/waitfordialtone
(Reported by Steve Davies)
Improvements made in this release:
-----------------------------------
* ASTERISK-23649 - [patch]Support for DTLS retransmission
(Reported by NITESH BANSAL)
* ASTERISK-23564 - [patch]TLS/SRTP status of channel not currently
available in a CLI command (Reported by Patrick Laimbock)
* ASTERISK-23754 - [patch] Use var/lib directory for log file
configured in asterisk.conf (Reported by Igor Goncharovsky)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.10.0
Thank you for your continued support of Asterisk!
-----
The Asterisk Development Team has announced the release of Asterisk 11.9.0.
The release of Asterisk 11.9.0 resolves several issues reported by
the community and would have not been possible without your
participation. Thank you!
The following are the issues resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-22790 - check_modem_rate() may return incorrect rate
for V.27 (Reported by Paolo Compagnini)
* ASTERISK-23034 - [patch] manager Originate doesn't abort on
failed format_cap allocation (Reported by Corey Farrell)
* ASTERISK-23061 - [Patch] 'textsupport' setting not mentioned in
sip.conf.sample (Reported by Eugene)
* ASTERISK-23028 - [patch] Asterisk man pages contains unquoted
minus signs (Reported by Jeremy Lainé)
* ASTERISK-23046 - Custom CDR fields set during a GoSUB called
from app_queue are not inserted (Reported by Denis Pantsyrev)
* ASTERISK-23027 - [patch] Spelling typo "transfered" instead of
"transferred" (Reported by Jeremy Lainé)
* ASTERISK-23008 - Local channels loose CALLERID name when DAHDI
channel connects (Reported by Michael Cargile)
* ASTERISK-23100 - [patch] In chan_mgcp the ident in transmitted
request and request queue may differ - fix for locking (Reported
by adomjan)
* ASTERISK-22988 - [patch]T38 , SIP 488 after Rejecting image
media offer due to invalid or unsupported syntax (Reported by
adomjan)
* ASTERISK-22861 - [patch]Specifying a null time as parameter to
GotoIfTime or ExecIfTime causes segmentation fault (Reported by
Sebastian Murray-Roberts)
* ASTERISK-17837 - extconfig.conf - Maximum Include level (1)
exceeded (Reported by pz)
* ASTERISK-22662 - Documentation fix? - queues.conf says
persistentmembers defaults to yes, it appears to lie (Reported
by Rusty Newton)
* ASTERISK-23134 - [patch] res_rtp_asterisk port selection cannot
handle selinux port restrictions (Reported by Corey Farrell)
* ASTERISK-23220 - STACK_PEEK function with no arguments causes
crash/core dump (Reported by James Sharp)
* ASTERISK-19773 - Asterisk crash on issuing Asterisk-CLI 'reload'
command multiple times on cli_aliases (Reported by Joel Vandal)
* ASTERISK-22757 - segfault in res_clialiases.so on reload when
mapping "module reload" command (Reported by Gareth Blades)
* ASTERISK-17727 - [patch] TLS doesn't get all certificate chain
(Reported by LN)
* ASTERISK-23178 - devicestate.h: device state setting functions
are documented with the wrong return values (Reported by
Jonathan Rose)
* ASTERISK-23232 - LocalBridge AMI Event LocalOptimization value
is opposite to what's expected (Reported by Leon Roy)
* ASTERISK-23098 - [patch]possible null pointer dereference in
format.c (Reported by Marcello Ceschia)
* ASTERISK-23297 - Asterisk 12, pbx_config.so segfaults if
res_parking.so is not loaded, or if res_parking.conf has no
configuration (Reported by CJ Oster)
* ASTERISK-23069 - Custom CDR variable not recorded when set in
macro called from app_queue (Reported by Bryan Anderson)
* ASTERISK-19499 - ConfBridge MOH is not working for transferee
after attended transfer (Reported by Timo Teräs)
* ASTERISK-23261 - [patch]Output mixup in
${CHANNEL(rtpqos,audio,all)} (Reported by rsw686)
* ASTERISK-23279 - [patch]Asterisk doesn't support the dynamic
payload change in rtp mapping in the 200 OK response (Reported
by NITESH BANSAL)
* ASTERISK-23255 - UUID included for Redhat, but missing for
Debian distros in install_prereq script (Reported by Rusty
Newton)
* ASTERISK-23260 - [patch]ForkCDR v option does not keep CDR
variables for subsequent records (Reported by zvision)
* ASTERISK-23141 - Asterisk crashes on Dial(), in
pbx_find_extension at pbx.c (Reported by Maxim)
* ASTERISK-23336 - Asterisk warning "Don't know how to indicate
condition 33 on ooh323c" on outgoing calls from H323 to SIP peer
(Reported by Alexander Semych)
* ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
to minrate=2400, then res_fax refuse to load (Reported by David
Brillert)
* ASTERISK-23135 - Crash - segfault in ast_channel_hangupcause_set
- probably introduced in 11.7.0 (Reported by OK)
* ASTERISK-23323 - [patch]chan_sip: missing p->owner checks in
handle_response_invite (Reported by Walter Doekes)
* ASTERISK-23406 - [patch]Fix typo in "sip show peer" (Reported by
ibercom)
* ASTERISK-23310 - bridged channel crashes in bridge_p2p_rtp_write
(Reported by Jeremy Lainé)
* ASTERISK-22911 - [patch]Asterisk fails to resume WebRTC call
from hold (Reported by Vytis Valentinavičius)
* ASTERISK-23104 - Specifying the SetVar AMI without a Channel
cause Asterisk to crash (Reported by Joel Vandal)
* ASTERISK-21930 - [patch]WebRTC over WSS is not working.
(Reported by John)
* ASTERISK-23383 - Wrong sense test on stat return code causes
unchanged config check to break with include files. (Reported by
David Woolley)
* ASTERISK-20149 - Crash when faxing SIP to SIP with strictrtp set
to yes (Reported by Alexandr Gordeev)
* ASTERISK-17523 - Qualify for static realtime peers does not work
(Reported by Maciej Krajewski)
* ASTERISK-21406 - [patch] chan_sip deadlock on monlock between
unload_module and do_monitor (Reported by Corey Farrell)
* ASTERISK-23373 - [patch]Security: Open FD exhaustion with
chan_sip Session-Timers (Reported by Corey Farrell)
* ASTERISK-23340 - Security Vulnerability: stack allocation of
cookie headers in loop allows for unauthenticated remote denial
of service attack (Reported by Matt Jordan)
* ASTERISK-23311 - Manager - MoH Stop Event fails to show up when
leaving Conference (Reported by Benjamin Keith Ford)
* ASTERISK-23420 - [patch]Memory leak in manager_add_filter
function in manager.c (Reported by Etienne Lessard)
* ASTERISK-23488 - Logic error in callerid checksum processing
(Reported by Russ Meyerriecks)
* ASTERISK-23461 - Only first user is muted when joining
confbridge with 'startmuted=yes' (Reported by Chico Manobela)
* ASTERISK-20841 - fromdomain not honored on outbound INVITE
request (Reported by Kelly Goedert)
* ASTERISK-22079 - Segfault: INTERNAL_OBJ (user_data=0x6374652f)
at astobj2.c:120 (Reported by Jamuel Starkey)
* ASTERISK-23509 - [patch]SayNumber for Polish language tries to
play empty files for numbers divisible by 100 (Reported by
zvision)
* ASTERISK-23103 - [patch]Crash in ast_format_cmp, in ao2_find
(Reported by JoshE)
* ASTERISK-23391 - Audit dialplan function usage of channel
variable (Reported by Corey Farrell)
* ASTERISK-23548 - POST to ARI sometimes returns no body on
success (Reported by Scott Griepentrog)
* ASTERISK-23460 - ooh323 channel stuck if call is placed directly
and gatekeeper is not available (Reported by Dmitry Melekhov)
Improvements made in this release:
-----------------------------------
* ASTERISK-22980 - [patch]Allow building cdr_radius and cel_radius
against libfreeradius-client (Reported by Jeremy Lainé)
* ASTERISK-22661 - Unable to exit ChanSpy if spied channel does
not have a call in progress (Reported by Chris Hillman)
* ASTERISK-23099 - [patch] WSS: enable ast_websocket_read()
function to read the whole available data at first and then wait
for any fragmented packets (Reported by Thava Iyer)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.9.0
Thank you for your continued support of Asterisk!
-----
The Asterisk Development Team has announced security releases for
Certified Asterisk 1.8.15, 11.6, and Asterisk 1.8, 11, and 12. The
available security releases are released as versions 1.8.15-cert5,
11.6-cert2, 1.8.26.1, 11.8.1, and 12.1.1.
The release of these versions resolve the following issues:
* AST-2014-001: Stack overflow in HTTP processing of Cookie headers.
Sending a HTTP request that is handled by Asterisk with a large number of
Cookie headers could overflow the stack.
Another vulnerability along similar lines is any HTTP request with a
ridiculous number of headers in the request could exhaust system memory.
* AST-2014-002: chan_sip: Exit early on bad session timers request
This change allows chan_sip to avoid creation of the channel and
consumption of associated file descriptors altogether if the inbound
request is going to be rejected anyway.
These issues and their resolutions are described in the security advisories.
For more information about the details of these vulnerabilities,
please read security advisories AST-2014-001, AST-2014-002,
AST-2014-003, and AST-2014-004, which were released at the same
time as this announcement.
For a full list of changes in the current releases, please see the ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.8.1
The security advisories are available at:
* http://downloads.asterisk.org/pub/security/AST-2014-001.pdf
* http://downloads.asterisk.org/pub/security/AST-2014-002.pdf
Thank you for your continued support of Asterisk!
-----
The Asterisk Development Team has announced the release of Asterisk 11.8.0.
The release of Asterisk 11.8.0 resolves several issues reported by
the community and would have not been possible without your
participation. Thank you!
The following are the issues resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-22544 - Italian prompt vm-options has advertisement in
it (Reported by Rusty Newton)
* ASTERISK-21383 - STUN Binding Requests Not Being Sent Back from
Asterisk to Chrome (Reported by Shaun Clark)
* ASTERISK-22478 - [patch]Can't use pound(hash) symbol for custom
DTMF menus in ConfBridge (processed as directive) (Reported by
Nicolas Tanski)
* ASTERISK-12117 - chan_sip creates a new local tag (from-tag) for
every register message (Reported by Pawel Pierscionek)
* ASTERISK-20862 - Asterisk min and max member penalties not
honored when set with 0 (Reported by Schmooze Com)
* ASTERISK-22746 - [patch]Crash in chan_dahdi during caller id
read (Reported by Michael Walton)
* ASTERISK-22788 - [patch] main/translate.c: access to variable f
after free in ast_translate() (Reported by Corey Farrell)
* ASTERISK-21242 - Segfault when T.38 re-invite retransmission
receives 200 OK (Reported by Ashley Winters)
* ASTERISK-22590 - BufferOverflow in unpacksms16() when receiving
16 bit multipart SMS with app_sms (Reported by Jan Juergens)
* ASTERISK-22905 - Prevent Asterisk functions that are 'dangerous'
from being executed from external interfaces (Reported by Matt
Jordan)
* ASTERISK-23021 - Typos in code : "avaliable" instead of
"available" (Reported by Jeremy Lainé)
* ASTERISK-22970 - [patch]Documentation fix for QUOTE() (Reported
by Gareth Palmer)
* ASTERISK-21960 - ooh323 channels stuck (Reported by Dmitry
Melekhov)
* ASTERISK-22350 - DUNDI - core dump on shutdown - segfault in
sqlite3_reset from /usr/lib/libsqlite3.so.0 (Reported by Birger
"WIMPy" Harzenetter)
* ASTERISK-22942 - [patch] - Asterisk crashed after
Set(FAXOPT(faxdetect)=t38) (Reported by adomjan)
* ASTERISK-22856 - [patch]SayUnixTime in polish reads minutes
instead of seconds (Reported by Robert Mordec)
* ASTERISK-22854 - [patch] - Deadlock between cel_pgsql unload and
core_event_dispatcher taskprocessor thread (Reported by Etienne
Lessard)
* ASTERISK-22910 - [patch] - REPLACE() calls strcpy on overlapping
memory when <replace-char> is empty (Reported by Gareth Palmer)
* ASTERISK-22871 - cel_pgsql module not loading after "reload" or
"reload cel_pgsql.so" command (Reported by Matteo)
* ASTERISK-23084 - [patch]rasterisk needlessly prints the
AST-2013-007 warning (Reported by Tzafrir Cohen)
* ASTERISK-17138 - [patch] Asterisk not re-registering after it
receives "Forbidden - wrong password on authentication"
(Reported by Rudi)
* ASTERISK-23011 - [patch]configure.ac and pbx_lua don't support
lua 5.2 (Reported by George Joseph)
* ASTERISK-22834 - Parking by blind transfer when lot full orphans
channels (Reported by rsw686)
* ASTERISK-23047 - Orphaned (stuck) channel occurs during a failed
SIP transfer to parking space (Reported by Tommy Thompson)
* ASTERISK-22946 - Local From tag regression with sipgate.de
(Reported by Stephan Eisvogel)
* ASTERISK-23010 - No BYE message sent when sip INVITE is received
(Reported by Ryan Tilton)
* ASTERISK-23135 - Crash - segfault in ast_channel_hangupcause_set
- probably introduced in 11.7.0 (Reported by OK)
Improvements made in this release:
-----------------------------------
* ASTERISK-22728 - [patch] Improve Understanding Of 'Forcerport'
When Running "sip show peers" (Reported by Michael L. Young)
* ASTERISK-22659 - Make a new core and extra sounds release
(Reported by Rusty Newton)
* ASTERISK-22919 - core show channeltypes slicing (Reported by
outtolunc)
* ASTERISK-22918 - dahdi show channels slices PRI channel dnid on
output (Reported by outtolunc)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.8.0
Thank you for your continued support of Asterisk!
Do it for all packages that
* mention perl, or
* have a directory name starting with p5-*, or
* depend on a package starting with p5-
like last time, for 5.18, where this didn't lead to complaints.
Let me know if you have any this time.
The Asterisk Development Team has announced the release of Asterisk 11.7.0.
The release of Asterisk 11.7.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following is a sample of the issues resolved in this release:
* --- app_confbridge: Can now set the language used for announcements
to the conference.
* --- app_queue: Fix CLI "queue remove member" queue_log entry.
* --- chan_sip: Do not increment the SDP version between 183 and 200
responses.
* --- chan_sip: Allow a sip peer to accept both AVP and AVPF calls
* --- chan_sip: Fix Realtime Peer Update Problem When Un-registering
And Expires Header In 200ok
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.7.0
Thank you for your continued support of Asterisk!
AST-2013-006 and AST-2013-007, and a minor bug fix update.
pkgsrc change: disable SRTP on NetBSD as it doesn't link
---- 11.6.1 ----
The Asterisk Development Team has announced security releases for Certified
Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available security
releases are released as versions 1.8.15-cert4, 11.2-cert3, 1.8.24.1, 10.12.4,
10.12.4-digiumphones, and 11.6.1.
The release of these versions resolve the following issues:
* A buffer overflow when receiving odd length 16 bit messages in app_sms. An
infinite loop could occur which would overwrite memory when a message is
received into the unpacksms16() function and the length of the message is an
odd number of bytes.
* Prevent permissions escalation in the Asterisk Manager Interface. Asterisk
now marks certain individual dialplan functions as 'dangerous', which will
inhibit their execution from external sources.
A 'dangerous' function is one which results in a privilege escalation. For
example, if one were to read the channel variable SHELL(rm -rf /) Bad
Things(TM) could happen; even if the external source has only read
permissions.
Execution from external sources may be enabled by setting 'live_dangerously'
to 'yes' in the [options] section of asterisk.conf. Although doing so is not
recommended.
These issues and their resolutions are described in the security advisories.
For more information about the details of these vulnerabilities, please read
security advisories AST-2013-006 and AST-2013-007, which were
released at the same time as this announcement.
For a full list of changes in the current releases, please see the ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.6.1
The security advisories are available at:
* http://downloads.asterisk.org/pub/security/AST-2013-006.pdf
* http://downloads.asterisk.org/pub/security/AST-2013-007.pdf
Thank you for your continued support of Asterisk!
----- 11.6.0 -----
The Asterisk Development Team has announced the release of Asterisk 11.6.0.
The release of Asterisk 11.6.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following is a sample of the issues resolved in this release:
* --- Confbridge: empty conference not being torn down
(Closes issue ASTERISK-21859. Reported by Chris Gentle)
* --- Let Queue wrap up time influence member availability
(Closes issue ASTERISK-22189. Reported by Tony Lewis)
* --- Fix a longstanding issue with MFC-R2 configuration that
prevented users
(Closes issue ASTERISK-21117. Reported by Rafael Angulo)
* --- chan_iax2: Fix saving the wrong expiry time in astdb.
(Closes issue ASTERISK-22504. Reported by Stefan Wachtler)
* --- Fix segfault for certain invalid WebSocket input.
(Closes issue ASTERISK-21825. Reported by Alfred Farrugia)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.6.0
Thank you for your continued support of Asterisk!
AST-2013-004 and AST-2013-005.
The Asterisk Development Team has announced security releases for
Certified Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The
available security rele ases are released as versions 1.8.15-cert2,
11.2-cert2, 1.8.23.1, 10.12.3, 10.12.3-di giumphones, and 11.5.1.
The release of these versions resolve the following issues:
* A remotely exploitable crash vulnerability exists in the SIP
channel driver if an ACK with SDP is received after the channel
has been terminated. The handling code incorrectly assumes that
the channel will always be present.
* A remotely exploitable crash vulnerability exists in the SIP
channel driver if an invalid SDP is sent in a SIP request that
defines media descriptions before connection information. The
handling code incorrectly attempts to reference the socket address
information even though that information has not yet been set.
These issues and their resolutions are described in the security advisories.
For more information about the details of these vulnerabilities,
please read security advisories AST-2013-004 and AST-2013-005,
which were released at the same time as this announcement.
For a full list of changes in the current releases, please see the ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.5.1
The security advisories are available at:
* http://downloads.asterisk.org/pub/security/AST-2013-004.pdf
* http://downloads.asterisk.org/pub/security/AST-2013-005.pdf
Thank you for your continued support of Asterisk!
pkgsrc changes:
- add dependency on libuuid
- work around NetBSD's incompatible implementation of IP_PKTINFO
The Asterisk Development Team has announced the release of Asterisk 11.5.0.
The release of Asterisk 11.5.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following is a sample of the issues resolved in this release:
* --- Fix Segfault In app_queue When "persistentmembers" Is Enabled
And Using Realtime
* --- IAX2: fix race condition with nativebridge transfers.
* --- Fix The Payload Being Set On CN Packets And Do Not Set Marker
Bit
* --- Fix One-Way Audio With auto_* NAT Settings When SIP Calls
Initiated By PBX
* --- chan_sip: NOTIFYs for BLF start queuing up and fail to be sent
out after retries fail
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.5.0
Thank you for your continued support of Asterisk!
- fix compile problem on newer NetBSD systems that have newlocale support
- fix a couple of cases where ctype functions called with plain char
- last two items from joerg@
to address issues with NetBSD-6(and earlier)'s fontconfig not being
new enough for pango.
While doing that, also bump freetype2 dependency to current pkgsrc
version.
Suggested by tron in PR 47882
a) refer 'perl' in their Makefile, or
b) have a directory name of p5-*, or
c) have any dependency on any p5-* package
Like last time, where this caused no complaints.
The Asterisk Development Team has announced the release of Asterisk 11.4.0.
The release of Asterisk 11.4.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following is a sample of the issues resolved in this release:
* --- Fix Sorting Order For Parking Lots Stored In Static Realtime
* --- Fix StopMixMonitor Hanging Up When Unable To Stop MixMonitor On
A Channel
* --- When a session timer expires during a T.38 call, re-invite with
correct SDP
* --- Fix white noise on SRTP decryption
* --- Fix reload skinny with active devices.
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.4.0
Thank you for your continued support of Asterisk!
The Asterisk Development Team has announced the release of Asterisk 11.3.0.
The release of Asterisk 11.3.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following is a sample of the issues resolved in this release:
* --- Fix issue where chan_mobile fails to bind to first available port
* --- Fix Queue Log Reporting Every Call COMPLETECALLER With "h"
Extension Present
* --- Retain XMPP filters across reconnections so external modules
continue to function as expected.
* --- Ensure that a declined media stream is terminated with a '\r\n'
* --- Fix pjproject compilation in certain circumstances
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.3.0
Thank you for your continued support of Asterisk!
AST-2013-001, AST-2013-002, and AST-2013-003.
The Asterisk Development Team has announced security releases for Certified
Asterisk 1.8.15 and Asterisk 1.8, 10, and 11. The available security releases
are released as versions 1.8.15-cert2, 1.8.20.2, 10.12.2, 10.12.2-digiumphones,
and 11.2.2.
The release of these versions resolve the following issues:
* A possible buffer overflow during H.264 format negotiation. The format
attribute resource for H.264 video performs an unsafe read against a media
attribute when parsing the SDP.
This vulnerability only affected Asterisk 11.
* A denial of service exists in Asterisk's HTTP server. AST-2012-014, fixed
in January of this year, contained a fix for Asterisk's HTTP server for a
remotely-triggered crash. While the fix prevented the crash from being
triggered, a denial of service vector still exists with that solution if an
attacker sends one or more HTTP POST requests with very large Content-Length
values.
This vulnerability affects Certified Asterisk 1.8.15, Asterisk 1.8, 10, and 11
* A potential username disclosure exists in the SIP channel driver. When
authenticating a SIP request with alwaysauthreject enabled, allowguest
disabled, and autocreatepeer disabled, Asterisk discloses whether a user
exists for INVITE, SUBSCRIBE, and REGISTER transactions in multiple ways.
This vulnerability affects Certified Asterisk 1.8.15, Asterisk 1.8, 10, and 11
These issues and their resolutions are described in the security advisories.
For more information about the details of these vulnerabilities, please read
security advisories AST-2013-001, AST-2013-002, and AST-2013-003, which were
released at the same time as this announcement.
For a full list of changes in the current releases, please see the ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.2.2
The security advisories are available at:
* http://downloads.asterisk.org/pub/security/AST-2013-001.pdf
* http://downloads.asterisk.org/pub/security/AST-2013-002.pdf
* http://downloads.asterisk.org/pub/security/AST-2013-003.pdf
Thank you for your continued support of Asterisk!
----- 11.2.1:
The Asterisk Development Team has announced the release of Asterisk 11.2.1.
The release of Asterisk 11.2.1 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
* --- Fix astcanary startup problem due to wrong pid value from before
daemon call
* --- Update init.d scripts to handle stderr; readd splash screen for
remote consoles
* --- Reset RTP timestamp; sequence number on SSRC change
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.2.1
Thank you for your continued support of Asterisk!
----- 11.2.0:
The Asterisk Development Team has announced the release of Asterisk 11.2.0.
The release of Asterisk 11.2.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following is a sample of the issues resolved in this release:
* --- app_meetme: Fix channels lingering when hung up under certain
conditions
* --- Fix stuck DTMF when bridge is broken.
* --- Add missing support for "who hung up" to chan_motif.
* --- Remove a fixed size limitation for producing SDP and change how
ICE support is disabled by default.
* --- Fix chan_sip websocket payload handling
* --- Fix pjproject compilation in certain circumstances
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.2.0
Thank you for your continued support of Asterisk!
and AST-2012-015. Apparently the last update didn't completely
fix the issues.
The Asterisk Development Team has announced a security release for
Asterisk 11, Asterisk 11.1.2. This release addresses the security
vulnerabilities reported in AST-2012-014 and AST-2012-015, and
replaces the previous version of Asterisk 11 released for these
security vulnerabilities. The prior release left open a vulnerability
in res_xmpp that exists only in Asterisk 11; as such, other versions
of Asterisk were resolved correctly by the previous releases.
The release of these versions resolve the following two issues:
* Stack overflows that occur in some portions of Asterisk that manage a TCP
connection. In SIP, this is exploitable via a remote unauthenticated session;
in XMPP and HTTP connections, this is exploitable via remote authenticated
sessions. The vulnerabilities in SIP and HTTP were corrected in a prior
release of Asterisk; the vulnerability in XMPP is resolved in this release.
* A denial of service vulnerability through exploitation of the device state
cache. Anonymous calls had the capability to create devices in Asterisk that
would never be disposed of. Handling the cachability of device states
aggregated via XMPP is handled in this release.
These issues and their resolutions are described in the security advisories.
For more information about the details of these vulnerabilities, please read
security advisories AST-2012-014 and AST-2012-015.
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.1.2
The security advisories are available at:
* http://downloads.asterisk.org/pub/security/AST-2012-014.pdf
* http://downloads.asterisk.org/pub/security/AST-2012-015.pdf
Thank you for your continued support of Asterisk - and we apologize for having
to do this twice!
