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The Asterisk Development Team has announced the release of Asterisk 1.8.31.0.
The release of Asterisk 1.8.31.0 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-24019 - When a Music On Hold stream starts it restarts
at beginning of file. (Reported by Jason Richards)
* ASTERISK-24211 - testsuite: Fix the dial_LS_options test
(Reported by Matt Jordan)
* ASTERISK-24249 - SIP debugs do not stop (Reported by Avinash
Mohod)
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-1.8.31.0
Thank you for your continued support of Asterisk!
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numerous general bugs. The vulnerabilities fixed are: AST-2014-001,
AST-2014-002, 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.
The security patches for AST-2014-007 have been updated with the
fix for the regression, and are available at
http://downloads.asterisk.org/pub/security
Please note that the release of these versions resolves the following security
vulnerabilities:
* 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-1.8.28.2
The security advisories are available at:
* 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.
These releases are available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases
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.
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-1.8.28.1
The security advisories are available at:
* 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 1.8.28.0.
The release of Asterisk 1.8.28.0 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-22846 - testsuite: masquerade super test fails on all
branches (still) (Reported by Matt Jordan)
* 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-18331 - app_sms failure (Reported by David Woodhouse)
* ASTERISK-19465 - P-Asserted-Identity Privacy (Reported by
Krzysztof Chmielewski)
* ASTERISK-23707 - Realtime Contacts: Apparent mismatch between
PGSQL database state and Asterisk state (Reported by Mark
Michelson)
* ASTERISK-23665 - Wrong mime type for codec H263-1998 (h263+)
(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)
* ASTERISK-23650 - Intermittent segfault in string functions
(Reported by Roel van Meer)
Improvements made in this release:
-----------------------------------
* 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-1.8.28.0
Thank you for your continued support of Asterisk!
-----
The Asterisk Development Team has announced the release of Asterisk 1.8.27.0.
The release of Asterisk 1.8.27.0 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-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-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-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-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-23382 - [patch]Build System: make -qp can corrupt
menuselect-tree and related files (Reported by Corey Farrell)
* 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-23104 - Specifying the SetVar AMI without a Channel
cause Asterisk to crash (Reported by Joel Vandal)
* ASTERISK-23383 - Wrong sense test on stat return code causes
unchanged config check to break with include files. (Reported by
David Woolley)
* 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-23488 - Logic error in callerid checksum processing
(Reported by Russ Meyerriecks)
* 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-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)
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)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.27.0
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-1.8.26.1
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 1.8.26.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 1.8.26.0 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-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-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)
Improvements made in this release:
-----------------------------------
* ASTERISK-22659 - Make a new core and extra sounds release
(Reported by Rusty Newton)
* 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-1.8.26.0
Thank you for your continued support of Asterisk!
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These are now handled dynamically if INIT_SYSTEM is set to "rc.d", or
ignored otherwise.
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pkgsrc changes:
- add work around for NetBSD's incompatible implementation of IP_PKTINFO
- core sounds package was updated to 1.4.24
The Asterisk Development Team has announced the release of Asterisk 1.8.23.0.
The release of Asterisk 1.8.23.0 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 a memory copying bug in slinfactory which was causing
mixmonitor issues.
* --- IAX2: fix race condition with nativebridge transfers.
* --- Fix crash in chan_sip when a core initiated op occurs at the
same time as a BYE
* --- Fix The Payload Being Set On CN Packets And Do Not Set Marker
Bit
* --- chan_sip: Session-Expires: Set timer to correctly expire at
(~2/3) of the interval when not the refresher
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.23.0
Thank you for your continued support of Asterisk!
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The Asterisk Development Team has announced the release of Asterisk 1.8.21.0.
The release of Asterisk 1.8.21.0 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 station ringback; trunk hangup issues in SLA
* --- Fix Queue Log Reporting Every Call COMPLETECALLER With "h"
Extension Present
* --- Fix Record-Route parsing for large headers.
* --- Fix AMI redirect action with two channels failing to redirect
both channels.
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.21.0
Thank you for your continued support of Asterisk!
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This is the second attempt to fix the build problem that some people
have seen (I have received inconsistent reports). This should
force chan_mgcp to build on systems where it can. It was tested
on NetBSD 5.0, thus ensuring that it doesn't break previously
working systems; and NetBSD 6.99.7, where I finally saw the problem
that some people were reporting.
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being built and others weren't by detecting the situation when it
would be built and adjusting the PLIST accordingly.
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- this package is marked OWNER= for a reason!
- need to figure out why chan_mgcp is built only in some situations
instead of adding gross hacks
- upgrade to Asterisk 1.8.14.1: this is a bugfix release
The release of Asterisk 1.8.14.1 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!
The following is the issue resolved in this release:
* --- Remove a superfluous and dangerous freeing of an SSL_CTX.
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.14.1
Thank you for your continued support of Asterisk!
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Defined new PLIST.mgcp variable for new file:
lib/asterisks/modules/chan_mgcp.so
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The Asterisk Development Team has announced the release of Asterisk
1.8.13.0.
The release of Asterisk 1.8.13.0 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:
* --- Turn off warning message when bind address is set to any.
* --- Prevent overflow in calculation in ast_tvdiff_ms on 32-bit
machines
* --- Make DAHDISendCallreroutingFacility wait 5 seconds for a reply
before disconnecting the call.
* --- Fix recalled party B feature flags for a failed DTMF atxfer.
* --- Fix DTMF atxfer running h exten after the wrong bridge ends.
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.13.0
Thank you for your continued support of Asterisk!
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pkgsrc change: eliminate ilbc option now that the iLBC codec is always built
The Asterisk Development Team has announced the release of Asterisk 1.8.11.0.
The release of Asterisk 1.8.11.0 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:
* --- Fix potential buffer overrun and memory leak when executing "sip
show peers"
* --- Fix ACK routing for non-2xx responses.
* --- Remove possible segfaults from res_odbc by adding locks around
usage of odbc handle
* --- Fix blind transfer parking issues if the dialed extension is not
recognized as a parking extension.
* --- Copy CDR variables when set during a bridge
* --- push 'outgoing' flag from sig_XXX up to chan_dahdi
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.11.0
Thank you for your continued support of Asterisk!
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pkgsrc changes: adapt to having iLBC coded included in the asterisk
tarball and newer version of sounds tarball.
----- 1.8.10.0 -----
The Asterisk Development Team has announced the release of Asterisk 1.8.10.0.
The release of Asterisk 1.8.10.0 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:
* --- Prevent outbound SIP NOTIFY packets from displaying a port of 0 ---
* --- Include iLBC source code for distribution with Asterisk ---
* --- Fix callerid of originated calls ---
* --- Fix outbound DTMF for inband mode of chan_ooh323 ---
* --- Create and initialize udptl only when dialog requests image media ---
* --- Don't prematurely stop SIP session timer ---
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.10.0
Thank you for your continued support of Asterisk!
----- 1.8.10.1 -----
The Asterisk Development Team has announced security releases for
Asterisk 1.4, 1.6.2, 1.8, and 10. The available security releases
are released as versions 1.4.44, 1.6.2.23, 1.8.10.1, and 10.2.1.
The release of Asterisk 1.8.10.1 and 10.2.1 resolve two issues.
First, they resolve the issue in app_milliwatt, wherein a buffer
can potentially be overrun on the stack, but no remote code execution
is possible. Second, they resolve an issue in HTTP AMI where digest
authentication information can be used to overrun a buffer on the
stack, allowing for code injection and execution.
These issues and their resolution are described in the security
advisory.
For more information about the details of these vulnerabilities,
please read the security advisories AST-2012-002 and AST-2012-003,
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-1.8.10.1
The security advisories are available at:
* http://downloads.asterisk.org/pub/security/AST-2012-002.pdf
* http://downloads.asterisk.org/pub/security/AST-2012-003.pdf
Thank you for your continued support of Asterisk!
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The Asterisk Development Team is pleased to announce the release of
Asterisk 1.8.9.0. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.8.9.0 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:
* AST-2012-001: prevent crash when an SDP offer
is received with an encrypted video stream when support for video
is disabled and res_srtp is loaded. (closes issue ASTERISK-19202)
Reported by: Catalin Sanda
* Handle AST_CONTROL_UPDATE_RTP_PEER frames in local bridge loop. Failing
to handle AST_CONTROL_UPDATE_RTP_PEER frames in the local bridge loop
causes the loop to exit prematurely. This causes a variety of negative side
effects, depending on when the loop exits. This patch handles the frame by
essentially swallowing the frame in the local loop, as the current channel
drivers expect the RTP bridge to handle the frame, and, in the case of the
local bridge loop, no additional action is necessary.
(closes issue ASTERISK-19095) Reported by: Stefan Schmidt Tested
by: Matt Jordan
* Fix timing source dependency issues with MOH. Prior to this patch,
res_musiconhold existed at the same module priority level as the timing
sources that it depends on. This would cause a problem when music on
hold was reloaded, as the timing source could be changed after
res_musiconhold was processed. This patch adds a new module priority
level, AST_MODPRI_TIMING, that the various timing modules are now loaded
at. This now occurs before loading other resource modules, such
that the timing source is guaranteed to be set prior to resolving
the timing source dependencies.
(closes issue ASTERISK-17474) Reporter: Luke H Tested by: Luke H,
Vladimir Mikhelson, zzsurf, Wes Van Tlghem, elguero, Thomas Arimont
Patched by elguero
* Fix RTP reference leak. If a blind transfer were initiated using a
REFER without a prior reINVITE to place the call on hold, AND if Asterisk
were sending RTCP reports, then there was a reference leak for the
RTP instance of the transferrer.
(closes issue ASTERISK-19192) Reported by: Tyuta Vitali
* Fix blind transfers from failing if an 'h' extension
is present. This prevents the 'h' extension from being run on the
transferee channel when it is transferred via a native transfer
mechanism such as SIP REFER. (closes issue ASTERISK-19173) Reported
by: Ross Beer Tested by: Kristjan Vrban Patches: ASTERISK-19173 by
Mark Michelson (license 5049)
* Restore call progress code for analog ports. Extracting sig_analog
from chan_dahdi lost call progress detection functionality. Fix
analog ports from considering a call answered immediately after
dialing has completed if the callprogress option is enabled.
(closes issue ASTERISK-18841)
Reported by: Richard Miller Patched by Richard Miller
* Fix regression that 'rtp/rtcp set debup ip' only works when a port
was also specified.
(closes issue ASTERISK-18693) Reported by: Davide Dal Reviewed by:
Walter Doekes
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.9.0
Thank you for your continued support of Asterisk!
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share/doc/asterisk/AST.{txt,pdf} has been replaced with
share/doc/asterisk/Asterisk_Admin_Guide. You will need a browser
to read the latter.
----- Asterisk 1.8.8.1 -----
The release of Asterisk 1.8.8.1 resolves a regression introduced
in Asterisk 1.8.8.0 reported by the community, and would have not
been possible without your participation. Thank you!
The following is the issue resolved in this release:
* Handle AST_CONTROL_UPDATE_RTP_PEER frames in local bridge loop
Failing to handle AST_CONTROL_UPDATE_RTP_PEER frames in the local
bridge loop causes the loop to exit prematurely. This causes a
variety of negative side effects, which may include having Music
On Hold failing during a SIP Hold.
For a full description of the changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.8.1
Thank you for your continued support of Asterisk!
----- Asterisk 1.8.8.0 -----
The release of Asterisk 1.8.8.0 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:
* Updated SIP 484 handling; added Incomplete control frame
When a SIP phone uses the dial application and receives a 484
Address Incomplete response, if overlapped dialing is enabled
for SIP, then the 484 Address Incomplete is forwarded back to
the SIP phone and the HANGUPCAUSE channel variable is set to
28. Previously, the Incomplete application dialplan logic was
automatically triggered; now, explicit dialplan usage of the
application is required.
* Prevent IAX2 from getting IPv6 addresses via DNS
IAX2 does not support IPv6 and getting such addresses from DNS
can cause error messages on the remote end involving bad IPv4
address casts in the presence of IPv6/IPv4 tunnels.
* Fix bad RTP media bridges in directmedia calls on peers separated by
multiple Asterisk nodes.
* Fix crashes in ast_rtcp_write()
* Fix for incorrect voicemail duration in external notifications.
This patch fixes an issue where the voicemail duration was being
reported with a duration significantly less than the actual
sound file duration.
* Prevent segfault if call arrives before Asterisk is fully booted.
* Fix remote Crash Vulnerability in SIP channel driver (AST-2011-012)
http://downloads.asterisk.org/pub/security/AST-2011-012.pdf
* Fix locking order in app_queue.c which caused deadlocks
* Fix regression in configure script for libpri capability checks
* Prevent BLF subscriptions from causing deadlocks.
* Fix deadlock if peer is destroyed while sending MWI notice.
* Fix issue with setting defaultenabled on categories that are already
enabled by default.
* Don't crash on INFO automon request with no channel
AST-2011-014. When automon was enabled in features.conf, it
was possible to crash Asterisk by sending an INFO request if
no channel had been created yet.
* Fixed crash from orphaned MWI subscriptions in chan_sip
This patch resolves the issue where MWI subscriptions are orphaned
by subsequent SIP SUBSCRIBE messages.
* Default to nat=yes; warn when nat in general and peer differ
AST-2011-013. It is possible to enumerate SIP usernames when
the general and user/peer nat settings differ in whether to
respond to the port a request is sent from or the port listed
for responses in the Via header.
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.8.0
Thank you for your continued support of Asterisk!
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pkgsrc change: now what sqlite3 has been imported into NetBSD, enable it
Asterisk Project Security Advisory - AST-2011-012
Product Asterisk
Summary Remote crash vulnerability in SIP channel driver
Nature of Advisory Remote crash
Susceptibility Remote authenticated sessions
Severity Critical
Exploits Known No
Reported On October 4, 2011
Reported By Ehsan Foroughi
Posted On October 17, 2011
Last Updated On October 17, 2011
Advisory Contact Terry Wilson <twilson@digium.com>
CVE Name CVE-2011-4063
Description A remote authenticated user can cause a crash with a
malformed request due to an unitialized variable.
Resolution Ensure variables are initialized in all cases when parsing
the request.
Affected Versions
Product Release Series
Asterisk Open Source 1.8.x All versions
Asterisk Open Source 10.x All versions (currently in beta)
Corrected In
Product Release
Asterisk Open Source 1.8.7.1, 10.0.0-rc1
Patches
Download URL Revision
http://downloads.asterisk.org/pub/security/AST-2011-012-1.8.diff 1.8
http://downloads.asterisk.org/pub/security/AST-2011-012-10.diff 10
Links
Asterisk Project Security Advisories are posted at
http://www.asterisk.org/security
This document may be superseded by later versions; if so, the latest
version will be posted at
http://downloads.digium.com/pub/security/AST-2011-012.pdf and
http://downloads.digium.com/pub/security/AST-2011-012.html
Revision History
Date Editor Revisions Made
Asterisk Project Security Advisory - AST-2011-012
Copyright (c) 2011 Digium, Inc. All Rights Reserved.
Permission is hereby granted to distribute and publish this advisory in its
original, unaltered form.
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This update adds a "jabber" option which is enabled by default.
This option pulls in iksemel which is used by the res_jabber.
Doing this allows chan_jingle (jabber) and chan_gtalk to work.
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pkgsrc changes:
- adjust for ilbc changes after it was acquired by Google
- install AST.pdf IAX2-security.pdf into share/doc/asterisk
1.8.7.0:
========
The release of Asterisk 1.8.7.0 resolves several issues reported
by the community and would have not been possible without your
participation. Thank you!
Please note that a significant numbers of changes and fixes have
gone into features.c in this release (call parking, built-in
transfers, call pickup, etc.).
NOTE:
Recently, we were notified that the mechanism included in our
Asterisk source code releases to download and build support for
the iLBC codec had stopped working correctly; a little investigation
revealed that this occurred because of some changes on the
ilbcfreeware.org website. These changes occurred as a result of
Google's acquisition of GIPS, who produced (and provided licenses
for) the iLBC codec.
If you are a user of Asterisk and iLBC together, and you've already
executed a license agreement with GIPS, we believe you can continue
using iLBC with Asterisk. If you are a user of Asterisk and iLBC
together, but you had not executed a license agreement with GIPS,
we encourage you to research the situation and consult with your
own legal representatives to determine what actions you may want
to take (or avoid taking).
More information is available on the Asterisk blog:
http://blogs.asterisk.org/2011/09/19/ilbc-support-in-asterisk-after-googles-acquisition-of-gips/
The following is a sample of the issues resolved in this release:
* Added the 'storesipcause' option to sip.conf to allow the user to
disable the setting of HASH(SIP_CAUSE,) on the channel. Having
chan_sip set HASH(SIP_CAUSE,) on the channel carries a significant
performance penalty because of the usage of the MASTER_CHANNEL()
dialplan function.
We've decided to disable this feature by default in future 1.8
versions. This would be an unexpected behavior change for anyone
depending on that SIP_CAUSE update in their dialplan. Please
refer to the asterisk-dev mailing list more information:
http://lists.digium.com/pipermail/asterisk-dev/2011-August/050626.html
* Significant fixes and improvements to parking lots.
(Closes issues ASTERISK-17183, ASTERISK-17870, ASTERISK-17430,
ASTERISK-17452, ASTERISK-17452, ASTERISK-15792.)
* Numerous issues have been reported for deadlocks that are caused
by a blocking read in res_timing_timerfd on a file descriptor
that will never be written to.
A change to Asterisk adds some checks to make sure that the
timerfd is both valid and armed before calling read(). Should
fix: ASTERISK-18142, ASTERISK-18197, ASTERISK-18166 and possibly
others. (In essence, this change should make res_timing_timerfd
usable.)
* Resolve segfault when publishing device states via XMPP and not connected.
(Closes issue ASTERISK-18078.)
* Refresh peer address if DNS unavailable at peer creation.
(Closes issue ASTERISK-18000)
* Fix the missing DAHDI channels when using the newer chan_dahdi.conf
sections for channel configuration.
(Closes issue ASTERISK-18496.)
* Remove unnecessary libpri dependency checks in the configure script.
(Closes issue ASTERISK-18535.)
* Update get_ilbc_source.sh script to work again.
(Closes issue ASTERISK-18412)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.7.0
Thank you for your continued support of Asterisk!
1.8.6.0:
========
The release of Asterisk 1.8.6.0 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:
* Fix an issue with Music on Hold classes losing files in playlist
when realtime is used. (Closes issue ASTERISK-17875.)
* Resolve a potential crash in chan_sip when utilizing auth= and
performing a 'sip reload' from the console. (Closes issue
ASTERISK-17939.)
* Address some improper sql statements in res_odbc that would cause
an update to fail on realtime peers due to trying to set as
"(NULL)" rather than an actual NULL. (Closes issue ASTERISK-17791.)
* Resolve issue where 403 Forbidden would always be sent maximum
number of times regardless to receipt of ACK.
* Resolve issue where if a call to MeetMe includes both the dynamic(D)
and always request PIN(P) options, MeetMe will ask for the PIN
two times: once for creating the conference and once for entering
the conference.
* Fix New Zealand indications profile based on
http://www.telepermit.co.nz/TNA102.pdf
(Closes issue ASTERISK-16263.)
* Segfault in shell_helper in func_shell.c
(Closes issue ASTERISK-18109.)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.6.0
Thank you for your continued support of Asterisk!
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The release of Asterisk 1.8.5.0 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 Deadlock with attended transfer of SIP call
* Fixes thread blocking issue in the sip TCP/TLS implementation.
* Be more tolerant of what URI we accept for call completion PUBLISH requests.
* Fix a nasty chanspy bug which was causing a channel leak every time a spied on
channel made a call.
* This patch fixes a bug with MeetMe behavior where the 'P' option for always
prompting for a pin is ignored for the first caller.
* Fix issue where Asterisk does not hangup a channel after endpoint hangs up. If
the call that the dialplan started an AGI script for is hungup while the AGI
script is in the middle of a command then the AGI script is not notified of
the hangup.
* Resolve issue where leaving a voicemail, the MWI message is never sent. The
same thing happens when checking a voicemail and marking it as read.
* Resolve issue where wait for leader with Music On Hold allows crosstalk
between participants. Parenthesis in the wrong position. Regression from issue
#14365 when expanding conference flags to use 64 bits.
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.5.0
Thank you for your continued support of Asterisk!
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Asterisk Project Security Advisory - AST-2011-011
+------------------------------------------------------------------------+
| Product | Asterisk |
|--------------------+---------------------------------------------------|
| Summary | Possible enumeration of SIP users due to |
| | differing authentication responses |
|--------------------+---------------------------------------------------|
| Nature of Advisory | Unauthorized data disclosure |
|--------------------+---------------------------------------------------|
| Susceptibility | Remote unauthenticated sessions |
|--------------------+---------------------------------------------------|
| Severity | Moderate |
|--------------------+---------------------------------------------------|
| Exploits Known | No |
|--------------------+---------------------------------------------------|
| CVE Name | CVE-2011-2536 |
+------------------------------------------------------------------------+
+------------------------------------------------------------------------+
| Description | Asterisk may respond differently to SIP requests from an |
| | invalid SIP user than it does to a user configured on |
| | the system, even when the alwaysauthreject option is set |
| | in the configuration. This can leak information about |
| | what SIP users are valid on the Asterisk system. |
+------------------------------------------------------------------------+
+------------------------------------------------------------------------+
| Resolution | Respond to SIP requests from invalid and valid SIP users |
| | in the same way. Asterisk 1.4 and 1.6.2 do not respond |
| | identically by default due to backward-compatibility |
| | reasons, and must have alwaysauthreject=yes set in |
| | sip.conf. Asterisk 1.8 defaults to alwaysauthreject=yes. |
| | |
| | IT IS ABSOLUTELY IMPERATIVE that users of Asterisk 1.4 |
| | and 1.6.2 set alwaysauthreject=yes in the general section |
| | of sip.conf. |
+------------------------------------------------------------------------+
|
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AST-2011-002, AST-2011-003, AST-2011-004, AST-2011-005, AST-2011-006,
and AST-2011-007.
pkgsrc changes:
- add patch for autosupport script; == -> =
- patch configure to not unconditionally set PBX_LAUNCHD=1
- this allows res_timing_kqueue.so to build
This last change brings a timing source to NetBSD which allows IAX
trunking and allows the bridging modules to work, a rather major
piece that was missing. Note that I haven't extensively tested
it. But, have at it...
===========================================================================
1.8.4.2:
The Asterisk Development Team has announced the release of Asterisk
version 1.8.4.2, which is a security release for Asterisk 1.8.
The release of Asterisk 1.8.4.2 resolves an issue with SIP URI parsing
which can lead to a remotely exploitable crash:
Remote Crash Vulnerability in SIP channel driver (AST-2011-007)
The issue and resolution is described in the AST-2011-007 security
advisory.
For more information about the details of this vulnerability, please
read the security advisory AST-2011-007, which was released at the same
time as this announcement.
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.4.2
Security advisory AST-2011-007 is available at:
http://downloads.asterisk.org/pub/security/AST-2011-007.pdf
===========================================================================
1.8.4.1:
The Asterisk Development Team has announced the release of Asterisk 1.8.4.1.
The release of Asterisk 1.8.4.1 resolves several issues reported by the
community. Without your help this release would not have been possible.
Thank you!
Below is a list of issues resolved in this release:
* Fix our compliance with RFC 3261 section 18.2.2. (aka Cisco phone fix)
* Resolve a change in IPv6 header parsing due to the Cisco phone fix issue.
This issue was found and reported by the Asterisk test suite.
* Resolve potential crash when using SIP TLS support.
* Improve reliability when using SIP TLS.
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.4.1
===========================================================================
1.8.4:
The Asterisk Development Team has announced the release of Asterisk 1.8.4.
The release of Asterisk 1.8.4 resolves several issues reported by the community.
Without your help this release would not have been possible. Thank you!
Below is a sample of the issues resolved in this release:
* Use SSLv23_client_method instead of old SSLv2 only.
* Resolve crash in ast_mutex_init()
* Resolution of several DTMF based attended transfer issues.
NOTE: Be sure to read the ChangeLog for more information about these changes.
* Resolve deadlocks related to device states in chan_sip
* Resolve an issue with the Asterisk manager interface leaking memory when
disabled.
* Support greetingsfolder as documented in voicemail.conf.sample.
* Fix channel redirect out of MeetMe() and other issues with channel softhangup
* Fix voicemail sequencing for file based storage.
* Set hangup cause in local_hangup so the proper return code of 486 instead of
503 when using Local channels when the far sides returns a busy. Also affects
CCSS in Asterisk 1.8+.
* Fix issues with verbose messages not being output to the console.
* Fix Deadlock with attended transfer of SIP call
Includes changes per AST-2011-005 and AST-2011-006
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.4
Information about the security releases are available at:
http://downloads.asterisk.org/pub/security/AST-2011-005.pdf
http://downloads.asterisk.org/pub/security/AST-2011-006.pdf
===========================================================================
1.8.3.3:
The Asterisk Development Team has announced security releases for Asterisk
branches 1.4, 1.6.1, 1.6.2, and 1.8. The available security releases are
released as versions 1.4.40.1, 1.6.1.25, 1.6.2.17.3, and 1.8.3.3.
The releases of Asterisk 1.4.40.1, 1.6.1.25, 1.6.2.17.3, and 1.8.3.3 resolve two
issues:
* File Descriptor Resource Exhaustion (AST-2011-005)
* Asterisk Manager User Shell Access (AST-2011-006)
The issues and resolutions are described in the AST-2011-005 and AST-2011-006
security advisories.
For more information about the details of these vulnerabilities, please read the
security advisories AST-2011-005 and AST-2011-006, which were released at the
same time as this announcement.
For a full list of changes in the current releases, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.3.3
Security advisory AST-2011-005 and AST-2011-006 are available at:
http://downloads.asterisk.org/pub/security/AST-2011-005.pdf
http://downloads.asterisk.org/pub/security/AST-2011-006.pdf
===========================================================================
1.8.3.2:
he Asterisk Development Team has announced security releases for Asterisk
branches 1.6.1, 1.6.2, and 1.8. The available security releases are
released as versions 1.6.1.24, 1.6.2.17.2, and 1.8.3.2.
** This is a re-release of Asterisk 1.6.1.23, 1.6.2.17.1 and 1.8.3.1 which
contained a bug which caused duplicate manager entries (issue #18987).
The releases of Asterisk 1.6.1.24, 1.6.2.17.2, and 1.8.3.2 resolve two issues:
* Resource exhaustion in Asterisk Manager Interface (AST-2011-003)
* Remote crash vulnerability in TCP/TLS server (AST-2011-004)
The issues and resolutions are described in the AST-2011-003 and AST-2011-004
security advisories.
For more information about the details of these vulnerabilities, please read the
security advisories AST-2011-003 and AST-2011-004, which were released at the
same time as this announcement.
For a full list of changes in the current releases, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.3.2
Security advisory AST-2011-003 and AST-2011-004 are available at:
http://downloads.asterisk.org/pub/security/AST-2011-003.pdf
http://downloads.asterisk.org/pub/security/AST-2011-004.pdf
===========================================================================
1.8.3.1:
The Asterisk Development Team has announced security releases for Asterisk
branches 1.6.1, 1.6.2, and 1.8. The available security releases are
released as versions 1.6.1.23, 1.6.2.17.1, and 1.8.3.1.
The releases of Asterisk 1.6.1.23, 1.6.2.17.1, and 1.8.3.1 resolve two issues:
* Resource exhaustion in Asterisk Manager Interface (AST-2011-003)
* Remote crash vulnerability in TCP/TLS server (AST-2011-004)
The issues and resolutions are described in the AST-2011-003 and AST-2011-004
security advisories.
For more information about the details of these vulnerabilities, please read the
security advisories AST-2011-003 and AST-2011-004, which were released at the
same time as this announcement.
For a full list of changes in the current releases, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.3.1
Security advisory AST-2011-003 and AST-2011-004 are available at:
http://downloads.asterisk.org/pub/security/AST-2011-003.pdf
http://downloads.asterisk.org/pub/security/AST-2011-004.pdf
===========================================================================
1.8.3:
The Asterisk Development Team has announced the release of Asterisk 1.8.3.
The release of Asterisk 1.8.3 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:
* Resolve duplicated data in the AstDB when using DIALGROUP()
* Ensure the ipaddr field in realtime is large enough to handle IPv6 addresses.
* Reworking parsing of mwi => lines to resolve a segfault. Also add a set of
unit tests for the function that does the parsing.
* When using cdr_pgsql the billsec field was not populated correctly on
unanswered calls.
* Resolve memory leak in iCalendar and Exchange calendaring modules.
* This version of Asterisk includes the new Compiler Flags option
BETTER_BACKTRACES which uses libbfd to search for better symbol information
within both the Asterisk binary, as well as loaded modules, to assist when
using inline backtraces to track down problems.
* Resolve issue where no Music On Hold may be triggered when using
res_timing_dahdi.
* Resolve a memory leak when the Asterisk Manager Interface is disabled.
* Reimplemented fax session reservation to reverse the ABI breakage introduced
in r297486.
* Fix regression that changed behavior of queues when ringing a queue member.
* Resolve deadlock involving REFER.
Additionally, this release has the changes related to security bulletin
AST-2011-002 which can be found at
http://downloads.asterisk.org/pub/security/AST-2011-002.pdf
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.3
===========================================================================
1.8.2.4:
The Asterisk Development Team has announced security releases for Asterisk
branches 1.4, 1.6.1, 1.6.2, and 1.8. The available security releases are
released as versions 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4.
The releases of Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 resolve an
issue that when decoding UDPTL packets, multiple stack and heap based arrays can
be made to overflow by specially crafted packets. Systems configured for
T.38 pass through or termination are vulnerable. The issue and resolution are
described in the AST-2011-002 security advisory.
For more information about the details of this vulnerability, please read the
security advisory AST-2011-002, which was released at the same time as this
announcement.
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.2.4
Security advisory AST-2011-002 is available at:
http://downloads.asterisk.org/pub/security/AST-2011-002.pdf
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to enable res_fax_spandsp.so. Don't bother with a PKGREVISION bump since
this doesn't change default builds and there is no need tobother people
that don't need the option.
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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 1.8 is a long term support version (i.e. it will be
supported for four years with an additional year of security only
fixes). See:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
What's new:
Asterisk 1.8 is the next major release series of Asterisk.
The release of Asterisk 1.8.0 would not have been possible without the support
and contributions of the community. Since Asterisk 1.6.2, we've had over 500
reporters, more than 300 testers and greater than 200 developers contributed to
this release.
You can find a summary of the work involved with the 1.8.0 release in the
sumary:
http://svn.asterisk.org/svn/asterisk/tags/1.8.0/asterisk-1.8.0-summary.txt
A short list of available features includes:
* Secure RTP
* IPv6 Support in the SIP channel driver
* Connected Party Identification Support
* Calendaring Integration
* A new call logging system, Channel Event Logging (CEL)
* Distributed Device State using Jabber/XMPP PubSub
* Call Completion Supplementary Services support
* Advice of Charge support
* Much, much more!
A full list of new features can be found in the CHANGES file.
http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=markup
For a full list of changes in the current release candidate, please see the
ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0
-----
The Asterisk Development Team has announced the release of Asterisk 1.8.1.
The release of Asterisk 1.8.1 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 when using directmedia. Asterisk needs to limit the codecs offered
to just the ones that both sides recognize, otherwise they may end up sending
audio that the other side doesn't understand.
(Closes issue #17403. Reported, patched by one47. Tested by one47, falves11)
* Resolve issue where Party A in an analog 3-way call would continue to hear
ringback after party C answers.
(Patched by rmudgett)
* Fix playback failure when using IAX with the timerfd module.
(Closes issue #18110. Reported, tested by tpanton. Patched by jpeeler)
* Fix problem with qualify option packets for realtime peers never stopping.
The option packets not only never stopped, but if a realtime peer was not in
the peer list multiple options dialogs could accumulate over time.
(Closes issue #16382. Reported by lftsy. Tested by zerohalo. Patched by
jpeeler)
* Fix issue where it is possible to crash Asterisk by feeding the curl engine
invalid data.
(Closes issue #18161. Reported by wdoekes. Patched by tilghman)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.1
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