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Revision 1.3, Sun Sep 22 19:56:09 2019 UTC (4 years, 6 months ago) by jnemeth
Branch: MAIN
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Changes since 1.2: +1 -1
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FILE REMOVED
delete ancient Asterisk 11.*
Revision 1.2 / (download) - annotate - [select for diffs], Wed Jul 2 03:06:24 2014 UTC (9 years, 9 months ago) by jnemeth
Branch: MAIN
CVS Tags: pkgsrc-2019Q2-base,
pkgsrc-2019Q2,
pkgsrc-2019Q1-base,
pkgsrc-2019Q1,
pkgsrc-2018Q4-base,
pkgsrc-2018Q4,
pkgsrc-2018Q3-base,
pkgsrc-2018Q3,
pkgsrc-2018Q2-base,
pkgsrc-2018Q2,
pkgsrc-2018Q1-base,
pkgsrc-2018Q1,
pkgsrc-2017Q4-base,
pkgsrc-2017Q4,
pkgsrc-2017Q3-base,
pkgsrc-2017Q3,
pkgsrc-2017Q2-base,
pkgsrc-2017Q2,
pkgsrc-2017Q1-base,
pkgsrc-2017Q1,
pkgsrc-2016Q4-base,
pkgsrc-2016Q4,
pkgsrc-2016Q3-base,
pkgsrc-2016Q3,
pkgsrc-2016Q2-base,
pkgsrc-2016Q2,
pkgsrc-2016Q1-base,
pkgsrc-2016Q1,
pkgsrc-2015Q4-base,
pkgsrc-2015Q4,
pkgsrc-2015Q3-base,
pkgsrc-2015Q3,
pkgsrc-2015Q2-base,
pkgsrc-2015Q2,
pkgsrc-2015Q1-base,
pkgsrc-2015Q1,
pkgsrc-2014Q4-base,
pkgsrc-2014Q4,
pkgsrc-2014Q3-base,
pkgsrc-2014Q3
Changes since 1.1: +4 -4
lines
Diff to previous 1.1 (colored)
Update to Asterisk 11.10.2: this fixes multiple security issues along
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 Valentinaviius)
* 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!
Revision 1.1 / (download) - annotate - [select for diffs], Tue Dec 11 08:22:49 2012 UTC (11 years, 4 months ago) by jnemeth
Branch: MAIN
CVS Tags: pkgsrc-2014Q2-base,
pkgsrc-2014Q2,
pkgsrc-2014Q1-base,
pkgsrc-2014Q1,
pkgsrc-2013Q4-base,
pkgsrc-2013Q4,
pkgsrc-2013Q3-base,
pkgsrc-2013Q3,
pkgsrc-2013Q2-base,
pkgsrc-2013Q2,
pkgsrc-2013Q1-base,
pkgsrc-2013Q1,
pkgsrc-2012Q4-base,
pkgsrc-2012Q4
Update to Asterisk 11.1.0: this is a major new long term support release.
As this is a major release, you should read the information about updating:
https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11
You can also find documentation in: /usr/pkg/share/doc/asterisk
----- 11.1.0:
The Asterisk Development Team has announced the release of Asterisk 11.1.0.
The release of Asterisk 11.1.0 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 execution of 'i' extension due to uninitialized variable.
* --- Prevent resetting of NATted realtime peer address on reload.
* --- Fix ConfBridge crash if no timing module loaded.
* --- Fix the Park 'r' option when a channel parks itself.
* --- Fix an issue where outgoing calls would fail to establish audio
due to ICE negotiation failures.
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.1.0
----- 11.0.1:
The Asterisk Development Team has announced the release of Asterisk 11.0.1.
The release of Asterisk 11.0.1 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:
* --- chan_sip: Fix a bug causing SIP reloads to remove all entries
from the registry
* --- confbridge: Fix a bug which made conferences not record with
AMI/CLI commands
* --- Fix an issue with res_http_websocket where the chan_sip
WebSocket handler could not be registered.
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.0.1
Thank you for your continued support of Asterisk!
----- 11.0.0:
The Asterisk Development Team is pleased to announce the release of
Asterisk 11.0.0.
Asterisk 11 is the next major release series of Asterisk. It is a Long Term
Support (LTS) release, similar to Asterisk 1.8. For more information about
support time lines for Asterisk releases, see the Asterisk versions page:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
For important information regarding upgrading to Asterisk 11, please see the
Asterisk wiki:
https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11
A short list of new features includes:
* A new channel driver named chan_motif has been added which provides support
for Google Talk and Jingle in a single channel driver. This new channel
driver includes support for both audio and video, RFC2833 DTMF, all codecs
supported by Asterisk, hold, unhold, and ringing notification. It is also
compliant with the current Jingle specification, current Google Jingle
specification, and the original Google Talk protocol.
* Support for the WebSocket transport for chan_sip.
* SIP peers can now be configured to support negotiation of ICE candidates.
* The app_page application now no longer depends on DAHDI or app_meetme. It
has been re-architected to use app_confbridge internally.
* Hangup handlers can be attached to channels using the CHANNEL() function.
Hangup handlers will run when the channel is hung up similar to the h
extension; however, unlike an h extension, a hangup handler is associated with
the actual channel and will execute anytime that channel is hung up,
regardless of where it is in the dialplan.
* Added pre-dial handlers for the Dial and Follow-Me applications. Pre-dial
allows you to execute a dialplan subroutine on a channel before a call is
placed but after the application performing a dial action is invoked. This
means that the handlers are executed after the creation of the callee
channels, but before any actions have been taken to actually dial the callee
channels.
* Log messages can now be easily associated with a certain call by looking at
a new unique identifier, "Call Id". Call ids are attached to log messages for
just about any case where it can be determined that the message is related
to a particular call.
* Introduced Named ACLs as a new way to define Access Control Lists (ACLs) in
Asterisk. Unlike traditional ACLs defined in specific module configuration
files, Named ACLs can be shared across multiple modules.
* The Hangup Cause family of functions and dialplan applications allow for
inspection of the hangup cause codes for each channel involved in a call.
This allows a dialplan writer to determine, for each channel, who hung up and
for what reason(s).
* Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
lets you set some of the configuration options from the general section
of features.conf on a per-channel basis. FEATUREMAP() lets you customize
the key sequence used to activate built-in features, such as blindxfer,
and automon.
* Support for DTLS-SRTP in chan_sip.
* Support for named pickupgroups/callgroups, allowing any number of pickupgroups
and callgroups to be defined for several channel drivers.
* IPv6 Support for AMI, AGI, ExternalIVR, and the SIP Security Event Framework.
More information about the new features can be found on the Asterisk wiki:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Documentation
A full list of all new features can also be found in the CHANGES file.
http://svnview.digium.com/svn/asterisk/branches/11/CHANGES
For a full list of changes in the current release, please see the ChangeLog.
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.0.0
Thank you for your continued support of Asterisk!