----- 1.8.20.1
The Asterisk Development Team has announced the release of Asterisk 1.8.20.1.
The release of Asterisk 1.8.20.1 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-1.8.20.1
Thank you for your continued support of Asterisk!
----- 1.8.20.0
The Asterisk Development Team has announced the release of Asterisk 1.8.20.0.
The release of Asterisk 1.8.20.0 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.
* --- Improve Code Readability And Fix Setting natdetected Flag
* --- Fix extension matching with the '-' char.
* --- Fix call files when astspooldir is relative.
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.20.0
Thank you for your continued support of Asterisk!
and AST-2012-015.
Approved for commit during freeze by: agc
The Asterisk Development Team has announced security releases for Certified
Asterisk 1.8.11 and Asterisk 1.8, 10, and 11. The available security releases
are released as versions 1.8.11-cert10, 1.8.19.1, 10.11.1, 10.11.1-digiumphones,
and 11.1.1.
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.
* 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.
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, 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-11.1.1
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!
----- 1.8.19.0:
The Asterisk Development Team has announced the release of Asterisk 1.8.19.0.
The release of Asterisk 1.8.19.0 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 resetting of NATted realtime peer address on reload.
* --- Do not use a FILE handle when doing SIP TCP reads.
* --- Fix execution of 'i' extension due to uninitialized variable.
* --- Ensure that the Queue application tracks busy members in off
nominal situations
* --- Properly extract the Body information of an EWS calendar item
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.19.0
Thank you for your continued support of Asterisk!
----- 1.8.18.1:
The Asterisk Development Team has announced the release of Asterisk 1.8.18.1.
The release of Asterisk 1.8.18.1 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:
* --- chan_local: Fix local_pvt ref leak in local_devicestate().
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.18.1
Thank you for your continued support of Asterisk!
The Asterisk Development Team has announced the release of Asterisk 1.8.18.0.
The release of Asterisk 1.8.18.0 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:
* --- dsp.c User Configurable DTMF_HITS_TO_BEGIN and
DTMF_MISSES_TO_END
* --- Fix error where improper IMAP greetings would be deleted.
* --- iax2-provision: Fix improper return on failed cache retrieval
* --- Fix T.38 support when used with chan_local in between.
* --- Fix an issue where media would not flow for situations where the
legacy STUN code is in use.
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.18.0
Thank you for your continued support of Asterisk!
The Asterisk Development Team has announced the release of Asterisk 1.8.17.0.
The release of Asterisk 1.8.17.0 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 channel reference leak in ChanSpy.
* --- dsp.c: Fix multiple issues when no-interdigit delay is present,
and fast DTMF 50ms/50ms
* --- Fix bug where final queue member would not be removed from
memory.
* --- Fix memory leak when CEL is successfully written to PostgreSQL
database
* --- Fix DUNDi message routing bug when neighboring peer is
unreachable
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.17.0
Thank you for your continued support of Asterisk!
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.
AST-2012-013, and some general bugs.
The Asterisk Development Team has announced the release of Asterisk 1.8.16.0.
The release of Asterisk 1.8.16.0 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-2012-012: Resolve AMI User Unauthorized Shell Access through
ExternalIVR
* --- AST-2012-013: Resolve ACL rules being ignored during calls by
some IAX2 peers
* --- Handle extremely out of order RFC 2833 DTMF
* --- Resolve severe memory leak in CEL logging modules.
* --- Only re-create an SRTP session when needed; respond with correct
crypto policy
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.16.0
Thank you for your continued support of Asterisk!
The Asterisk Development Team has announced security releases for Certified
Asterisk 1.8.11 and Asterisk 1.8 and 10. The available security releases are
released as versions 1.8.11-cert7, 1.8.15.1, 10.7.1, and 10.7.1-digiumphones.
The release of Asterisk 1.8.11-cert7, 1.8.15.1, 10.7.1, and 10.7.1-digiumphones
resolve the following two issues:
* A permission escalation vulnerability in Asterisk Manager Interface. This
would potentially allow remote authenticated users the ability to execute
commands on the system shell with the privileges of the user running the
Asterisk application. Please note that the README-SERIOUSLY.bestpractices.txt
file delivered with Asterisk has been updated due to this and other related
vulnerabilities fixed in previous versions of Asterisk.
* When an IAX2 call is made using the credentials of a peer defined in a
dynamic Asterisk Realtime Architecture (ARA) backend, the ACL rules for that
peer are not applied to the call attempt. This allows for a remote attacker
who is aware of a peer's credentials to bypass the ACL rules set for that
peer.
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-2012-012 and AST-2012-013, 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-1.8.15.1
The security advisories are available at:
* http://downloads.asterisk.org/pub/security/AST-2012-012.pdf
* http://downloads.asterisk.org/pub/security/AST-2012-013.pdf
Thank you for your continued support of Asterisk!
The release of Asterisk 1.8.15.0 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 potential with ast_set_hangupsource() calls.
* --- Fix request routing issue when outboundproxy is used.
* --- Make the address family filter specific to the transport.
* --- Fix NULL pointer segfault in ast_sockaddr_parse()
* --- Do not perform install on existing directories
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.15.0
Thank you for your continued support of Asterisk!
- 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!
The Asterisk Development Team has announced the release of Asterisk 1.8.14.0.
The release of Asterisk 1.8.14.0 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:
* --- format_mp3: Fix a possible crash in mp3_read().
* --- Fix local channel chains optimizing themselves out of a call.
* --- Update a peer's LastMsgsSent when the peer is notified of
waiting messages
* --- Prevent sip_pvt refleak when an ast_channel outlasts its
corresponding sip_pvt.
* --- Send more accurate identification information in dialog-info SIP
NOTIFYs.
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.14.0
Thank you for your continued support of Asterisk!
AST-2012-010 and AST-2012-011
The Asterisk Development Team has announced security releases for Certified
Asterisk 1.8.11 and Asterisk 1.8 and 10. The available security releases are
released as versions 1.8.11-cert4, 1.8.13.1, 10.5.2, and 10.5.2-digiumphones.
The release of Asterisk 1.8.11-cert4, 1.8.13.1, 10.5.2, and 10.5.2-digiumphones
resolve the following two issues:
* If Asterisk sends a re-invite and an endpoint responds to the re-invite with
a provisional response but never sends a final response, then the SIP dialog
structure is never freed and the RTP ports for the call are never released. If
an attacker has the ability to place a call, they could create a denial of
service by using all available RTP ports.
* If a single voicemail account is manipulated by two parties simultaneously,
a condition can occur where memory is freed twice causing a crash.
These issues and their resolution are described in the security advisories.
For more information about the details of these vulnerabilities, please read
security advisories AST-2012-010 and AST-2012-011, 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-1.8.13.1
The security advisories are available at:
* http://downloads.asterisk.org/pub/security/AST-2012-010.pdf
* http://downloads.asterisk.org/pub/security/pST-2012-011.pdf
Thank you for your continued support of Asterisk!
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!
and AST-2012-008 along with some general bug fixes.
----- 1.8.12.1 -----
The Asterisk Development Team has announced security releases for
Certified Asterisk 1.8.11 and Asterisk 1.8 and 10. The available
security releases are released as versions 1.8.11-cert2, 1.8.12.1,
and 10.4.1.
The release of Asterisk 1.8.11-cert2, 1.8.12.1, and 10.4.1 resolve
the following two issues:
* A remotely exploitable crash vulnerability exists in the IAX2
channel driver if an established call is placed on hold without
a suggested music class. Asterisk will attempt to use an invalid
pointer to the music on hold class name, potentially causing a
crash.
* A remotely exploitable crash vulnerability was found in the Skinny
(SCCP) Channel driver. When an SCCP client closes its connection
to the server, a pointer in a structure is set to NULL. If the
client was not in the on-hook state at the time the connection
was closed, this pointer is later dereferenced. This allows remote
authenticated connections the ability to cause a crash in the
server, denying services to legitimate users.
These issues and their resolution are described in the security
advisories.
For more information about the details of these vulnerabilities,
please read security advisories AST-2012-007 and AST-2012-008,
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-1.8.12.1http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.4.1
The security advisories are available at:
* http://downloads.asterisk.org/pub/security/AST-2012-007.pdf
* http://downloads.asterisk.org/pub/security/AST-2012-008.pdf
Thank you for your continued support of Asterisk!
----- 1.8.12.2 -----
The Asterisk Development Team has announced the release of Asterisk
1.8.12.2.
The release of Asterisk 1.8.12.2 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:
* --- Resolve crash in subscribing for MWI notifications
(Closes issue ASTERISK-19827. Reported by B. R)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.12.2
Thank you for your continued support of Asterisk!
pkgsrc changes:
- set OPTIMIZE to -O3 as levels above are poorly defined and can
cause problems
- maintain current patch namimg convention
-----
The Asterisk Development Team has announced the release of Asterisk 1.8.12.0.
The release of Asterisk 1.8.12.0 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:
* --- Prevent chanspy from binding to zombie channels
* --- Fix Dial m and r options and forked calls generating warnings
for voice frames.
* --- Remove ISDN hold restriction for non-bridged calls.
* --- Fix copying of CDR(accountcode) to local channels.
* --- Ensure Asterisk acknowledges ACKs to 4xx on Replaces errors
* --- Eliminate double close of file descriptor in manager.c
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.12.0
Thank you for your continued support of Asterisk!
and AST-2012-006.
The Asterisk Development Team has announced security releases for
Asterisk 1.6.2 , 1.8, and 10. The available security releases are
released as versions 1.6.2.24, 1.8.11.1, and 10.3.1.
The release of Asterisk 1.6.2.24, 1.8.11.1, and 10.3.1 resolve the
following two issues:
* A permission escalation vulnerability in Asterisk Manager
Interface. This would potentially allow remote authenticated
users the ability to execute commands on the system shell with
the privileges of the user running the Asterisk application.
* A heap overflow vulnerability in the Skinny Channel driver.
The keypad button message event failed to check the length of
a fixed length buffer before appending a received digit to the
end of that buffer. A remote authenticated user could send
sufficient keypad button message events that th e buffer would
be overrun.
In addition, the release of Asterisk 1.8.11.1 and 10.3.1 resolve
the following issue:
* A remote crash vulnerability in the SIP channel driver when
processing UPDATE requests. If a SIP UPDATE request was received
indicating a connected line update after a channel was terminated
but before the final destruction of the associated SIP dialog,
Asterisk would attempt a connected line update on a non-existing
channel, causing a crash.
These issues and their resolution are described in the security
advisories.
For more information about the details of these vulnerabilities,
please read security advisories AST-2012-004, AST-2012-005, and
AST-2012-006, 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-1.6.2.24http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.11.1http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.3.1
The security advisories are available at:
* http://downloads.asterisk.org/pub/security/AST-2012-004.pdf
* http://downloads.asterisk.org/pub/security/AST-2012-005.pdf
* http://downloads.asterisk.org/pub/security/AST-2012-006.pdf
Thank you for your continued support of Asterisk!
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!
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!
pkgsrc changes:
- maintain patch naming convention
- detect kqueue properly
The Asterisk Development Team has announced the release of Asterisk 1.8.9.3.
The release of Asterisk 1.8.9.3 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 ACK routing for non-2xx responses.
(Closes issue ASTERISK-19389. Reported by: Karsten Wemheuer)
* --- Fix regressions with regards to route-set creation on early dialogs ---
(Closes issue ASTERISK-19358. Reported-by: Karsten Wemheuer)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.9.3
Thank you for your continued support of Asterisk!
The release of Asterisk 1.8.9.2 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolve
The release of Asterisk 1.8.9.1 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:
* --- Fixes deadlocks occuring in chan_agent ---
* --- Ensure entering T.38 passthrough does not cause an infinite loop ---
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.9.1
Thank you for your continued support of Asterisk!
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!
Asterisk Project Security Advisory - AST-2012-001
+------------------------------------------------------------------------+
| Product | Asterisk |
|----------------------+-------------------------------------------------|
| Summary | SRTP Video Remote Crash Vulnerability |
|----------------------+-------------------------------------------------|
| Nature of Advisory | Denial of Service |
|----------------------+-------------------------------------------------|
| Susceptibility | Remote unauthenticated sessions |
|----------------------+-------------------------------------------------|
| Severity | Moderate |
|----------------------+-------------------------------------------------|
| Exploits Known | No |
|----------------------+-------------------------------------------------|
| Reported On | 2012-01-15 |
|----------------------+-------------------------------------------------|
| Reported By | Catalin Sanda |
|----------------------+-------------------------------------------------|
| Posted On | 2012-01-19 |
|----------------------+-------------------------------------------------|
| Last Updated On | January 19, 2012 |
|----------------------+-------------------------------------------------|
| Advisory Contact | Joshua Colp < jcolp AT digium DOT com > |
|----------------------+-------------------------------------------------|
| CVE Name | |
+------------------------------------------------------------------------+
+------------------------------------------------------------------------+
| Description | An attacker attempting to negotiate a secure video |
| | stream can crash Asterisk if video support has not been |
| | enabled and the res_srtp Asterisk module is loaded. |
+------------------------------------------------------------------------+
+------------------------------------------------------------------------+
| Resolution | Upgrade to one of the versions of Asterisk listed in the |
| | "Corrected In" section, or apply a patch specified in the |
| | "Patches" section. |
+------------------------------------------------------------------------+
+------------------------------------------------------------------------+
| Affected Versions |
|------------------------------------------------------------------------|
| Product | Release Series | |
|-------------------------------+----------------+-----------------------|
| Asterisk Open Source | 1.8.x | All versions |
|-------------------------------+----------------+-----------------------|
| Asterisk Open Source | 10.x | All versions |
+------------------------------------------------------------------------+
+------------------------------------------------------------------------+
| Corrected In |
|------------------------------------------------------------------------|
| Product | Release |
|------------------------------------------+-----------------------------|
| Asterisk Open Source | 1.8.8.2 |
|------------------------------------------+-----------------------------|
| Asterisk Open Source | 10.0.1 |
+------------------------------------------------------------------------+
+------------------------------------------------------------------------+
| Patches |
|------------------------------------------------------------------------|
| SVN URL |Branch|
|-----------------------------------------------------------------+------|
|http://downloads.asterisk.org/pub/security/AST-2012-001-1.8.diff |v1.8 |
|-----------------------------------------------------------------+------|
|http://downloads.asterisk.org/pub/security/AST-2012-001-10.diff |v10 |
+------------------------------------------------------------------------+
+------------------------------------------------------------------------+
| Links | https://issues.asterisk.org/jira/browse/ASTERISK-19202 |
+------------------------------------------------------------------------+
+------------------------------------------------------------------------+
| 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-2012-001.pdf and |
| http://downloads.digium.com/pub/security/AST-2012-001.html |
+------------------------------------------------------------------------+
+------------------------------------------------------------------------+
| Revision History |
|------------------------------------------------------------------------|
| Date | Editor | Revisions Made |
|-----------------+--------------------+---------------------------------|
| 12-01-19 | Joshua Colp | Initial release |
+------------------------------------------------------------------------+
Asterisk Project Security Advisory - AST-2012-001
Copyright (c) 2012 Digium, Inc. All Rights Reserved.
Permission is hereby granted to distribute and publish this advisory in its
original, unaltered form.
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!
Asterisk Project Security Advisory - AST-2011-013
Product Asterisk
Summary Possible remote enumeration of SIP endpoints with
differing NAT settings
Nature of Advisory Unauthorized data disclosure
Susceptibility Remote unauthenticated sessions
Severity Minor
Exploits Known Yes
Reported On 2011-07-18
Reported By Ben Williams
Posted On
Last Updated On December 7, 2011
Advisory Contact Terry Wilson <twilson at digium.com>
CVE Name
Description 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. In 1.4 and 1.6.2, this would
mean if one setting was nat=yes or nat=route and the other
was either nat=no or nat=never. In 1.8 and 10, this would
mean when one was nat=force_rport or nat=yes and the other
was nat=no or nat=comedia.
Resolution Handling NAT for SIP over UDP requires the differing
behavior introduced by these options.
To lessen the frequency of unintended username disclosure,
the default NAT setting was changed to always respond to the
port from which we received the request-the most commonly
used option.
Warnings were added on startup to inform administrators of
the risks of having a SIP peer configured with a different
setting than that of the general setting. The documentation
now strongly suggests that peers are no longer configured
for NAT individually, but through the global setting in the
"general" context.
Affected Versions
Product Release Series
Asterisk Open Source All All versions
Corrected In
As this is more of an issue with SIP over UDP in general, there is no
fix supplied other than documentation on how to avoid the problem. The
default NAT setting has been changed to what we believe the most
commonly used setting for the respective version in Asterisk 1.4.43,
1.6.2.21, and 1.8.7.2.
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-2011-013.pdf and
http://downloads.digium.com/pub/security/AST-2011-013.html
Revision History
Date Editor Revisions Made
Asterisk Project Security Advisory - AST-2011-013
Copyright (c) 2011 Digium, Inc. All Rights Reserved.
Permission is hereby granted to distribute and publish this advisory in its
original, unaltered form.
__________________________________________________________________
Asterisk Project Security Advisory - AST-2011-014
Product Asterisk
Summary Remote crash possibility with SIP and the "automon"
feature enabled
Nature of Advisory Remote crash vulnerability in a feature that is
disabled by default
Susceptibility Remote unauthenticated sessions
Severity Moderate
Exploits Known Yes
Reported On November 2, 2011
Reported By Kristijan Vrban
Posted On 2011-11-03
Last Updated On December 7, 2011
Advisory Contact Terry Wilson <twilson at digium.com>
CVE Name
Description When the "automon" feature is enabled in features.conf, it
is possible to send a sequence of SIP requests that cause
Asterisk to dereference a NULL pointer and crash.
Resolution Applying the referenced patches that check that the pointer
is not NULL before accessing it will resolve the issue. The
"automon" feature can be disabled in features.conf as a
workaround.
Affected Versions
Product Release Series
Asterisk Open Source 1.6.2.x All versions
Asterisk Open Source 1.8.x All versions
Corrected In
Product Release
Asterisk Open Source 1.6.2.21, 1.8.7.2
Patches
Download URL Revision
http://downloads.asterisk.org/pub/security/AST-2011-014-1.6.2.diff 1.6.2.20
http://downloads.asterisk.org/pub/security/AST-2011-014-1.8.diff 1.8.7.1
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-2011-014.pdf and
http://downloads.digium.com/pub/security/AST-2011-014.html
Revision History
Date Editor Revisions Made
Asterisk Project Security Advisory - AST-2011-014
Copyright (c) 2011 Digium, Inc. All Rights Reserved.
Permission is hereby granted to distribute and publish this advisory in its
original, unaltered form.
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.
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.
