The Asterisk Development Team has announced the release of Asterisk 1.6.2.22. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.6.2.22 corrects two flaws in sip.conf.sample related to AST-2011-013:

* The sample file listed *two* values for the ‘nat’ option as being the default. Only ‘yes’ is the default.

* The warning about having differing ‘nat’ settings confusingly referred to both peers and users.

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.22

Thank you for your continued support of Asterisk!

The AstLinux Team is happy to announce the release of AstLinux 1.0.0. This release includes significant changes and improvements over past releases. Specific upgrade and new installation instructions are available at: http://www.astlinux.org

Some of the highlights include:

* Using eglibc instead of uClibc. This allows binary compatibility with add-ons that are provided as binary only (G.729 CODEC, Fax for Asterisk etc).
* Newer Kernel which better supports newer hardware
* Support for Jabber/Gtalk
* Removed mISDN support (the zaphfc DAHDI driver is included for single port ISDN cards)

A full changelog is available on the release pages. We provide versions with Asterisk 1.8 and 1.4.

Because this is a major version change, there are some special considerations when upgrading. Please read the instructions very carefully to ensure no step is skipped.

http://doc.astlinux.org/userdoc:upgrade-0.7

Please report any issues with the release back to the AstLinux mailing list.

Enjoy,

The AstLinux Team

The Asterisk Development Team is proud to announce the release of Asterisk 10.0.0. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

Asterisk 10 is the next major release series of Asterisk. It will be a Standard support release, similar to Asterisk 1.6.2. For more
information about support time lines for Asterisk releases, see the Asterisk versions page:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

With the release of the Asterisk 10 branch, the preceding ’1.’ has been removed from the version number per the blog post available at

http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/

The release of Asterisk 10 would not have been possible without the support and contributions of the community.

You can find an overview of the work involved with the 10.0.0 release in the summary:

http://svn.asterisk.org/svn/asterisk/tags/10.0.0/asterisk-10.0.0-summary.txt

A short list of available features includes:

* T.38 gateway functionality has been added to res_fax.
* Protocol independent out-of-call messaging support. Text messages not associated with an active call can now be routed through the Asterisk
dialplan. SIP and XMPP are supported so far.
* New highly optimized and customizable ConfBridge application capable of mixing audio at sample rates ranging from 8kHz-192kHz
* Addition of video_mode option in confbridge.conf to provide basic video conferencing in the ConfBridge() dialplan application.
* Support for defining hints has been added to pbx_lua.
* Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB).
* Much, much more!

A full list of new features can be found in the CHANGES file.

http://svn.asterisk.org/svn/asterisk/branches/10/CHANGES

Also, when upgrading a system between major versions, it is imperative that you read and understand the contents of the UPGRADE.txt file,  which is located at:

http://svn.asterisk.org/svn/asterisk/branches/10/UPGRADE.txt

Thank you for your continued support of Asterisk!

The Asterisk Development Team is pleased to announce the release of Asterisk 1.8.8.0. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

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.
(Closes ASTERISK-17288. Reported by: Mikael Carlsson Tested by: Matthew
Jordan Review: https://reviewboard.asterisk.org/r/1416/)

* 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.
(Closes issue ASTERISK-18090. Patched by Kinsey Moore)

* Fix bad RTP media bridges in directmedia calls on peers separated by multiple Asterisk nodes.
(Closes issue ASTERISK-18340. Reported by: Thomas Arimont. Closes issue
ASTERISK-17725. Reported by: kwk. Tested by: twilson, jrose)

* Fix crashes in ast_rtcp_write()
(Closes issue ASTERISK-18570)
Related issues that look like they are the same problem:
(Issue ASTERISK-17560, ASTERISK-15406, ASTERISK-15257, ASTERISK-13334, ASTERISK-9977, ASTERISK-9716)
Review: https://reviewboard.asterisk.org/r/1444/
Patched by: Russell Bryant

* 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.
(Closes ASTERISK-16981. Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1443)

* Prevent segfault if call arrives before Asterisk is fully booted.
(Patched by alecdavis. https://reviewboard.asterisk.org/r/1407/)

* 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
(Closes issue ASTERISK-18101. Reported by Paul Rolfe, patched by Gregory Nietsky)
(Closes issue ASTERISK-18487. Reported by Jason Legault, patched by Gregory Nietsky)

* Fix regression in configure script for libpri capability checks
(Closes issue ASTERISK-18687. Reported by norbert, patched by Richard Mudgett)

* Prevent BLF subscriptions from causing deadlocks.
(Closes issue ASTERISK-18663)
Review: https://reviewboard.asterisk.org/r/1563/

* Fix deadlock if peer is destroyed while sending MWI notice.
(Closes issue ASTERISK-18747)
Reported by: Gregory Hinton Nietsky

* Fix issue with setting defaultenabled on categories that are already enabled by default.
(Closes issue ASTERISK-18738)
Reported by: Paul Belanger

* 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!

The Asterisk Development Team has announced security releases for Asterisk 1.4, 1.6.2 and 1.8. The available security releases are released as versions 1.4.43, 1.6.2.21 and 1.8.7.2.

These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases

The release of Asterisk versions 1.4.43, 1.6.2.21, and 1.8.7.2 resolves an issue with possible remote enumeration of SIP endpoints with differing NAT settings.

The release of Asterisk versions 1.6.2.21 and 1.8.7.2 resolves a remote crash possibility with SIP when the “automon” feature is enabled.

The issues and resolutions are described in the AST-2011-013 and AST-2011-014 security advisories.

For more information about the details of these vulnerabilities, please read the security advisories AST-2011-013 and AST-2011-014, 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-1.4.43
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.21
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.7.2

Security advisory AST-2011-013 is available at:

* http://downloads.asterisk.org/pub/security/AST-2011-013.pdf

Security advisory AST-2011-014 is available at:

* http://downloads.asterisk.org/pub/security/AST-2011-013.pdf

Thank you for your continued support of Asterisk!