The Asterisk Development Team has announced the release of Asterisk 1.6.2.15.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.6.2.15 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:

* When using chan_skinny, don’t crash when parking a non-bridged call.
(Closes issue #17680. Reported, tested by jmhunter. Patched, tested by DEA)

* Add ability for Asterisk to try both the encoded and unencoded subscription
URI for a match in hints.
(Closes issue #17785. Reported, tested by ramonpeek. Patched by tilghman)

* Set the caller id on CDRs when it is set on the parent channel.
(Closes issue #17569. Reported, patched by tbelder)

* Ensure user portion of SIP URI matches dialplan when using encoded characters
(Closes issue #17892. Reported by wdoekes. Patched by jpeeler)

* Resolve issue where Party A in an analog 3-way call would continue to hear
ringback after party C answers.
(Patched by rmudgett)

* 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)

* Multiple fixes related to Local channels.

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.15

Thank you for your continued support of Asterisk!

The Asterisk Development Team has announced the release of Asterisk 1.4.38. This
release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.4.38 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:

* Add ability for Asterisk to try both the encoded and unencoded subscription
URI for a match in hints.
(Closes issue #17785. Reported, tested by ramonpeek. Patched by tilghman)

* Set the caller id on CDRs when it is set on the parent channel.
(Closes issue #17569. Reported, patched by tbelder)

* Ensure user portion of SIP URI matches dialplan when using encoded characters
(Closes issue #17892. Reported by wdoekes. Patched by jpeeler)

* Fix a crash in res_jabber by ensuring that we don’t alter memory after it’s
freed.
(Closes issue #17387. Reported, tested by jmls. Patched by tilghman)

* 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)

* Multiple fixes related to Local channels.

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.38

Thank you for your continued support of Asterisk!

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

The release of Asterisk 1.6.2.14 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 session timers would be advertised as supported even
when session-timers=refuse was set in sip.conf. Also fix
interoperability problems with session timer behavior in Asterisk.
(Closes issue #17005. Reported by alexcarey. Patched by dvossel)

* Parse all “Accept” headers for SIP SUBSCRIBE requests.
(Closes issue #17758. Reported by ibc. Patched by dvossel)

* Fix issue where queue stats would be reset on reload.
(Closes issue #17535. Reported by raarts. Patched by tilghman)

* Fix issue where MoH files were no longer rescanned on during a
reload.
(Closes issue #16744. Reported by pj. Patched by Qwell)

* Fix issue with dialplan pattern matching where the specificity for
pattern ranges and pattern characters was inconsistent.
(Closes issue #16903. Reported, patched by Nick_Lewis)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.14

Thank you for your continued support of Asterisk!

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

The release of Asterisk 1.4.37 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 with decoding ^-escaped characters in realtime (res_pgsql)
(Closes issue #17790. Reported denzs. Patched by Qwell)

* Don’t send a devstate change on poke_noanswer if the state did not
change.
(Closes issue #17741. Reported, patched by schmidts)

* Transmit silence when reading DTMF in ast_readstring. Otherwise you
could get issues with DTMF timeouts causing hangups.
(Closes issue #17370. Reported, patched by makoto)

* Fix to SIP extension state update (deadlock issues)
(Closes issue #17888. Reported by zerohalo. Patched by dvossel)

* Fix issue with MoH where it doesn’t recover cleanly when it can’t
play a file and would just stop, instead of continuing to find the
next playable file in the MoH class.
(Closes issue #17807. Reported by kshumard. Patched by bbryant)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.37

Thank you for your continued support of Asterisk!

The Asterisk Development Team has announced the second release candidate of
Asterisk 1.8.0. This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

Asterisk 1.8.0-rc1 was not released due to an issue found prior to release.

* Make AMI honor enabled=no
(Closes issue #18040. Reported by: twilson
Review: https://reviewboard.asterisk.org/r/938/)

All interested users of Asterisk are encouraged to participate in the 1.8
testing process. Please report any issues found to the issue tracker,
https://issues.asterisk.org/. It is also very useful to see successful test
reports. Please post those to the asterisk-dev mailing list.

Asterisk 1.8 is the next major release series of Asterisk. It will be a Long
Term Support (LTS) release, similar to Asterisk 1.4. For more information about
support time lines for Asterisk releases, see the Asterisk versions page.

http://www.asterisk.org/asterisk-versions

With the availability of the Asterisk 1.8.0 release candidates, the binary
add-on modules for Asterisk produced by Digium have been updated with new
versions that are compatible with Asterisk 1.8. The availability of these
modules will assist with the testing of Asterisk 1.8.0 in a wider variety of
situations.

This release candidate contains fixes since the last beta release as reported by
the community. A sampling of the changes in this release candidate include:

* Add slin16 support for format_wav (new wav16 file extension)
(Closes issue #15029. Reported, patched by andrew. Tested by Qwell)

* Fixes a bug in manager.c where the default configuration values weren’t reset
when the manager configuration was reloaded.
(Closes issue #17917. Reported by lmadsen. Patched by bbryant)

* Various fixes for the calendar modules.
(Patched by Jan Kalab.
Reviewboard: https://reviewboard.asterisk.org/r/880/
Closes issue #17877. Review: https://reviewboard.asterisk.org/r/916/
Closes issue #17776. Review: https://reviewboard.asterisk.org/r/921/)

* Add CHANNEL(checkhangup) to check whether a channel is in the process of
being hung up.
(Closes issue #17652. Reported, patched by kobaz)

* Fix a bug with MeetMe where after announcing the amount of time left in a
conference, if music on hold was playing, it doesn’t restart.
(Closes issue #17408, Reported, patched by sysreq)

* Fix interoperability problems with session timer behavior in Asterisk.
(Closes issue #17005. Reported by alexcarey. Patched by dvossel)

* Rate limit calls to fsync() to 1 per second after astdb updates. Astdb was
determined to be one of the most significant bottlenecks in SIP registration
processing. This patch improved the speed of an astdb load test by 50000%
(yes, Fifty-Thousand Percent). On this particular load test setup, this
doubled the number of SIP registrations the server could handle.
(Review: https://reviewboard.asterisk.org/r/825/)

* Don’t clear the username from a realtime database when a registration
expires. Non-realtime chan_sip does not clear the username from memory when a
registration expiries so realtime probably shouldn’t either.
(Closes issue #17551. Reported, patched by: ricardolandim. Patched by
mnicholson)

* Don’t hang up a call on an SRTP unprotect failure. Also make it more obvious
when there is an issue en/decrypting.
(Closes issue #17563. Reported by Alexcr. Patched by sfritsch. Tested by
twilson)

* Many more issues. This is a significant upgrade over Asterisk 1.8.0 beta 5!

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=checkout

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-rc2

Thank you for your continued support of Asterisk!