Asterisk PBX 1.8.0 Release Candidate 2 Now Available

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!

Asterisk PBX 1.8 Release Candidate 4 Now Available

The Asterisk Development Team has announced the fourth release candidate of Asterisk  1.8 ,this release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

With all currently 1.8.0 blocker issues closed, Asterisk 1.8.0-rc4 is currently scheduled to become the full release of Asterisk 1.8.0.

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 release candidate as reported by the community. A sampling of the changes in this release candidate include:

 * Additional fixups in chan_gtalk that allow outbound calls to both Google
Talk and Google Voice recipients. Adds new chan_gtalk enhancements externip
and stunaddr.
(Closes issue #13971. Patched by dvossel)

 * Resolve manager crash issue.
(Closes issue #17994. Reported by vrban. Patchd by dvossel)

 * Documentation updates for sample configuration files.
(Closes issues #18107, #18101. Reported, patched by lathama, lmadsen)

 * Resolve issue where faxdetect would only detect the first fax call in
chan_dahdi.
(Closes issue #18116. Reported by seandarcy. Patched by rmudgett)

 * Resolve issue where a channel that is setup and torn down *very* quickly may not have the right call disposition or ${DIALSTATUS}.
(Closes issue #16946. Reported by davidw. Review
https://reviewboard.asterisk.org/r/740/)

 * Set TCLASS field of IPv6 header when SIP QoS options are set.
(Closes issue #18099. Reported by jamesnet. Patched by dvossel)

 * Resolve issue where Asterisk could crash on shutdown when using SRTP.
(Closes issue #18085. Reported by st. Patched by twilson)

 * Fix issue where peers host port would be lost on a SIP reload.
(Closes issue #18135. Reported, tested by lmadsen. Patched by dvossel)

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-rc4

Thank you for your continued support of Asterisk!