[asterisk-dev] Asterisk 1.8.0 Release Candidate 4 Now Available

[Asterisk Development Team]

The Asterisk Development Team has announced the fourth release candidate of
Asterisk 1.8.0. 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!