[asterisk-users] Asterisk 1.8.0 Release Candidate 3 Now Available

[Asterisk Development Team]

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

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

  * Still build chan_sip even if res_crypto cannot be built (use, but not depend)
    (Reported by a user on the mailing list. Patched by tilghman)

  * Get notifications for call files only when a file is closed, not when created
    (Closes issue #17924. Reported by mkeuter. Patched by abeldeck)

  * Fixes to chan_gtalk to allow outbound DTMF support to work correctly. Gtalk
    expects the DTMF to arrive on the RTP stream and not via jingle DTMF
    signalling.
    (Patched by dvossel. Tested by malcolmd)

  * Fixes to allow chan_gtalk to communicate with the Gmail web client.
    (Patched by phsultan and dvossel)

  * Fix to GET DATA to allow audio to be streamed via an AGI.
    (Closes issue #18001. Reported by jamicque. Patched by tilghman)

  * Resolve dnsmgr memory corruption in chan_iax2.
    (Closes issue #17902. Reported by afried. Patched by russell, 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=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-rc3

Thank you for your continued support of Asterisk!