[asterisk-announce] Asterisk 1.8.0-beta5 Now Available

Asterisk Development Team asteriskteam at digium.com
Wed Sep 8 11:51:24 CDT 2010


The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta5.
This release 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,
http://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

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

  * Fix issue where TOS is no longer set on RTP packets.
    (Closes issue #17890. Reported, patched by elguero)

  * Change pedantic default value in chan_sip from 'no' to 'yes'

  * Asterisk now dynamically builds the "Supported" header depending on what is
    enabled/disabled in sip.conf. Session timers used to always be advertised as
    being supported even when they were disabled in the configuration.
    (Related to issue #17005. Patched by dvossel)

  * Convert MOH to use generic timers.
    (Closes issue #17726. Reported by lmadsen. Patched by tilghman)

  * Fix SRTP for changing SSRC and multiple a=crypto SDP lines. Adding code to
    Asterisk that changed the SSRC during bridges and masquerades broke SRTP
    functionality. Also broken was handling the situation where an incoming
    INVITE had more than one crypto offer.
    (Closes issue #17563. Reported by Alexcr. Patched by twilson)

Asterisk 1.8 contains many new features over previous releases of Asterisk.
A short list of included features includes:

     * Secure RTP
     * IPv6 Support in the SIP Channel
     * 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, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-beta5

Thank you for your continued support of Asterisk!



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