[AsteriskBrasil] Fwd: [asterisk-dev] Asterisk 1.8.0-beta5 Now Available

[Denis Galvão - Gmail]

Begin forwarded message:

> From: Asterisk Development Team <asteriskteam em digium.com>
> Date: 8 de setembro de 2010 13:51:24 BRT
> To: Asterisk Development Team <asteriskteam em digium.com>
> Subject: [asterisk-dev] Asterisk 1.8.0-beta5 Now Available
> Reply-To: Asterisk Developers Mailing List <asterisk-dev em lists.digium.com>
> 
> 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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