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> From: Asterisk Development Team <asteriskt...@digium.com> > Date: 8 de setembro de 2010 13:51:24 BRT > To: Asterisk Development Team <asteriskt...@digium.com> > Subject: [asterisk-dev] Asterisk 1.8.0-beta5 Now Available > Reply-To: Asterisk Developers Mailing List <asterisk-...@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! > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev
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