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> From: Asterisk Development Team <asteriskt...@digium.com> > Date: 24 de agosto de 2010 12:35:14 BRT > To: asteriskt...@digium.com > Subject: [asterisk-dev] Asterisk 1.8.0-beta4 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-beta4. > 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 parsing of IPv6 address literals in outboundproxy > (Closes issue #17757. Reported by oej. Patched by sperreault) > > * Change the default value for alwaysauthreject in sip.conf to "yes". > (Closes issue #17756. Reported by oej) > > * Remove current STUN support from chan_sip.c. This change removes the > current > broken/useless STUN support from chan_sip. > (Closes issue #17622. Reported by philipp2. > Review: https://reviewboard.asterisk.org/r/855/) > > * PRI CCSS may use a stale dial string for the recall dial string. If an > outgoing call negotiates a different B channel than initially requested, > the > saved original dial string was not transferred to the new B channel. CCSS > uses that dial string to generate the recall dial string. > (Patched by rmudgett) > > * Split _all_ arguments before parsing them. This fixes multicast RTP paging > using linksys mode. > (Patched by russellb) > > * Expand cel_custom.conf.sample. Include the usage of CSV_QUOTE() to ensure > data has valid CSV formatting. Also list the special CEL variables that are > available for use in the mapping. There are also several other CEL fixes in > this release. > (Patched by russellb) > > > 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-beta4 > > 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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