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!
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