The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta3.
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 a regression where HTTP would always be enabled regardless of setting.
(Closes issue #17708. Reported, patched by pabelanger) - ACL errors displayed on screen when using dynamic_exclude_static in sip.conf
(Closes issue #17717. Reported by Dennis DeDonatis. Patched by mmichelson) - Support “channels” in addition to “channel” in chan_dahdi.conf.
(https://reviewboard.asterisk.org/r/804) - Fix parsing error in sip_sipredirect(). The code was written in a way that did a bad job of parsing the port out of a URI. Specifically, it would do badly when dealing with an IPv6 address.
(Closes issue #17661. Reported by oej. Patched by mmichelson) - Fix inband DTMF detection on outgoing ISDN calls.
(Patched by russellb and rmudgett) - Fixes issue with translator frame not getting freed. This issue prevented g729 licenses from being freed up.
(Closes issue #17630. Reported by manvirr. Patched by dvossel) - Fixed IPv6-related SIP parsing bugs and updated documention.
(Reported by oej. Patched by sperreault) - Add new, self-contained feature FIELDNUM(). Returns a 1-based index into a list of a specified item. Matches up with FIELDQTY() and CUT().
(Closes #17713. Reported, patched by gareth. Tested by tilghman)
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-beta3
Thank you for your continued support of Asterisk!
