Posts Tagged LTS

Asterisk 1.8.0 Now Available!

Asterisk The Open Source PBX & Telephony Platform

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team is proud to announce the release of Asterisk 1.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

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

The release of Asterisk 1.8.0 would not have been possible without the support and contributions of the community. Since Asterisk 1.6.2, we’ve had over 500 reporters, more than 300 testers and greater than 200 developers contributed to this release.

You can find a summary of the work involved with the 1.8.0 release in the sumary:

http://svn.asterisk.org/svn/asterisk/tags/1.8.0/asterisk-1.8.0-summary.txt

A short list of available features includes:

  • Secure RTP
  • IPv6 Support in the SIP channel driver
  • 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=markup

For a full list of changes in the current release candidate, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0

Thank you for your continued support of Asterisk!

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Asterisk 1.8.0-Beta3 Now Available

Asterisk The Open Source PBX & Telephony Platform

Asterisk The Open Source PBX & Telephony Platform

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!

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Asterisk 1.8.0-Beta2 Now Available

Asterisk The Open Source PBX & Telephony Platform

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta2.
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. Some of the changes include:

  • Remove duplicate -c flag when using $(INSTALL)
    (Closes issue #17695. Reported, patched by pabelanger)
  • Don’t re-register CDR module on reload.
    (Closes issue #17304. Reported, tested by jnemeth. Patched by tilghman)
  • Don’t assume qlog is open.
    (Closes issue #17704. Reported, tested by vrban. Patched by pabelanger)
  • Expand the correct value within AST_OPTION_ONLY.
    (Closes issue #17703. Reported by stuarth. Patched by seanbright)
  • Allow for systems without locale support to be usable.
    (Closes issue #17697. Reported, patched by pprindeville. Tested by mmichelson)
  • Fixes for sounds/Makefile to install on systems using older GNU make.
    (Closes issue #17716. Reported by farisraouf. Patched by tilghman, qwell, seanbright)
  • Update logger.conf.sample to include documentation about new ‘fax’ logger level.
    (Closes issue #17715. Reported, tested by vrban. Patched by pabelanger)

Asterisk 1.8 contains many new features over previous releases of Asterisk.
A short list of included features includes:

  • Secure RTP
  • IPv6 Support
  • 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-beta2

Thank you for your continued support of Asterisk!

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