Posts Tagged asterisk

Asterisk 1.4.26-rc5 released

Posted by admin in asterisk on July 2, 2009

Asterisk The Open Source PBX & Telephony Platform

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team has announced the fifth release candidate of Asterisk 1.4.26. Asterisk 1.4.26-rc5 is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

This release resolves the following issues, plus some other minor issues:

* Improve the mapping of extension states from combined device states (issue #15413).

* Added an error message to make it clear why a SIP peer was not processed when a DNS lookup fails on a host or outbound proxy (issue #13432).

* Fix a problem where chan_sip would ignore “old” but valid responses (issue #11231).

* Resolve a crash in chan_iax2 (issue #15377).

* Resolve a crash related to a T.38 reinvite race condition.

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

http://svn.asterisk.org/svn/asterisk/tags/1.4.26-rc5/ChangeLog

Issues found in this release candidate can be reported at https://issues.asterisk.org

Thank you for your continued support of Asterisk!

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Asterisk 1.6.0.11-rc1 released

Posted by admin in asterisk on June 25, 2009

Asterisk The Open Source PBX & Telephony Platform

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team is pleased to announce the first release candidate of Asterisk 1.6.0.11. Asterisk 1.6.0.11-rc1 is available for immediate download at http://downloads.digium.com/pub/asterisk/

This release candidate fixes several issues reported by the community since the last 1.6.0 release.  Some of these issues include:

* Fix a possible infinite loop in SDP parsing during a SIP reinvite glare (issues #15213, #14464, and #15345).

* Fix memory corruption and leakage related to reloads of non-files mode music on hold (issues #15109, #15123, and #15195).

* Safely handle AMI connections/reload requests that occur during startup (issues #15189, and #13778)

* Add flags to chanspy audiohook so that audio stays in sync (issue #13745)

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

http://svn.asterisk.org/svn/asterisk/tags/1.6.0.11-rc1/ChangeLog

Issues found in this release candidate can be reported at http://issues.asterisk.org

Thank you for your continued support of Asterisk!

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Asterisk 1.4.26-rc4 released

Posted by admin in asterisk on June 25, 2009

Asterisk The Open Source PBX & Telephony Platform

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team has announced the fourth release candidate of Asterisk 1.4.26. Asterisk 1.4.26-rc4 is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

Asterisk 1.4.26-rc4 has primarily been released to resolve a possible infinite loop in SDP parsing during a SIP reinvite glare situation. Additionally, this release also resolves several minor issues.

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

http://svn.asterisk.org/svn/asterisk/tags/1.4.26-rc4/ChangeLog

Issues found in this release candidate can be reported at http://issues.asterisk.org/.

Thank you for your continued support of Asterisk!

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Asterisk 1.4.26-rc3 released

Posted by admin in asterisk on June 18, 2009

Asterisk The Open Source PBX & Telephony Platform

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team is pleased to announce the third release candidate of Asterisk 1.4.26. Asterisk 1.4.26-rc3 is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

This is an incremental release to resolve three moderately important issues.

  • Race condition with StopMixMonitor() (issue #15259)
  • Fix memory corruption and leakage related to reloads of non files mode MoH classes (issues #15109, #15123, #15195)
  • Proactively avoid a potential crash in app_queue by changing the datastore traversal in ast_do_masquerade to use a safe list traversal

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

http://svn.asterisk.org/svn/asterisk/tags/1.4.26-rc3/ChangeLog

Issues found in this release candidate can be reported at http://issues.asterisk.org

Thank you for your continued support of Asterisk!

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Asterisk 1.6.2.0-beta3 released

Posted by admin in asterisk on June 15, 2009

Asterisk The Open Source PBX & Telephony Platform

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team is pleased to announce the third beta of Asterisk 1.6.2.0. Asterisk-1.6.2.0-beta3 is available for immediate download at http://downloads.digium.com/pub/asterisk/

This is an incremental release of the 1.6.2.0 branch as the previous beta was released just over a month ago, and many issues have been resolved since then.
Included in this release are the following issues reported by the community:

  • Update spiral support in trunk and 1.6.x branches to match what is in 1.4 (related to issue #13630).
  • Fix RFC2833 issues with DTMF getting duplicated and with duration wrapping over (issue #14815).
  • Fix a bug where the codecs of the called party leg were not properly sent back to the call leg when reinvited (issue #13569).
  • Fix broken attended transfers (issue #15183).
  • Add flags to chanspy audiohook so that audio stays in sync (issue #13745).
  • Resolve issues with choppy sound when using res_timing_pthread (issue #14412)

Additionally, an update to chan_iax2 related to issue AST-2009-001 is included in this beta release. For more information, see:

http://downloads.asterisk.org/pub/security/AST-2009-001.html

For a full list of changes in this beta, please see the ChangeLog:

http://svn.digium.com/svn/asterisk/tags/1.6.2.0-beta3/ChangeLog

You can get more information about the new features and various changes in Asterisk 1.6.2.0 at:

http://svn.digium.com/svn/asterisk/tags/1.6.2.0-beta3/CHANGES

And if you’re upgrading from previous versions of Asterisk see this file:

http://svn.digium.com/svn/asterisk/tags/1.6.2.0-beta3/UPGRADE.txt

Issues discovered in testing of this beta can be reported at http://issues.asterisk.org

Thank you for your continued support of Asterisk!

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Asterisk 1.4.26-rc2 released

Posted by admin in asterisk on June 11, 2009

Asterisk The Open Source PBX & Telephony Platform

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team is pleased to announce the second release candidate of Asterisk 1.4.26. Asterisk 1.4.26-rc2 is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

This release fixes some issues reported by the community since the first release candidate. The most significant issues resolved include:

  • Treat an empty FORWARD_CONTEXT channel variable the same way we treat a non-existant one (issue #15056).
  • If using a deprecated musiconhold.conf format, reloading the module would cause the class to disappear (issue #14759).
  • Creation of generic call forward API, ast_call_forward() which is used to resolve a potentional SIP spiral detection problem (issue #13630).
  • Resolve a regression introduced after changes to module load order that were necessary to close another issue. The regression caused issues with the usage of #exec, affecting FreePBX users (issue #15189).

Additionally, updates that went into Asterisk 1.4.25.1 to resolve a chan_iax2 issue have been merged into this release candidate (AST-2009-001).

The original security advisory can be found here:

http://downloads.asterisk.org/pub/security/AST-2009-001.html

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

http://svn.asterisk.org/svn/asterisk/tags/1.4.26-rc2/ChangeLog

Issues found in this release candidate can be reported at http://issues.asterisk.org

Thank you for your continued support of Asterisk!

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Asterisk 1.2.33, Asterisk 1.4.25.1, Asterisk 1.6.0.10, Asterisk 1.6.1.1 released

Posted by admin in asterisk on June 5, 2009

Asterisk The Open Source PBX & Telephony Platform

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team has released Asterisk versions 1.2.33, 1.4.25.1, 1.6.0.10, and 1.6.1.1. The released versions are available at http://downloads.asterisk.org/pub/telephony/asterisk/

This release fixes a REGAUTH loop related to security issue AST-2009-001.
Asterisk release 1.2.33 also addresses a small compile time error in chan_spy.

For more information about the security issue, please see:

http://downloads.asterisk.org/pub/security/AST-2009-001.html

For a summary of the changes in this release, please see the release summary:

http://svn.asterisk.org/svn/asterisk/tags/1.2.33/asterisk-1.2.33-summary.txt
http://svn.asterisk.org/svn/asterisk/tags/1.4.25.1/asterisk-1.4.25.1-summary.txt
http://svn.asterisk.org/svn/asterisk/tags/1.6.0.10/asterisk-1.6.0.10-summary.txt
http://svn.asterisk.org/svn/asterisk/tags/1.6.1.1/asterisk-1.6.1.1-summary.txt

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

http://svn.asterisk.org/svn/asterisk/tags/1.2.33/ChangeLog
http://svn.asterisk.org/svn/asterisk/tags/1.4.25.1/ChangeLog
http://svn.asterisk.org/svn/asterisk/tags/1.6.0.10/ChangeLog
http://svn.asterisk.org/svn/asterisk/tags/1.6.1.1/ChangeLog

Thank you for your continued support of Asterisk!

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Asterisk 1.4.26-rc1 released

Posted by admin in asterisk on June 1, 2009

Asterisk The Open Source PBX & Telephony Platform

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team is pleased to announce the first release candidate of Asterisk 1.4.26. Asterisk 1.4.26-rc1 is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

This release is primarily a fix for an issue (#14867, #14717) related to security fix AST-2009-001 where IAX was not sending REGREJ to terminate invalid registrations. Instead it sent another REGAUTH if the authentication challenge
failed. This caused a loop of REGREQ and REGAUTH frames. Additionally, an issue (#15183) with broken attended transfers where the bridge was terminating immediately after the transfer is resolved in this release, along with some
other issues reported by the community.

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

http://svn.asterisk.org/svn/asterisk/tags/1.4.26-rc1/ChangeLog

The original security advisory can be found here:

http://downloads.asterisk.org/pub/security/AST-2009-001.html

Issues found in this release candidate can be reported at
http://issues.asterisk.org

Thank you for your continued support of Asterisk!

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Asterisk 1.4.25 released

Posted by admin in asterisk on May 21, 2009

Asterisk The Open Source PBX & Telephony Platform

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team is pleased to announce the release of Asterisk 1.4.25. Asterisk 1.4.25 is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

This release resolves several crash issues, DTMF related issues, and CDR related issues.

For a summary of the changes in this release, please see the release summary:

http://svn.asterisk.org/svn/asterisk/tags/1.4.25/asterisk-1.4.25-summary.txt

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

http://svn.asterisk.org/svn/asterisk/tags/1.4.25/ChangeLog

The following list of issues were resolved with the participation of the community, and this release would not have been possible without your help!

* Allow H.323 Plus library to be used in addition to the OpenH323 library.
- Closes issue #11261. Reported by vhatz. Patched by jthurman.

* Delay signalling progress until a PRI channel really signals progress.
- Closes issue #13034. Reported, patched, and tested by klaus3000.

* Avoid destroying the CLI line when moving the cursor backward and trying to autocomplete.
- Closes issue #14373. Reported, and patched by eliel.

* Fix a crash due to too few arguments to RetryDial.
- Closes issue #14852. Reported, and patched by junky.

* Fix RFC2833 issues with DTMF getting duplicated and with duration wrapping over.
- Closes issue #14815 and #14460. Reported by geoff2010 and moliveras. Patched by dimas. Tested by geoff2010, file, dimas, ZX81, moliveras.

Thank you for your continued support of Asterisk!

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Asterisk open source project servers have new names!

Posted by admin in asterisk on May 15, 2009

In order to more closely align the services that Digium provides to the Asterisk open source community with the Asterisk project itself, we’ve recently renamed many of the servers that provide these services.

Effective immediately:

bugs.digium.com has moved to issues.asterisk.org

There are no content or functional changes (except for the new site being SSL/TLS enabled), only a renaming of the site. The old URLs will continue to operate indefinitely, automatically redirecting the user to the new site.

reviewboard.digium.com has moved to reviewboard.asterisk.org

There are no content or functional changes (except for the new site being SSL/TLS enabled), only a renaming of the site. The old URLs will continue to operate indefinitely, automatically redirecting the user to the new site.

svn.digium.com has moved to svn.asterisk.org

There are no content or functional changes, and the old URLs will continue to operate indefinitely, *without* redirects, as Subversion does not handle redirects in a transparent fashion and we don’t want to break users’ existing checkouts.

downloads.digium.com has partially moved to downloads.asterisk.org

The open source Asterisk project content has moved to the new site, which contains *only* open source content. The Digium commercial products present on downloads.digium.com will continue to be hosted there. URLs to open source content that used to be present on downloads.digium.com will automatically redirect to downloads.asterisk.org.

Hopefully these changes have been made in as transparent a fashion as possible, and you won’t experience any problems. If you do, please don’t hesitate to post on the asterisk-users mailing list and we’ll try to get the problem addressed as quickly as possible.

Thanks for using Asterisk!

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