Archive for category asterisk

Asterisk 1.4.26-rc6 released

Posted by admin in asterisk on July 14, 2009

Asterisk The Open Source PBX & Telephony Platform

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team has announced the sixth release candidate of Asterisk 1.4.26. Asterisk 1.4.26-rc6 is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

This release resolves the following issues, in addition to some 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-rc6/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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Fax For Asterisk

Posted by admin in asterisk, digium, g.711, sip, switchvox, t.38 on July 13, 2009

T.38 fax for Asterisk

T.38 fax for Asterisk

Digium‘s Fax For Asterisk is a commercial facsimile (Fax) termination and origination solution designed to enhance the capabilities of Open Source and commercial Asterisk as well as Switchvox. Fax For Asterisk bundles a suite of user-friendly Asterisk applications and a licensed version of the industry’s leading fax modem software from Commetrex. Fax For Asterisk provides low speed (14400bps) PSTN faxing via DAHDI-compatible telephony boards as well as VoIP faxing to T.38-compatible SIP endpoints and service providers. Licensed on a per-channel basis, Digium‘s Fax For Asterisk provides a complete, cost-effective, commercial fax solution for Asterisk users.

Fax For Asterisk provides two components: res_fax and res_fax_digium. Res_fax is an Asterisk resource module that adds fax termination and origination functionality in Asterisk. It provides Asterisk dialplan functions and dialplan applications to enable the user to build highly-customizable fax solutions. Res_fax_digium provides core fax processing functionality in the form of several supported fax modems — V.21, V.27ter, V.29, and V.17 — to achieve speeds up to 14400bps.

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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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