Asterisk News

Asterisk Releases

Asterisk 11.13.0-rc1 Now Available

Sep 19, 2014

The Asterisk Development Team has announced the first release candidate of Asterisk 11.13.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 11.13.0-rc1 resolves several issues reported by the community and would have not been possible without your participation.

Thank you!

The following are the issues resolved in this release candidate:

Bug

  • [ASTERISK-22252] - res_musiconhold cleanup - REF_DEBUG reload warnings and ref leaks
  • [ASTERISK-23577] - res_rtp_asterisk: Crash in ast_rtp_on_turn_rtp_state when RTP instance is NULL
  • [ASTERISK-23634] - With TURN Asterisk crashes on multiple (7-10) concurrent WebRTC (avpg/encryption/icesupport) calls
  • [ASTERISK-23767] - [patch] Dynamic IAX2 registration stops trying if ever not able to resolve
  • [ASTERISK-23997] - chan_sip: port incorrectly incremented for RTCP ICE candidates in SDP answer
  • [ASTERISK-24019] - When a Music On Hold stream starts it restarts at beginning of file.
  • [ASTERISK-24032] - Gentoo compilation emits warning: "_FORTIFY_SOURCE" redefined
  • [ASTERISK-24178] - [patch]fromdomainport used even if not set
  • [ASTERISK-24211] - testsuite: Fix the dial_LS_options test
  • [ASTERISK-24225] - Dial option z is broken
  • [ASTERISK-24249] - SIP debugs do not stop
  • [ASTERISK-24301] - Security: Out of call MESSAGE requests processed via Message channel driver can crash Asterisk

Improvement

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.13.0-rc1

Thank you for your continued support of Asterisk!


Asterisk 1.8.31.0-rc1 Now Available

Sep 19, 2014

The Asterisk Development Team has announced the first release candidate of Asterisk 1.8.31.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 1.8.31.0-rc1 resolves several issues reported by the community and would have not been possible without your participation.

Thank you!

The following are the issues resolved in this release candidate:

Bug

Improvement

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.31.0-rc1

Thank you for your continued support of Asterisk!


Asterisk 11.6-cert6, 11.12.1, 12.5.1 Now Available (Security Release)

Sep 18, 2014

The Asterisk Development Team has announced security releases for Certified Asterisk 11.6 and Asterisk 11 and 12. The available security releases are released as versions 11.6-cert6, 11.12.1, and 12.5.1. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases

Please note that the release of these versions resolves the following security vulnerability:

  • AST-2014-010: Remote Crash when Handling Out of Call Message in Certain Dialplan Configurations

Additionally, the release of Asterisk 12.5.1 resolves the following security vulnerability:

  • AST-2014-009: Remote Crash Based on Malformed SIP Subscription Requests

Note that the crash described in AST-2014-010 can be worked around through dialplan configuration. Given the likelihood of the issue, an advisory was deemed to be warranted.

For more information about the details of these vulnerabilities, please read security advisories AST-2014-009 and AST-2014-010, which were released at the same time as this announcement.

For a full list of changes in the current releases, please see the ChangeLogs:

The security advisories are available at:

Thank you for your continued support of Asterisk!


Asterisk 12.5.0 Now Available

Aug 19, 2014

The Asterisk Development Team has announced the release of Asterisk 12.5.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 12.5.0 resolves several issues reported by the community and would have not been possible without your participation.

Thank you!

The following are the issues resolved in this release:

Bug

  • [ASTERISK-18345] - [patch] sips connection dropped by asterisk with a large INVITE
  • [ASTERISK-23508] - Memory Corruption in __ast_string_field_ptr_build_va
  • [ASTERISK-23814] - No call started after peer dialed
  • [ASTERISK-23818] - PBX_Lua: after asterisk startup module is loaded, but dialplan not available
  • [ASTERISK-23825] - Alembic scripts - table queue_members missing unique index on column uniqueid
  • [ASTERISK-23847] - Alembic voicemail script - 'recording' column should be longblob on MySQL
  • [ASTERISK-23852] - ARI mixing bridges should propagate linkedids.
  • [ASTERISK-23909] - Alembic scripts - table sippeers could use a longer useragent column
  • [ASTERISK-23911] - URIENCODE/URIDECODE: WARNING about passing an empty string is a bit over zealous
  • [ASTERISK-23941] - ARI: Attended transfers of channels into Stasis application lose information
  • [ASTERISK-23969] - SendMessage AMI action Cant Send Text Message Over PJSIP
  • [ASTERISK-23985] - PresenceState Action response does not contain ActionID; duplicates Message Header
  • [ASTERISK-23987] - BridgeWait: channel entering into holding bridge that is being destroyed fails to successfully join the newly created holding bridge
  • [ASTERISK-24087] - [patch]chan_sip: sip_subscribe_mwi_destroy should not call sip_destroy

Improvement

  • [ASTERISK-21178] - Improve documentation for manager command Getvar, Setvar
  • [ASTERISK-23692] - ARI: Add a Messaging Capability
  • [ASTERISK-24036] - ARI: Recording resource should allow copying a recording
  • [ASTERISK-24037] - ARI: RecordingFinished event should return duration of recording

New Feature

  • [ASTERISK-24000] - chan_pjsip: Add accountcode setting
  • [ASTERISK-24119] - HEP: Add module that exports RTCP information to a Homer Capture Server

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-12.5.0

Thank you for your continued support of Asterisk!


Asterisk 11.12.0 Now Available

Aug 19, 2014

The Asterisk Development Team has announced the release of Asterisk 11.12.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 11.12.0 resolves several issues reported by the community and would have not been possible without your participation.

Thank you!

The following are the issues resolved in this release:

Bug

  • [ASTERISK-18345] - [patch] sips connection dropped by asterisk with a large INVITE
  • [ASTERISK-23508] - Memory Corruption in __ast_string_field_ptr_build_va
  • [ASTERISK-23814] - No call started after peer dialed
  • [ASTERISK-23818] - PBX_Lua: after asterisk startup module is loaded, but dialplan not available
  • [ASTERISK-23911] - URIENCODE/URIDECODE: WARNING about passing an empty string is a bit over zealous
  • [ASTERISK-23985] - PresenceState Action response does not contain ActionID; duplicates Message Header
  • [ASTERISK-24087] - [patch]chan_sip: sip_subscribe_mwi_destroy should not call sip_destroy
  • [ASTERISK-24194] - Loading AST_MODFLAG_DEFAULT in pbx_lua.c causes undefined symbol error

Improvement

  • [ASTERISK-21178] - Improve documentation for manager command Getvar, Setvar

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.12.0

Thank you for your continued support of Asterisk!


Pages

Subscribe to