Asterisk News

Asterisk Releases

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!


Asterisk 1.8.30.0 Now Available

Aug 19, 2014

The Asterisk Development Team has announced the release of Asterisk 1.8.30.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 1.8.30.0 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-24087] - [patch]chan_sip: sip_subscribe_mwi_destroy should not call sip_destroy

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-1.8.30.0

Thank you for your continued support of Asterisk!


Asterisk 13.0.0-beta1 Now Available!

Aug 11, 2014

The Asterisk Development Team is pleased to announce the first beta release of Asterisk 13.0.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases

All interested users of Asterisk are encouraged to participate in the Asterisk 13 testing process. Please report any issues found to the issue tracker, https://issues.asterisk.org/jira. All Asterisk users are invited to participate in the #asterisk-bugs channel to help communicate issues found to the Asterisk developers. It is also very useful to see successful test reports. Please post those to the asterisk-dev mailing list (http://lists.digium.com). Asterisk 13 is the next major release series of Asterisk. It will be a Long Term Support (LTS) release, similar to Asterisk 11.

For more information about support time lines for Asterisk releases, see the Asterisk versions page:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

For important information regarding upgrading to Asterisk 13, please see the Asterisk wiki:

https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+13

A short list of new features includes:

  • Asterisk security events are now provided via AMI, allowing end users to monitor their Asterisk system in real time for security related issues.
  • Both AMI and ARI now allow external systems to control the state of a mailbox. Using AMI actions or ARI resources, external systems can programmatically trigger Message Waiting Indicators (MWI) on subscribed phones. This is of particular use to those who want to build their own VoiceMail application using ARI.
  • ARI now supports the reception/transmission of out of call text messages using any supported channel driver/protocol stack through ARI. Users receive out of call text messages as JSON events over the ARI websocket connection, and can send out of call text messages using HTTP requests.
  • The PJSIP stack now supports RFC 4662 Resource Lists, allowing Asterisk to act as a Resource List Server. This includes defining lists of presence state, mailbox state, or lists of presence state/mailbox state; managing subscriptions to lists; and batched delivery of NOTIFY requests to subscribers.
  • The PJSIP stack can now be used as a means of distributing device state or mailbox state via PUBLISH requests to other Asterisk instances. This is analogous to Asterisk's clustering support using XMPP or Corosync; unlike existing clustering mechanisms, using the PJSIP stack to perform the distribution of state does not rely on another daemon or server to perform the work.

And much more!

More information about the new features can be found on the Asterisk wiki:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Documentation

A full list of all new features can also be found in the CHANGES file:

http://svnview.digium.com/svn/asterisk/branches/13/CHANGES

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

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.0-beta1

Thank you for your continued support of Asterisk!


Asterisk 12.5.0-rc1 Now Available

Aug 11, 2014

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

The release of Asterisk 12.5.0-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-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 candidate, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-12.5.0-rc1

Thank you for your continued support of Asterisk!


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