Asterisk News

Asterisk Releases

Asterisk 13.0.0-beta2 Now Available!

Sep 19, 2014

The Asterisk Development Team is pleased to announce the second 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-beta2

Thank you for your continued support of Asterisk!


Asterisk 12.6.0-rc1 Now Available

Sep 19, 2014

The Asterisk Development Team has announced the first release candidate of Asterisk 12.6.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 12.6.0-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-23994] - res_pjsip_sdp_rtp: owner address in SDP may not be fully qualified domainname
  • [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-24027] - MixMonitor AMI action called during AGI execution from bridge feature causes channel to leave AGI has hung up
  • [ASTERISK-24032] - Gentoo compilation emits warning: "_FORTIFY_SOURCE" redefined
  • [ASTERISK-24043] - ARI /continue fails to actually continue into the dialplan
  • [ASTERISK-24136] - Security: Crash in Asterisk's PJSIP code when subscribing to an event with an unexpected body type
  • [ASTERISK-24143] - pjsip: Outbound call to WebRTC UA fails to transmit ACK on received 200 OK
  • [ASTERISK-24147] - ARI: channel hangup crashes asterisk process
  • [ASTERISK-24161] - PJSIPShowEndpoint gives inaccurate count of list items
  • [ASTERISK-24178] - [patch]fromdomainport used even if not set
  • [ASTERISK-24212] - testsuite: Sporadic crash due to assert on stopping RTP engine
  • [ASTERISK-24225] - Dial option z is broken
  • [ASTERISK-24229] - ARI: playback of sounds implicitly answers channel, preventing early media playback
  • [ASTERISK-24231] - crash: CLI execution of realtime destroy sippeers id 1 causes crash due to NULL name provided to ast_variable
  • [ASTERISK-24234] - app_meetme: Crash on conference shutdown due to NULL channel passed to meetme_stasis_generate_msg()
  • [ASTERISK-24236] - res_hep_rtcp: Module incorrectly depends on pjsip
  • [ASTERISK-24237] - CDR: FRACK With PJSIP blonde transfer.
  • [ASTERISK-24241] - crash: CDRs recursively attempt to update Party B information in a multi-party bridge, overrunning the stack
  • [ASTERISK-24245] - gcc 4.1.2 complains of files that do not end with newlines
  • [ASTERISK-24249] - SIP debugs do not stop
  • [ASTERISK-24254] - CDRs: Application/args/dialplan CEP updated during dial operation
  • [ASTERISK-24264] - ARI: Adding a channel to a holding bridge automatically starts MOH
  • [ASTERISK-24290] - Endpoint identifier match value fails to parse when CIDR network format is specified
  • [ASTERISK-24301] - Security: Out of call MESSAGE requests processed via Message channel driver can crash Asterisk
  • [ASTERISK-24331] - Unexpected Errors in Asterisk Manager Interface Output

Improvement

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-12.6.0-rc1

Thank you for your continued support of Asterisk!


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!


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