Latest News About Asterisk » Release Candidates

Asterisk 10.0.0-rc3 Now Available

admin — Mon, 12 Dec 2011 07:22:14 +0000

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team has announced the third release candidate of Asterisk 10.0.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 10.0.0-rc3 resolves several issues reported by the community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release candidate:

Add ASTSBINDIR to the list of configurable pathsThis patch also makes astdb2sqlite3 and astcanary use the configured directory instead of relying on $PATH.
Don’t crash on INFO automon request with no channel
AST-2011-014. When automon was enabled in features.conf, it was possible to crash Asterisk by sending an INFO request if no channel had been created yet.
Fixed crash from orphaned MWI subscriptions in chan_sip
This patch resolves the issue where MWI subscriptions are orphaned by subsequent SIP SUBSCRIBE messages.
Fix a change in behavior in ‘database show’ from 1.8.In 1.8 and previous versions, one could use any fullword portion of the key name, including the full key, to obtain the record. Until this patch, this did not work for the full key.
Default to nat=yes; warn when nat in general and peer differ
AST-2011-013. It is possible to enumerate SIP usernames when the general and user/peer nat settings differ in whether to respond to the port a request is sent from or the port listed for responses in the Via header. In 1.4 and
1.6.2, this would mean if one setting was nat=yes or nat=route and the other was either nat=no or nat=never. In 1.8 and 10, this would mean when one was nat=force_rport and the other was nat=no.In order to address this problem, it was decided to switch the default behavior to nat=yes/force_rport as it is the most commonly used option and to strongly discourage setting nat per-peer/user when at all possible.
Fixed SendMessage stripping extension from To: header in SIP MESSAGEWhen using the MessageSend application to send a SIP MESSAGE to a non-peer, chan_sip stripped off the extension and failed to add it back to the sip_pvt structure before transmitting. This patch adds the full URI passed in from the message core to the sip_pvt structure.

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.0.0-rc3

Thank you for your continued support of Asterisk!

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Asterisk 1.8.8.0-rc5 Now Available

admin — Fri, 09 Dec 2011 07:19:22 +0000

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team has announced the fifth release candidate of Asterisk 1.8.8.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.8.0-rc5 resolves several issues reported by the community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release candidate:

Don’t crash on INFO automon request with no channel
AST-2011-014. When automon was enabled in features.conf, it was possible to crash Asterisk by sending an INFO request if no channel had been created yet.
Fixed crash from orphaned MWI subscriptions in chan_sip
This patch resolves the issue where MWI subscriptions are orphaned by subsequent SIP SUBSCRIBE messages.
Default to nat=yes; warn when nat in general and peer differ
AST-2011-013. It is possible to enumerate SIP usernames when the general and user/peer nat settings differ in whether to respond to the port a request is sent from or the port listed for responses in the Via header. In 1.4 and 1.6.2, this would mean if one setting was nat=yes or nat=route and the other was either nat=no or nat=never. In 1.8 and 10, this would mean when one was nat=force_rport and the other was nat=no.In order to address this problem, it was decided to switch the default behavior to nat=yes/force_rport as it is the most commonly used option and to strongly discourage setting nat per-peer/user when at all possible.

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.8.0-rc5

Thank you for your continued support of Asterisk!

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Asterisk 1.8.8.0-rc4 Now Available

admin — Thu, 17 Nov 2011 08:39:41 +0000

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team has announced the fourth release candidate of Asterisk 1.8.8.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.8.0-rc4 resolves a particular issue with BLF subscriptions. A change in Asterisk 1.8.8.0-rc3 had the potential to cause a segfault, and this release candidate was created to resolve that.

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.8.0-rc4

Thank you for your continued support of Asterisk!

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Asterisk 10.0.0-rc2 Now Available

admin — Wed, 16 Nov 2011 03:45:18 +0000

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team is pleased to announce the second release candidate of Asterisk 10.0.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

All Asterisk users are encouraged to participate in the Asterisk 10 testing process. Please report any issues found to the issue tracker, https://issues.asterisk.org/jira. It is also very useful to see successful test reports. Please post those to the asterisk-dev mailing list.

All Asterisk users are invited to participate in the #asterisk-testing channel on IRC to work together in testing the many parts of Asterisk.

Asterisk 10 is the next major release series of Asterisk. It will be a Standard support release, similar to Asterisk 1.6.2. For more information about support time lines for Asterisk releases, see the Asterisk versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

A short list of features includes:

T.38 gateway functionality has been added to res_fax.
Protocol independent out-of-call messaging support. Text messages not
associated with an active call can now be routed through the Asterisk

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Asterisk 10.0.0-rc1 Now Available

admin — Thu, 10 Nov 2011 06:40:49 +0000

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team is pleased to announce the first release candidate of Asterisk 10.0.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

All Asterisk users are invited to participate in the #asterisk-testing channel on IRC to work together in testing the many parts of Asterisk.

A short list of features includes:

T.38 gateway functionality has been added to res_fax.
Protocol independent out-of-call messaging support. Text messages not associated with an active call can now be routed through the Asterisk dialplan. SIP and XMPP are supported so far.
New highly optimized and customizable ConfBridge application capable of mixing audio at sample rates ranging from 8kHz-192kHz (More information available at https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10 )
Addition of video_mode option in confbridge.conf to provide basic video conferencing in the ConfBridge() dialplan application.
Support for defining hints has been added to pbx_lua.
Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB).
Much, much more!

A full list of new features can be found in the CHANGES file.

http://svnview.digium.com/svn/asterisk/branches/10/CHANGES

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

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

Thank you for your continued support of Asterisk!

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Asterisk 1.8.8.0-rc3 Now Available

admin — Thu, 10 Nov 2011 06:38:37 +0000

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team has announced the third release candidate of Asterisk 1.8.8.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.8.0-rc3 resolves several issues reported by the community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release candidate:

Prevent BLF subscriptions from causing deadlocks.
(Closes issue ASTERISK-18663)
Review: https://reviewboard.asterisk.org/r/1563/
Fix deadlock if peer is destroyed while sending MWI notice.
(Closes issue ASTERISK-18747)
Reported by: Gregory Hinton Nietsky
Fix issue with setting defaultenabled on categories that are already enabled by default.
(Closes issue ASTERISK-18738)
Reported by: Paul Belanger

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.8.0-rc3

Thank you for your continued support of Asterisk!

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Asterisk 1.8.8.0-rc2 Now Available

admin — Wed, 19 Oct 2011 05:23:16 +0000

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team has announced the second release candidate of Asterisk 1.8.8.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.8.0-rc2 resolves several issues reported by the community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release candidate:

Fix remote Crash Vulnerability in SIP channel driver (AST-2011-012)
http://downloads.asterisk.org/pub/security/AST-2011-012.pdf
Fix locking order in app_queue.c which caused deadlocks
(Closes issue ASTERISK-18101. Reported by Paul Rolfe, patched by Gregory Nietsky)
(Closes issue ASTERISK-18487. Reported by Jason Legault, patched by Gregory Nietsky)
Fix regression in configure script for libpri capability checks
(Closes issue ASTERISK-18687. Reported by norbert, patched by Richard Mudgett)
Properly ignore AST_CONTROL_UPDATE_RTP_PEER in more places
(Closes issue ASTERISK-18610. Reported by Kristijan_Vrban, patched by Terry Wilson, and again by Kristijan_Vrban)
Fix issue with removing peers by IP
(Closes issue ASTERISK-18696. Reported by rsw686, patched by Terry Wilson)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.8.0-rc2

Thank you for your continued support of Asterisk!

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Asterisk 1.8.8.0-rc1 Now Available

admin — Thu, 06 Oct 2011 09:12:22 +0000

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team announces the first release candidate of Asterisk 1.8.8.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.8.0-rc1 resolves several issues reported by the community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release candidate:

Updated SIP 484 handling; added Incomplete control frame When a SIP phone uses the dial application and receives a 484 Address
Incomplete response, if overlapped dialing is enabled for SIP, then the 484 Address Incomplete is forwarded back to the SIP phone and the HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete application dialplan logic was automatically triggered; now, explicit dialplan usage of the application is required.
(Closes ASTERISK-17288. Reported by: Mikael Carlsson Tested by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/1416/)
Prevent IAX2 from getting IPv6 addresses via DNS IAX2 does not support IPv6 and getting such addresses from DNS can cause error messages on the remote end involving bad IPv4 address casts in the presence of IPv6/IPv4 tunnels.
(Closes issue ASTERISK-18090. Patched by Kinsey Moore)
Fix bad RTP media bridges in directmedia calls on peers separated by multiple Asterisk nodes.
(Closes issue ASTERISK-18340. Reported by: Thomas Arimont. Closes issue ASTERISK-17725. Reported by: kwk. Tested by: twilson, jrose)
Fix crashes in ast_rtcp_write()
(Closes issue ASTERISK-18570)
Related issues that look like they are the same problem:
(Issue ASTERISK-17560, ASTERISK-15406, ASTERISK-15257, ASTERISK-13334, ASTERISK-9977, ASTERISK-9716)
Review: https://reviewboard.asterisk.org/r/1444/
Patched by: Russell Bryant
Fix for incorrect voicemail duration in external notifications. This patch fixes an issue where the voicemail duration was being reported with a duration significantly less than the actual sound file duration.
(Closes ASTERISK-16981. Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1443)
Prevent segfault if call arrives before Asterisk is fully booted.
(Patched by alecdavis. https://reviewboard.asterisk.org/r/1407/)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.8.0-rc1

Thank you for your continued support of Asterisk!

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Asterisk 1.8.7.0-rc1 Now Available

admin — Thu, 08 Sep 2011 03:56:23 +0000

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team announces the first release candidate of Asterisk 1.8.7.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.7.0-rc1 resolves several issues reported by the community and would have not been possible without your participation.
Thank you!

Please note that a significant numbers of changes and fixes have gone into features.c in this release (call parking, built-in transfers, call pickup, etc.) and a testing focus on those particular areas would be appreciated.

Also, a known issue with the iLBC installation script get_ilbc_source.sh found in the contrib/scripts/ sub-directory will be resolved prior to the full release of Asterisk 1.8.7.0.

The following is a sample of the issues resolved in this release candidate:

Added the ‘storesipcause’ option to sip.conf to allow the user to disable the setting of HASH(SIP_CAUSE,) on the channel. Having chan_sip set HASH(SIP_CAUSE,) on the channel carries a significant performance penalty because of the usage of the MASTER_CHANNEL() dialplan function.We’ve decided to disable this feature by default in future 1.8 versions. This would be an unexpected behavior change for anyone depending on that SIP_CAUSE update in their dialplan. Please refer to the asterisk-dev mailing list more information:
http://lists.digium.com/pipermail/asterisk-dev/2011-August/050626.html
Significant fixes and improvements to parking lots.
(Closes issues ASTERISK-17183, ASTERISK-17870, ASTERISK-17430, ASTERISK-17452, ASTERISK-17452, ASTERISK-15792. Reported by: David Cabrejos, Remi Quezada, Philippe Lindheimer, David Woolley, Mat Murdock. Patched by: rmudgett)
Numerous issues have been reported for deadlocks that are caused by a blocking read in res_timing_timerfd on a file descriptor that will never be written to.
A change to Asterisk adds some checks to make sure that the timerfd is both valid and armed before calling read(). Should fix: ASTERISK-18142, ASTERISK-18197, ASTERISK-18166 and possibly others.(In essence, this change should make res_timing_timerfd usable.)
Resolve segfault when publishing device states via XMPP and not connected.
(Closes issue ASTERISK-18078. Reported, patched by: Michael L. Young. Tested by Jonathan Rose)
Refresh peer address if DNS unavailable at peer creation.
(Closes issue ASTERISK-18000)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.7.0-rc1

Thank you for your continued support of Asterisk!

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Asterisk 1.8.6.0-rc3 Now Available

admin — Thu, 25 Aug 2011 10:05:34 +0000

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team has announced the third release candidate of Asterisk 1.8.6.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.6.0-rc3 resolves an issue reported by the community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release candidate:

Use of DESTDIR causes install to fail
(Closes issue ASTERISK-18290. Reported by Paul Belanger and others, patched by Jason Parker)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.6.0-rc3

Thank you for your continued support of Asterisk!

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