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New Releases for Asterisk Are Now Available

Recently The Asterisk Development Team announced the release of Asterisk versions 11.0.2, 10.10.1 and 1.8.18.1 and made them available for immediate download at:

All of the releases resolves one or more issues reported by the community, without whose participation it wouldn’t have been possible.

The following is the issue resolved in this release:

  •  chan_local: Fix local_pvt ref leak in local_devicestate(). (Closes issue ASTERISK-20769. Reported by rmudgett)

Please read the change logs for a full list of changes. Thank you for your continued support of Asterisk!

Asterisk 1.8.9.0 Now Available

The Asterisk Development Team is pleased to announce the release of Asterisk 1.8.9.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.9.0 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:

* AST-2012-001: prevent crash when an SDP offer is received with an encrypted video stream when support for video is disabled and res_srtp is loaded. (closes issue ASTERISK-19202)
Reported by: Catalin Sanda

* Handle AST_CONTROL_UPDATE_RTP_PEER frames in local bridge loop. Failing
to handle AST_CONTROL_UPDATE_RTP_PEER frames in the local bridge loop
causes the loop to exit prematurely. This causes a variety of negative side
effects, depending on when the loop exits. This patch handles the frame by
essentially swallowing the frame in the local loop, as the current channel
drivers expect the RTP bridge to handle the frame, and, in the case of the
local bridge loop, no additional action is necessary.
(closes issue ASTERISK-19095) Reported by: Stefan Schmidt Tested
by: Matt Jordan

* Fix timing source dependency issues with MOH. Prior to this patch,
res_musiconhold existed at the same module priority level as the timing
sources that it depends on. This would cause a problem when music on
hold was reloaded, as the timing source could be changed after
res_musiconhold was processed. This patch adds a new module priority
level, AST_MODPRI_TIMING, that the various timing modules are now loaded
at. This now occurs before loading other resource modules, such
that the timing source is guaranteed to be set prior to resolving
the timing source dependencies.
(closes issue ASTERISK-17474) Reporter: Luke H Tested by: Luke H,
Vladimir Mikhelson, zzsurf, Wes Van Tlghem, elguero, Thomas Arimont
Patched by elguero

* Fix RTP reference leak. If a blind transfer were initiated using a
REFER without a prior reINVITE to place the call on hold, AND if Asterisk
were sending RTCP reports, then there was a reference leak for the
RTP instance of the transferrer.
(closes issue ASTERISK-19192) Reported by: Tyuta Vitali

* Fix blind transfers from failing if an ‘h’ extension
is present. This prevents the ‘h’ extension from being run on the
transferee channel when it is transferred via a native transfer
mechanism such as SIP REFER. (closes issue ASTERISK-19173) Reported
by: Ross Beer Tested by: Kristjan Vrban Patches: ASTERISK-19173 by
Mark Michelson (license 5049)

* Restore call progress code for analog ports. Extracting sig_analog
from chan_dahdi lost call progress detection functionality. Fix
analog ports from considering a call answered immediately after
dialing has completed if the callprogress option is enabled.
(closes issue ASTERISK-18841)
Reported by: Richard Miller Patched by Richard Miller

* Fix regression that ‘rtp/rtcp set debup ip’ only works when a port
was also specified.
(closes issue ASTERISK-18693) Reported by: Davide Dal Reviewed by:
Walter Doekes

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.9.0

Thank you for your continued support of Asterisk!

Asterisk 1.8.7.0 Now Available

The Asterisk Development Team announces the release of Asterisk 1.8.7.0.

This  release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.7.0 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.).

NOTE:

Recently, we were notified that the mechanism included in our Asterisk source code releases to download and build support for the iLBC codec had stopped working correctly; a little investigation revealed that this occurred because of some changes on the ilbcfreeware.org website. These changes occurred as a result of Google’s acquisition of GIPS, who produced (and provided licenses for) the iLBC codec.

If you are a user of Asterisk and iLBC together, and you’ve already executed a license agreement with GIPS, we believe you can continue using iLBC with Asterisk. If you are a user of Asterisk and iLBC together, but you had not executed a license agreement with GIPS, we encourage you to research the situation and consult with your own legal representatives to determine what actions you may want to take (or avoid taking).

More information is available on the Asterisk blog:

http://blogs.asterisk.org/2011/09/19/ilbc-support-in-asterisk-after-googles-acquisition-of-gips/

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

* 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)

* Fix the missing DAHDI channels when using the newer chan_dahdi.conf sections for channel configuration.

(Closes issue ASTERISK-18496. Reported by Sean Darcy. Patched by Richard Mudgett)

* Remove unnecessary libpri dependency checks in the configure script.

(Closes issue ASTERISK-18535. Reported by Michael Keuter. Patched by Richard Mudgett)

* Update get_ilbc_source.sh script to work again.

(Closes issue ASTERISK-18412)

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

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

Thank you for your continued support of Asterisk!

Asterisk 1.8.6.0 Now Available

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

The release of Asterisk 1.8.6.0 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:

* Fix an issue with Music on Hold classes losing files in playlist when
realtime
is used.
(Closes issue ASTERISK-17875. Reported by David Cunningham. Patched
by Igor
Goncharovsky)

* Resolve a potential crash in chan_sip when utilizing auth= and
performing a
‘sip reload’ from the console.
(Closes issue ASTERISK-17939. Reported by wdoekes. Patched by Richard
Mudgett)

* Address some improper sql statements in res_odbc that would cause an
update
to fail on realtime peers due to trying to set as “(NULL)” rather than an
actual NULL.
(Closes issue ASTERISK-17791. Reported by marcelloceschia. Patched by
Tilghman
Lesher)

* Resolve issue where 403 Forbidden would always be sent maximum number
of times
regardless to receipt of ACK.
(Patched by Richard Mudgett)

* Resolve issue where if a call to MeetMe includes both the dynamic(D) and
always request PIN(P) options, MeetMe will ask for the PIN two times:
once for
creating the conference and once for entering the conference.
(Patched by Kinsey Moore)

* Fix New Zealand indications profile based on
http://www.telepermit.co.nz/TNA102.pdf
(Closes issue ASTERISK-16263. Reported, Patched by richardf)

* Segfault in shell_helper in func_shell.c
(Closes issue ASTERISK-18109. Reported by Michael Myles, patched by
Richard
Mudgett)

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

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

Thank you for your continued support of Asterisk!

Asterisk 1.6.2.20 Now Available

The Asterisk Development Team announces the release of Asterisk 1.6.2.20. This
release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.6.2.20 resolves a regression that was introduced just
prior to the release of Asterisk 1.6.2.19.

* Fix reload crash caused by destroying default parking lot.
(Closes issue ASTERISK-18103. Reported by 808blogger. Patched by jrose.

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

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.20

Thank you for your continued support of Asterisk!

Asterisk 10.0.0 Beta 1 Now Available

The Asterisk Development Team is pleased to announce the first beta release of
Asterisk 10.0.0-beta1. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

With the release of the Asterisk 10 branch, the preceding ’1.’ has been removed
from the version number per the blog post available at
http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/

All interested users of Asterisk 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.
Additionally users can make use of the RPM and DEB packages now being built for
all Asterisk releases. More information available at
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages

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 included 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
* 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://svn.digium.com/view/asterisk/branches/10/CHANGES?view=checkout

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

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.0.0-beta1

Thank you for your continued support of Asterisk!