ASTERWEB Blog

Rilasciato Asterisk 1.8.8.0-rc5

Il giorno 09 dicembre, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Asterisk 1.8.8.0-rc5

Dal post originale:
he 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

Rilasciato Asterisk 1.8.8.0-rc4

Il giorno 17 novembre, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Asterisk 1.8.8.0-rc4

Dal post originale:

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!

Rilasciato Asterisk 10.0.0-beta2

Il giorno 27 settembre, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Asterisk 10.0.0-beta2

Dal post originale:
The Asterisk Development Team is pleased to announce the second beta release of
Asterisk 10.0.0. 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-a...

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.

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
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://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-...

Rilasciato Astrisk 1.8.7.0

Il giorno 23 settembre, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Asterisk 1.8.7

Dal post originale:
he 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-goog...

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

Rilasciato Asterisk 1.8.6.0

Il giorno 31 agosto, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Asterisk 1.8.6

Dal post originale:
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

Rilasciato Asterisk 1.6.20.0

Il giorno 08 agosto, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Asterisk 1.6.20.0

Dal post originale:
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-...

Rilasciato Asterisk 10.0.0 Beta 1

Il giorno 22 luglio, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Asterisk 10.0.0 Beta 1

Dal post originale:
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-a...

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://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-...

Rilasciato Asterisk 1.8.5

Il giorno 11 luglio, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Asterisk 1.8.5

Dal post originale:
The release of Asterisk 1.8.5.0 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 Deadlock with attended transfer of SIP call
(Closes issue #18837. Reported, patched by alecdavis. Tested by Irontec, ZX81,
cmaj)
Fixes thread blocking issue in the sip TCP/TLS implementation.
(Closes issue #18497. Reported by vois. Patched by dvossel. Tested by vois,
rossbeer, kowalma, Freddi_Fonet)
Be more tolerant of what URI we accept for call completion PUBLISH requests.
(Closes issue #18946. Reported by GeorgeKonopacki. Patched by mmichelson)
Fix a nasty chanspy bug which was causing a channel leak every time a spied on
channel made a call.
(Closes issue #18742. Reported by jkister. Tested by jcovert, jrose)
This patch fixes a bug with MeetMe behavior where the 'P' option for always
prompting for a pin is ignored for the first caller.
(Closes issue #18070. Reported by mav3rick. Patched by bbryant)
Fix issue where Asterisk does not hangup a channel after endpoint hangs up. If
the call that the dialplan started an AGI script for is hungup while the AGI
script is in the middle of a command then the AGI script is not notified of
the hangup.
(Closes issue #17954, #18492. Reported by mn3250, devmod. Patched by rmudgett)
Resolve issue where leaving a voicemail, the MWI message is never sent. The
same thing happens when checking a voicemail and marking it as read.
(Closes issue ASTERISK-18002. Reported by Leif Madsen. Resolved by Richard
Mudgett)
Resolve issue where wait for leader with Music On Hold allows crosstalk
between participants. Parenthesis in the wrong position. Regression from issue
#14365 when expanding conference flags to use 64 bits.
(Closes issue #18418. Reported by MrHanMan. Patched by rmudgett)

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

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