Archive for category Releases
Asterisk 1.6.2.22 Now Available
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the release of Asterisk 1.6.2.22.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.2.22 corrects two flaws in sip.conf.sample related to AST-2011-013:
- The sample file listed *two* values for the ‘nat’ option as being the default. Only ‘yes’ is the default.
- The warning about having differing ‘nat’ settings confusingly referred to both peers and users.
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-…
Thank you for your continued support of Asterisk!
Asterisk 10.0.0 Is Released
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team is proud to announce the release of Asterisk 10.0.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/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
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…
The release of Asterisk 10 would not have been possible without the support and contributions of the community.
You can find an overview of the work involved with the 10.0.0 release in the summary:
http://svn.asterisk.org/svn/asterisk/tags/10.0.0/asterisk-10.0.0-summary…
A short list of available 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.asterisk.org/svn/asterisk/branches/10/CHANGES
Also, when upgrading a system between major versions, it is imperative that you read and understand the contents of the UPGRADE.txt file, which is located at:
http://svn.asterisk.org/svn/asterisk/branches/10/UPGRADE.txt
Thank you for your continued support of Asterisk!
Asterisk 1.8.8.0 Now Available
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team is pleased to announce the release of Asterisk 1.8.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.8.8.0 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:
- 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/) - 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) - 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 - 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
For a full list of changes in this release, please see the ChangeLog:
Thank you for your continued support of Asterisk!
Asterisk 1.8.7.0 Now Available
Asterisk The Open Source PBX & Telephony Platform
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-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:
Thank you for your continued support of Asterisk!
Asterisk 1.8.6.0 Now Available
Asterisk The Open Source PBX & Telephony Platform
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:
Thank you for your continued support of Asterisk!
Asterisk 1.6.2.20 Now Available
Asterisk The Open Source PBX & Telephony Platform
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-…
Thank you for your continued support of Asterisk!
Asterisk 1.8.5.0 Now Available
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team announces the release of Asterisk 1.8.5.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
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:
Thank you for your continued support of Asterisk!
Asterisk 1.6.2.19 Now Available (Final Maintenance Release)
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the final maintenance release of Asterisk, version 1.6.2.19. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
Please note that Asterisk 1.6.2.19 is the final maintenance release from the 1.6.2 branch. Support for security related issues will continue until April 21, 2012. For more information about support of the various Asterisk branches, see https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
The release of Asterisk 1.6.2.19 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:
- Don’t broadcast FullyBooted to every AMI connection The FullyBooted event should not be sent to every AMI connection every time someone connects via AMI. It should only be sent to the user who just connected.
(Closes issue #18168. Reported, patched by FeyFre) - Fix thread blocking issue in the sip TCP/TLS implementation.
(Closes issue #18497. Reported by vois. Tested by vois, rossbeer, kowalma, Freddi_Fonet. Patched by dvossel) - Don’t delay DTMF in core bridge while listening for DTMF features.
(Closes issue #15642, #16625. Reported by jasonshugart, sharvanek. Tested by globalnetinc, jde. Patched by oej, twilson) - Fix chan_local crashs in local_fixup() Thanks OEJ for tracking down the issue and submitting the patch.
(Closes issue #19053. Reported, patched by oej) - Don’t offer video to directmedia callee unless caller offered it as well
(Closes issue #19195. Reported, patched by one47)
Additionally security announcements AST-2011-008, AST-2011-010, and AST-2011-011 have been resolved in this release.
For a full list of changes in this release, please see the ChangeLog:
Thank you for your continued support of Asterisk!
Asterisk 1.4.42 Now Available (Final Maintenance Release)
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the final maintenance release of Asterisk, version 1.4.42. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
Please note that Asterisk 1.4.42 is the final maintenance release from the 1.4 branch. Support for security related issues will continue until April 21, 2012. For more information about support of the various Asterisk branches, see https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
The release of Asterisk 1.4.42 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:
- Resolve regression with ring groups in the Dial() application
(Closes issue ASTERISK-17874. Reported by mspuhler. Patched by elguero) - Resolve deadlock when using tab completion on the ‘meetme kick’ CLI command when an invalid (non-existent) conference room is specified.
(Closes issue ASTERISK-17771. Reported, patched by zvision) - Resolve issue where voice frames could be dropped when checking for T.38 during early media.
(Closes issue ASTERISK-17705. Reported, patched by oej) - Resolve issue where DYNAMIC_FEATURES would not activate after a recent DTMF fix.
(Closes issue ASTERISK-17914. Reported by vrban. Patched by twilson)
Additionally security announcements AST-2011-010, and AST-2011-011 have been resolved in this release.
For a full list of changes in this release, please see the ChangeLog:
Thank you for your continued support of Asterisk!
Asterisk 1.8.3 Now Available
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the release of Asterisk 1.8.3. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.8.3 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:
- Resolve duplicated data in the AstDB when using DIALGROUP()
(Closes issue #18091. Reported by bunny. Patched by tilghman) - Ensure the ipaddr field in realtime is large enough to handle IPv6 addresses.
(Closes issue #18464. Reported, patched by IgorG) - Reworking parsing of mwi => lines to resolve a segfault. Also add a set of unit tests for the function that does the parsing.
(Closes issue #18350. Reported by gbour. Patched by Marquis) - When using cdr_pgsql the billsec field was not populated correctly on unanswered calls.
(Closes issue #18406. Reported by joscas. Patched by tilghman) - Resolve memory leak in iCalendar and Exchange calendaring modules.
(Closes issue #18521. Reported, patched by pitel. Tested by cervajs) - This version of Asterisk includes the new Compiler Flags option BETTER_BACKTRACES which uses libbfd to search for better symbol information within both the Asterisk binary, as well as loaded modules, to assist when using inline backtraces to track down problems.
(Patched by tilghman) - Resolve issue where no Music On Hold may be triggered when using res_timing_dahdi.
(Closes issues #18262. Reported by francesco_r. Patched by cjacobson. Tested by francesco_r, rfrantik, one47) - Resolve a memory leak when the Asterisk Manager Interface is disabled.
(Reported internally by kmorgan. Patched by russellb) - Reimplemented fax session reservation to reverse the ABI breakage introduced in r297486.
(Reported internally. Patched by mnicholson) - Fix regression that changed behavior of queues when ringing a queue member.
(Closes issue #18747, #18733. Reported by vrban. Patched by qwell.) - Resolve deadlock involving REFER.
(Closes issue #18403. Reported, tested by jthurman. Patched by jpeeler.)
Additionally, this release has the changes related to security bulletin AST-2011-002 which can be found at http://downloads.asterisk.org/pub/security/AST-2011-002.pdf
For a full list of changes in this release, please see the ChangeLog:
Thank you for your continued support of Asterisk!
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