21/10/2010 - Asterisk 1.8 PBX Now Available For Download
The Asterisk Development Team is proud to announce the release of Asterisk 1.8. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
Asterisk 1.8 is the next major release series of Asterisk. It will be a Long Term Support (LTS) release, similar to Asterisk 1.4. For more information about support time lines for Asterisk releases, see the Asterisk versions page.
http://www.asterisk.org/asterisk-versions
The release of Asterisk 1.8.0 would not have been possible without the support and contributions of the community. Since Asterisk 1.6.2, we've had over 500 reporters, more than 300 testers and greater than 200 developers contributed to this release.
You can find a summary of the work involved with the 1.8.0 release in the summary:
http://svn.asterisk.org/svn/asterisk/tags/1.8.0/asterisk-1.8.0-summary.txt
A short list of available features includes:
* Secure RTP
* IPv6 Support in the SIP channel driver
* Connected Party Identification Support
* Calendaring Integration
* A new call logging system, Channel Event Logging (CEL)
* Distributed Device State using Jabber/XMPP PubSub
* Call Completion Supplementary Services support
* Advice of Charge support
* Much, much more!
A full list of new features can be found in the CHANGES file.
http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=markup
For a full list of changes in the current release candidate, please see the
ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0
Thank you for your continued support of Asterisk!
18/10/2010 - Asterisk PBX 1.8 Release Candidate 4 Now Available
http://downloads.asterisk.org/pub/telephony/asterisk/
With all currently 1.8.0 blocker issues closed, Asterisk 1.8.0-rc4 is currently scheduled to become the full release of Asterisk 1.8.0.
All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any issues found to the issue tracker, https://issues.asterisk.org/. It is also very useful to see successful test
reports. Please post those to the asterisk-dev mailing list.
Asterisk 1.8 is the next major release series of Asterisk. It will be a Long Term Support (LTS) release, similar to Asterisk 1.4. For more information about support time lines for Asterisk releases, see the Asterisk versions page.
http://www.asterisk.org/asterisk-versions
With the availability of the Asterisk 1.8.0 release candidates, the binary add-on modules for Asterisk produced by Digium have been updated with new versions that are compatible with Asterisk 1.8. The availability of these
modules will assist with the testing of Asterisk 1.8.0 in a wider variety of situations.
This release candidate contains fixes since the last release candidate as reported by the community. A sampling of the changes in this release candidate include:
* Additional fixups in chan_gtalk that allow outbound calls to both Google
Talk and Google Voice recipients. Adds new chan_gtalk enhancements externip
and stunaddr.
(Closes issue #13971. Patched by dvossel)
* Resolve manager crash issue.
(Closes issue #17994. Reported by vrban. Patchd by dvossel)
* Documentation updates for sample configuration files.
(Closes issues #18107, #18101. Reported, patched by lathama, lmadsen)
* Resolve issue where faxdetect would only detect the first fax call in
chan_dahdi.
(Closes issue #18116. Reported by seandarcy. Patched by rmudgett)
* Resolve issue where a channel that is setup and torn down *very* quickly may not have the right call disposition or ${DIALSTATUS}.
(Closes issue #16946. Reported by davidw. Review
https://reviewboard.asterisk.org/r/740/)
* Set TCLASS field of IPv6 header when SIP QoS options are set.
(Closes issue #18099. Reported by jamesnet. Patched by dvossel)
* Resolve issue where Asterisk could crash on shutdown when using SRTP.
(Closes issue #18085. Reported by st. Patched by twilson)
* Fix issue where peers host port would be lost on a SIP reload.
(Closes issue #18135. Reported, tested by lmadsen. Patched by dvossel)
A short list of available features includes:
* Secure RTP
* IPv6 Support in the SIP channel driver
* Connected Party Identification Support
* Calendaring Integration
* A new call logging system, Channel Event Logging (CEL)
* Distributed Device State using Jabber/XMPP PubSub
* Call Completion Supplementary Services support
* Advice of Charge support
* Much, much more!
A full list of new features can be found in the CHANGES file.
http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=markup
For a full list of changes in the current release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc4
Thank you for your continued support of Asterisk!
24/09/2010 - Asterisk PBX 1.8.0 Release Candidate 2 Now Available
The Asterisk Development Team has announced the second release candidate of Asterisk PBX 1.8.0. This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
Asterisk 1.8.0-rc1 was not released due to an issue found prior to release.
* Make AMI honor enabled=no
(Closes issue #18040. Reported by: twilson
Review: https://reviewboard.asterisk.org/r/938/)
All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any issues found to the issue tracker,
https://issues.asterisk.org/. It is also very useful to see successful test reports. Please post those to the asterisk-dev mailing list.
Asterisk 1.8 is the next major release series of Asterisk. It will be a Long Term Support (LTS) release, similar to Asterisk 1.4. For more information about support time lines for Asterisk releases, see the Asterisk versions page.
http://www.asterisk.org/asterisk-versions
With the availability of the Asterisk 1.8.0 release candidates, the binary add-on modules for Asterisk produced by Digium have been updated with new versions that are compatible with Asterisk 1.8. The availability of these
modules will assist with the testing of Asterisk 1.8.0 in a wider variety of situations.
* Add slin16 support for format_wav (new wav16 file extension)
(Closes issue #15029. Reported, patched by andrew. Tested by Qwell)
* Fixes a bug in manager.c where the default configuration values weren't reset
when the manager configuration was reloaded.
(Closes issue #17917. Reported by lmadsen. Patched by bbryant)
* Various fixes for the calendar modules.
(Patched by Jan Kalab.
Reviewboard: https://reviewboard.asterisk.org/r/880/
Closes issue #17877. Review: https://reviewboard.asterisk.org/r/916/
Closes issue #17776. Review: https://reviewboard.asterisk.org/r/921/)
* Add CHANNEL(checkhangup) to check whether a channel is in the process of
being hung up.
(Closes issue #17652. Reported, patched by kobaz)
* Fix a bug with MeetMe where after announcing the amount of time left in a
conference, if music on hold was playing, it doesn't restart.
(Closes issue #17408, Reported, patched by sysreq)
* Fix interoperability problems with session timer behavior in Asterisk.
(Closes issue #17005. Reported by alexcarey. Patched by dvossel)
* Rate limit calls to fsync() to 1 per second after astdb updates. Astdb was
determined to be one of the most significant bottlenecks in SIP registration
processing. This patch improved the speed of an astdb load test by 50000%
(yes, Fifty-Thousand Percent). On this particular load test setup, this
doubled the number of SIP registrations the server could handle.
(Review: https://reviewboard.asterisk.org/r/825/)
* Don't clear the username from a realtime database when a registration
expires. Non-realtime chan_sip does not clear the username from memory when a
registration expiries so realtime probably shouldn't either.
(Closes issue #17551. Reported, patched by: ricardolandim. Patched by
mnicholson)
* Don't hang up a call on an SRTP unprotect failure. Also make it more obvious
when there is an issue en/decrypting.
(Closes issue #17563. Reported by Alexcr. Patched by sfritsch. Tested by
twilson)
* Many more issues. This is a significant upgrade over Asterisk 1.8.0 beta 5!
A short list of available features includes:
* Secure RTP
* IPv6 Support in the SIP channel driver
* Connected Party Identification Support
* Calendaring Integration
* A new call logging system, Channel Event Logging (CEL)
* Distributed Device State using Jabber/XMPP PubSub
* Call Completion Supplementary Services support
* Advice of Charge support
* Much, much more!
A full list of new features can be found in the CHANGES file.
http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=checkout
For a full list of changes in the current release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc2
Thank you for your continued support of Asterisk!
15/09/2010 - Asterisk PBX 1.6.2.12 Now Available
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.2.12 resolves several issues reported by the community and would have not been possible without your participation.
The following is a sample of the issues resolved in this release:
* Fix issue where DNID does not get cleared on a new call when using
immediate=yes with ISDN signaling.
(Closes issue #17568. Reported by wuwu. Patched by rmudgett)
* Several updates to res_config_ldap.
(Closes issue #13573. Reported by navkumar. Patched by navkumar, bencer.
Tested by suretec)
* Prevent loss of Caller ID information set on local channel after masquerade.
(Closes issue #17138. Reported by kobaz, patched by jpeeler)
* Fix SIP peers memory leak.
(Closes issue #17774. Reported, patched by kkm)
* Add Danish support to say.conf.sample
(Closes issue #17836. Reported, patched by RoadKill)
* Ensure SSRC is changed when media source is changed to resolve audio delay.
(Closes issue #17404. Reported, tested by sdolloff. Patched by jpeeler)
* Only do magic pickup when notifycid is enabled.
A new way of doing BLF pickup was introduced into 1.6.2. This feature adds a
call-id value into the XML of a SIP_NOTIFY message sent to alert a subscriber
that a device is ringing. This option should only be enabled when the new
'notifycid' option is set, but this was not the case. Instead the call-id
value was included for every RINGING Notify message, which caused a
regression for people who used other methods for call pickup.
(Closes issue #17633. Reported, patched by urosh. Patched by dvossel.
Tested by: dvossel, urosh, okrief, alecdavis)
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.12
Thank you for your continued support of Asterisk!
15/09/2010 - Asterisk PBX 1.4.36 Now Available
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.36 resolves several issues reported by the community and would have not been possible without your participation.
The following is a sample of the issues resolved in this release candidate:
* Fix issue where DNID does not get cleared on a new call when using
immediate=yes with ISDN signaling.
(Closes issue #17568. Reported by wuwu. Patched by rmudgett)
* Fix issue where SIP promiscuous redirect could fail to dial the
redirect (app_queue).
* Fixes issue with translator frame not getting freed. This issue prevented
G.729 licenses from being freed up.
(Closes issue #17630. Reported by manvirr. Patched by dvossel)
* Ensure SSRC is changed when media source is changed to resolve audio delay.
(Closes issue #17404. Reported, tested by sdolloff. Patched by jpeeler)
* Q931 - Sending PROGRESS after sending ALERTING is a protocol error.
(Closes issue #17874. Reported, patched by nic_bellamy)
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.36
Thank you for your continued support of Asterisk!
23/07/2010 - Asterisk PBX 1.6.2.10 Now Available
he Asterisk Development Team has announced the release of Asterisk PBX 1.6.2.10. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.2.10 resolves several issues reported by the community and would have not been possible without your participation.
The following are a few of the issues resolved by community developers:
* Allow users to specify a port for DUNDI peers.
(Closes issue #17056. Reported, patched by klaus3000)
* Decrease the module ref count in sip_hangup when SIP_DEFER_BYE_ON_TRANSFER is
set.
(Closes issue #16815. Reported, patched by rain)
* If there is realtime configuration, it does not get re-read on reload unless
the config file also changes.
(Closes issue #16982. Reported, patched by dmitri)
* Send AgentComplete manager event for attended transfers.
(Closes issue #16819. Reported, patched by elbriga)
* Correct manager variable 'EventList' case.
(Closes issue #17520. Reported, patched by kobaz)
In addition, changes to res_timing_pthread that should make it more stable have
also been implemented.
For a full list of changes in the current release, please see the
ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.10
Thank you for your continued support of Asterisk!
23/07/2010 - Asterisk PBX 1.4.34 Now Available
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.34 resolves several issues reported by the community and would have not been possible without your participation. Thank you!
The following are a few of the issues resolved by community developers:
* Allow users to specify a port for DUNDi peers.
(Closes issue #17056. Reported, patched by klaus3000)
* Decrease the module ref count in sip_hangup when SIP_DEFER_BYE_ON_TRANSFER is
set.
(Closes issue #16815. Reported, patched by rain)
* First caller into a dynamic conference new enters the pin once.
(Closes issue #15878. Reported, patched by pabelanger)
* Send AgentComplete manager events in the event of blind and attended
transfers.
(Closes issue #16819. Reported, patched by elbriga)
For a full list of changes in the current release, please see the
ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.34
Thank you for your continued support of Asterisk!
30/06/2010 - Asterisk libpri 1.4.11.3 Now Available
http://downloads.asterisk.org/pub/telephony/libpri/
This release fixes a regression in the calling number assignment logic:
* Calling Number assignment logic change in libpri 1.4.11. Restored the old behaviour if there is more than one calling number in the incoming SETUP message. A network provided number is reported as ANI.
(Closes issue #17495. Reported and tested by ibercom. Patched by rmudgett)
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/libpri/releases/ChangeLog-1.4.11.3
Thank you for your continued support of Asterisk!
23/06/2010 - Asterisk PBX 1.4.33.1 Released
The Asterisk Development Team has announced the release of Asterisk 1.4.33.1. This release is available for immediate download at:
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.33.1 resolves a regression involving the use of FXO signaling in chan_dahdi where a channel could continue ringing after it has been answered.
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.33.1
Thank you for your continued support of Asterisk!
03/06/2010 - Asterisk PBX 1.6.2.8 Now Available
The Asterisk Development Team has announced the release of Asterisk PBX 1.6.2.8. This release is available for immediate download at:
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.2.8 resolves several issues reported by the community, and would have not been possible without your participation.
The following are a few of the issues resolved by community developers:
* Enable auto complete for CLI command 'logger set level'.
(Closes issue #17152. Reported, patched by pabelanger)
* Make the mixmonitor thread process audio frames faster.
(Closes issue #17078. Reported, tested by geoff2010. Patched by dhubbard)
* Add missing 'useragent' field to sip-friends.sql file.
(Closes issue #17171. Reported, patched by thehar)
* Add example dialplan for dialing ISN numbers (http://www.freenum.org)
(Closes issue #17058. Reported, patched by pprindeville)
* Fix issue with double "sip:" in header field.
(Closes issue #15847. Reported, patched by ebroad)
* Add ability to generate ASCII documentation from the TeX files by running
'make asterisk.txt'.
(Closes issue #17220. Reported by lmadsen. Tested, patched by pabelanger)
* When StopMonitor() is called, ensure that it will not be restarted by a
channel event.
(Closes issue #16590. Reported, patched by kkm)
* Small error in the T.140 RTP port verbose log.
(Closes issue #16998. Reported, patched by frawd. Tested by russell)
For a full list of changes in the current release, please see the
ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.8
Thank you for your continued support of Asterisk!
--
19/12/2009 - Asterisk PBX 1.6.1.12 Now Available
The Asterisk Development Team has announced the release of Asterisk PBX 1.6.1.12.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.1.12 resolved several issues reported by the community, and would have not been possible without your participation. Thank you!
* Fix multiple issues with musiconhold, which led to classes not getting
destroyed properly.
(closes issues #16279, #16207), reported by: parisioa, dcabot, patched by:
tilghman, tested by: parisioa, tilghman
* Fix compatibility with valgrind 3.3 and older.
(noticed in issue #16388), reported by: parisioa, patched by: atis, tested
by: atis, parisioa
* Prevent double closing of FDs by EIVR
(closes issue #16305), reported by: diLLec, patched, tested by: thedavidfactor
* Send ack (response/message) after receiving manager action userevent
(closes issue #16264), reported, patched by: dimas
* Make manager response to "Action: events" finish with empty line
(closes issue #16275), reported, patched by: vnovy
This release also contains significant improvements to T.38 support. Anyone who has tried T.38 faxing in the past should try again as most problems should now be resolved.
A summary of changes in this release can be found in the release summary:
http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.6.1.12-summary.txt
For a full list of changes in this releases, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.1.12
Thank you for your continued support of Asterisk!
19/12/2009 - Asterisk PBX 1.4.28 Now Available
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.28 resolved several issues reported by the community, and would have not been possible without your participation. Thank you!
* Send ack (response/message) after receiving manager action userevent
(closes issue #16264), reported, patched by: dimas
* Do not modify the gain settings on data calls in chan_dahdi.
(closes issue #15972), reported by: udosw, patched, tested by: alecdavis
* fixes solaris segfault on dial with verbosity >= 3
(closes issue #16193), reported by: asgaroth, patched by: snuffy, tested by:
snuffy, asgaroth
* fixes conditional jump or move depending on uninitialised STACK value
(closes issue #16261), reported, patched by: edguy3
* Copy the peer CDR's userfield to the bridge CDR if it exists.
(closes issue #14590), reported by: msetim, patched by Laureano, tested by:
Laureano, mnicholson
A summary of changes in this release can be found in the release summary:
http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.4.28-summary.txt
For a full list of changes in this releases, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.28
Thank you for your continued support of Asterisk!
30/11/2009 - Asterisk PBX 1.2.37, 1.4.27.1, 1.6.0.19, and 1.6.1.11 Available
These releases have been created in response to a SIP remote crash vulnerability.
Additionally, Asterisk versions 1.4.27.1, 1.6.0.19, and 1.6.1.11 also contain an SDP regression fix as described in issue #16268.
Asterisk 1.6.0.19, and 1.6.1.11 contain an additional SDP regression fix as described by issue #16238.
Information about the SDP issues can be found at:
https://issues.asterisk.org/view.php?id=16268
https://issues.asterisk.org/view.php?id=16238
For more information about the details of this vulnerability, please read the security advisory AST-2009-010, which was released at the same time as this announcement.
The security advisory is available at
http://downloads.asterisk.org/pub/security/AST-2009-010.pdf
For a full list of changes in the current releases, please see the ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.2.37
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.27.1
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.0.19
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.1.11
Thank you for your continued support of Asterisk!
27/10/2009 - Asterisk PBX 1.6.1.8 Available for Download
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of 1.6.1.8 resolves an issue where an ACL check is not present for verifying SIP INVITEs. For more information about the details of this vulnerability, please read the security advisory AST-2009-007, which was
released at the same time as this announcement.
The Asterisk 1.6.1 series is the only fully released version which contains this vulnerability. Releases from previous branches (1.6.0, 1.4, 1.2) are not affected.
In addition, Asterisk users may notice that we skipped the version number 1.6.1.7. This was intentional, in an effort to avoid confusion about what a particular release contains. Asterisk 1.6.1.7 had candidates for release made, so backtracking on those changes in a release with the same version number might be confusing. The next release candidate, which would have been 1.6.1.7-rc3, will be released with additional changes as 1.6.1.9-rc1.
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.1.8
Release announcement AST-2009-007 is available at:
http://downloads.asterisk.org/pub/security/AST-2009-007.pdf
Thank you for your continued support of Asterisk!
21/10/2009 - Libpri-1.4.10.2 for Asterisk Released
http://downloads.asterisk.org/pub/telephony/libpri/libpri-1.4.10.2.tar.gz
This release resolves various issues found in libpri 1.4.10.1 and earlier versions related to scheduler events not being deleted and new ones being created on top of them. This can cause the scheduler to be overfilled, as well as other Q.921 related badness because of runaway scheduled events.
Note, this can only happen when Q.931 messages are attempted to be sent during a D-Channel state transient (D-Channel goes down and back up).
For a full list of changes in this release, please see the ChangeLog:
http://svn.asterisk.org/svn/libpri/tags/1.4.10.2/ChangeLog
Thank you for your continued support of Asterisk!
07/10/2009 - Asterisk PBX 1.4.27-rc2, 1.6.0.16-rc2, 1.6.1.7-rc2, and 1.6.2.0-rc3 Now Available
http://downloads.asterisk.org/pub/telephony/asterisk/
The release candidates address issues that were reported by the community and resolved since the last round of bug fix releases.
For a full list of changes in the current release candidates, please see the
ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.27-rc2
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.0.16-rc2
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.1.7-rc2
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.0-rc3
Issues found in any of these release candidates should be reported to the Asterisk issue tracker at http://issues.asterisk.org
Thank you for your continued support of Asterisk!
31/08/2009 - Asterisk PBX 1.6.0.14 and 1.6.1.5 Now Available
Asterisk 1.6.0.14 is the first full, non-security release since 1.6.0.10. The release candidate 1.6.0.11-rc1 was redone as 1.6.0.14-rc1 (which this release has been created from) after security releases were done as 1.6.0.13. Asterisk version 1.6.0.12 was released and rescinded shortly thereafter due to a failed merge.
Asterisk 1.6.1.5 is the first full, non-security release since 1.6.1.2. The release candidate 1.6.1.3-rc1 was redone as 1.6.1.5-rc1, which this release has been created from.
These releases resolve an assortment of issues in a number of areas in Asterisk.
For a summary of the changes in these releases, please see the release summaries:
http://svn.asterisk.org/svn/asterisk/tags/1.6.0.14/asterisk-1.6.0.14-summary.txt
http://svn.asterisk.org/svn/asterisk/tags/1.6.1.5/asterisk-1.6.1.5-summary.txt
For a full list of changes in these releases, please see the ChangeLogs:
http://svn.asterisk.org/svn/asterisk/tags/1.6.0.14/ChangeLog
http://svn.asterisk.org/svn/asterisk/tags/1.6.1.5/ChangeLog
The following list of issues were resolved with the participation of the community, and these releases would not have been possible without your help!
* Fix SIP transport type issues.
(closes issue #13865. Reported by st. Tested by mmichelson, Kristijan, vrban, jmacz, dvossel. Patched by: mmichelson, vrban, Kristijan)
* Fix an issue where the 'h' extension may occasionally not fire when a Dial is executed from a Macro. Debugged in #asterisk with user tompaw. Fixed by Tilghman.
* Fix MWI NOTIFY if Asterisk listens on a non-standard port (5060) (closes issue #14659. Reported by klaus3000. Tested by dvossel, klaus3000.
Patched by klaus3000, dvossel)
* Check if polarityonanswerdelay has elapsed before setting a channel as
answered after a polarity reversal.
(closes issue #13917. Reported, tested, and patched by alecdavis)
* Don't fast forward past the end of a message.
(closes issue #14554. Reported, tested, and patched by lacoursj)
* Prevent phantom calls to queue members.
(closes issue #14631. Reported, tested, and patched by latinsud)
Thank you for your continued support of Asterisk!
24/07/2009 - Asterisk-Addons 1.4.9, 1.6.0.3, and 1.6.1.1 Released
The Asterisk Development Team has announced several Asterisk-Addons releases, including Asterisk-Addons 1.4.9, 1.6.0.3, and 1.6.1.1. These releases are available for immediate download at:
http://downloads.asterisk.org/pub/telephony/asterisk/
These releases are an incremental release after some community reported issues were resolved, primarily in the MySQL and chan_mobile realms.
* Using chan_local with chan_mobile (issue #15299, affects all 1.6.x versions)
* Don't reset a reconnect time unless a reconnect really occurred (issue #15375, affects all versions)
For a full list of changes in these releases, please see the ChangeLogs:
http://svn.asterisk.org/svn/asterisk-addons/tags/1.4.9/ChangeLog
http://svn.asterisk.org/svn/asterisk-addons/tags/1.6.0.3/ChangeLog
http://svn.asterisk.org/svn/asterisk-addons/tags/1.6.1.1/ChangeLog
Thank you for your continued support of Asterisk!
22/07/2009 - Asterisk PBX 1.4.26 Released
This release resolves a large assortment of issues reported by the community.
For a summary of the changes in this release, please see the release summary:
http://svn.asterisk.org/svn/asterisk/tags/1.4.26/asterisk-1.4.26-summary.txt
For a full list of changes in this release, please see the ChangeLog:
http://svn.asterisk.org/svn/asterisk/tags/1.4.26/ChangeLog
The following list of issues were resolved with the participation of the community, and this release would not have been possible without your help!
* Fix handling of the 'state_interface' option of the 'queue add member' CLI
command.
(closes issue #15181. Reported and tested by loloski. Patch by seanbright)
* Fix a possible crash in pbx_spool.
(closes issue #15072. Reported, and patched by garlew)
* MWI NOTIFY contains a wrong URI if Asterisk listens to non-standard port and
transport.
(closes issue #14659. Reported, patch, and testing by klaus3000)
* Don't fast forward past the end of a message.
(closes issue #14554. Reported and patched by lacoursj)
* Prevent phantom calls to queue members.
(closes issue #14631. Reported and patched by latinsud)
* No audio on calls from Asterisk to various ISDN devices until DTMF sent by
caller. (closes issues #15420, #15416, #15389, #15205. Reported by scottbmilne,
avinoash, alecdavis, vinsik. Tested by scottbmilne, alecdavis. Patched by
alecdavis)
Thank you for your continued support of Asterisk!
16/06/2009 - Asterisk PBX 1.6.2.0-beta3 Now Available
http://downloads.digium.com/pub/asterisk/
This is an incremental release of the 1.6.2.0 branch as the previous beta was released just over a month ago, and many issues have been resolved since then. Included in this release are the following issues reported by the community:
* Update spiral support in trunk and 1.6.x branches to match what is in 1.4
(related to issue #13630).
* Fix RFC2833 issues with DTMF getting duplicated and with duration wrapping
over (issue #14815).
* Fix a bug where the codecs of the called party leg were not properly sent
back to the call leg when reinvited (issue #13569).
* Fix broken attended transfers (issue #15183).
* Add flags to chanspy audiohook so that audio stays in sync (issue #13745).
* Resolve issues with choppy sound when using res_timing_pthread
(issue #14412)
Additionally, an update to chan_iax2 related to issue AST-2009-001 is included
in this beta release. For more information, see:
http://downloads.asterisk.org/pub/security/AST-2009-001.html
For a full list of changes in this beta, please see the ChangeLog:
http://svn.digium.com/svn/asterisk/tags/1.6.2.0-beta3/ChangeLog
You can get more information about the new features and various changes in
Asterisk 1.6.2.0 at:
http://svn.digium.com/svn/asterisk/tags/1.6.2.0-beta3/CHANGES
And if you're upgrading from previous versions of Asterisk see this file:
http://svn.digium.com/svn/asterisk/tags/1.6.2.0-beta3/UPGRADE.txt
Issues discovered in testing of this beta can be reported at
http://issues.asterisk.org
Thank you for your continued support of Asterisk!
06/06/2009 - Asterisk PBX 1.2.33, Asterisk 1.4.25.1, Asterisk 1.6.0.10, and Asterisk 1.6.1.1 Released
http://downloads.asterisk.org/pub/telephony/asterisk/
This release fixes a REGAUTH loop related to security issue AST-2009-001. Asterisk release 1.2.33 also addresses a small compile time error in chan_spy.
For more information about the security issue, please see:
http://downloads.asterisk.org/pub/security/AST-2009-001.html
For a summary of the changes in this release, please see the release summary:
http://svn.asterisk.org/svn/asterisk/tags/1.2.33/asterisk-1.2.33-summary.txt
http://svn.asterisk.org/svn/asterisk/tags/1.4.25.1/asterisk-1.4.25.1-summary.txt
http://svn.asterisk.org/svn/asterisk/tags/1.6.0.10/asterisk-1.6.0.10-summary.txt
http://svn.asterisk.org/svn/asterisk/tags/1.6.1.1/asterisk-1.6.1.1-summary.txt
For a full list of changes in this release, please see the ChangeLog:
http://svn.asterisk.org/svn/asterisk/tags/1.2.33/ChangeLog
http://svn.asterisk.org/svn/asterisk/tags/1.4.25.1/ChangeLog
http://svn.asterisk.org/svn/asterisk/tags/1.6.0.10/ChangeLog
http://svn.asterisk.org/svn/asterisk/tags/1.6.1.1/ChangeLog
Thank you for your continued support of Asterisk!
11/05/2009 - Asterisk PBX 1.6.2.0-beta2 Now Available
http://downloads.digium.com/pub/asterisk/
This release merges in changes to the device state code which caused a performance regression in Asterisk 1.6.1 and 1.6.2. The result of this device state code review is that performance has been positively affected while maintaining the new distributed device state functionality. Additional information about these changes can be found on reviewboard at
http://reviewboard.digium.com/r/205/
In addition, this release also resolves several community reported issues.
For a full list of changes in this beta, please see the ChangeLog:
http://svn.digium.com/svn/asterisk/tags/1.6.2.0-beta2/ChangeLog
You can get more information about the new features and various changes in Asterisk 1.6.2.0 at:
http://svn.digium.com/svn/asterisk/tags/1.6.2.0-beta2/CHANGES
And if you're upgrading from previous versions of Asterisk see this file:
http://svn.digium.com/svn/asterisk/tags/1.6.2.0-beta2/UPGRADE.txt
Issues discovered in testing of this beta can be reported at
http://bugs.digium.com
Thank you for your continued support of Asterisk!
29/04/2009 - Asterisk PBX 1.6.1.0 Now Available
http://downloads.digium.com/pub/asterisk/
This is the first release in the 1.6.1 branch, which has additional features added since 1.6.0. Please see the CHANGES file for more information about the additional functionality
For those upgrading from previous versions of Asterisk, it is advisable to review the UPGRADE.txt file:
Some highlights about changes in this release:
----------------------------------------------
* It is now possible to specify a pattern match as a hint. Once a phone subscribes to something that matches the pattern a hint will be created using the contents and variables evaluated.
* IAX2 encryption support has been improved to support periodic key rotation within a call for enhanced security. The option "keyrotate" has been provided to disable this functionality to preserve backwards compatibility with older versions of IAX2 that do not support key rotation.
* res_odbc no longer has a limit of 1023 total possible unshared connections, as some people were running into this limit. This limit has been increased to 4.2 billion.
* Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given adaptive capabilities. What this means in practical terms is that if your realtime table lacks critical fields, Asterisk will now emit warnings to
that effect. Also, some of the realtime drivers have the ability (if configured) to automatically add those columns to the table with the correct type and length.
* Config file variables may now be appended to, by using the '+=' append operator. This is most helpful when working with long SQL queries in func_odbc.conf, as the queries no longer need to be specified on a single line.
* Many many other changes that are too numerous to list here. See:
http://svn.digium.com/svn/asterisk/tags/1.6.1.0/CHANGES
For a summary of the changes in this release, please see the release summary:
http://svn.digium.com/svn/asterisk/tags/1.6.1.0/asterisk-1.6.1.0-summary.txt
For a full list of changes in this release, please see the ChangeLog:
http://svn.digium.com/svn/asterisk/tags/1.6.1.0/ChangeLog
The following list of issues were resolved with the participation of the community, and this release would not have been possible without your help!
* Allow disconnect feature before a call is bridged
- Closes issue #11583. Submitted by sobomax. Tested and additional coding by sobomax, dvossel, murf.
* Update app_fax to work with spandsp-0.0.6
- Closes issue #13688. Reported by and patched by irroot.
* chan_h323 with H323Plus for TRUNK (SVN rev. 89183)
- Closes issue #11261. Reported by vhatz. Patched by jthurman.
* Wrong usage of sscanf with use of uninitialized variable caused accidental
parsing of RTP/SAVP
- Closes issue #14000. Reported and patched by folke.
* Realtime peers are never qualified after 'sip reload'
- Closes issue #14196. Reported, tested, and patched by pdf.
Thank you for your continued support of Asterisk!
22/04/2009 - Asterisk PBX 1.6.1.0-rc5 Now Available
The Asterisk Development Team is pleased to announce the fifth release candidate of Asterisk 1.6.1.0. Asterisk 1.6.1.0-rc5 is available for immediate download at http://downloads.digium.com/pub/asterisk/
This release fixes a couple of issues with realtime music on hold that could cause Asterisk to crash, and an issue that caused hungup channels to stay up, leading to 100% CPU usage. Additionally, several minor issues and edge case scenarios have been resolved.
For a full list of changes in this release candidate, please see the ChangeLog:
http://svn.digium.com/svn/asterisk/tags/1.6.1.0-rc5/ChangeLog
Issues found in this release candidate can be reported at http://bugs.digium.com
Thank you for your continued support of Asterisk!
24/02/2009 - Asterisk PBX 1.6.0.6 released
The Asterisk.org development team is proud to announce the release of Asterisk 1.6.0.6. This release is available for download from http://downloads.digium.com/.
This release is a significant bug fix update for the 1.6.0 release series.
In addition, this release is recommended for all users of the Asterisk GUI. Two issues with the manager interface have been resolved. The first being with the manager interface improperly handling async. requests from the GUI (see issue #14364). It resulted in manager session file descriptors being improperly destroyed and overwritten. The other being an issue with the Originate action that would cause issues with the GUI. They have both been resolved in this release.
The following list of bugs were resolved with the participation of the community, and this release would not have been possible without your help!
* Set the initiator attribute to lowercase in our replies when receiving calls. This attribute contains a JID that identifies the initiator of the GoogleTalk voice session. The GoogleTalk client discards Asterisk's replies if the
initiator attribute contains uppercase characters.
- Closes issue #13984. Submitted and tested by: jcovert
* Fix odd "thank you" sound playing behavior in app_queue.c
- Closes issue #14227. Reported and tested by: caspy
* Don't do an SRV lookup if a port is specified RFC 3263 says to do A record lookups on a hostname if a port has been specified, so that's what we're going to do. See section 4.2.
- Closes issue #14419. Reported and patched by: klaus3000
* Properly set "seen" and "unseen" flags when moving messages from the new to the old folder when using IMAP for voicemail storage.
- Closes issue #13905. Reported and patched by: jaroth
* Fix devicestate problems for "always-on" agent channels.
- Closes issue #14173. Reported by: nathan. Tested by: nathan, aramirez
For a full list of changes, see the ChangeLog:
http://svn.digium.com/svn-view/asterisk/tags/1.6.0.6/ChangeLog?view=co
Thank you for your support of Asterisk!
