Posts Tagged 1.8
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 1.8.8.0-rc5 Now Available
Posted by admin in asterisk, Release Candidates on December 9, 2011
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:
Thank you for your continued support of Asterisk!
Asterisk 1.4.43, 1.6.2.21, and 1.8.7.2 Now Available (Security Release)
Posted by admin in asterisk, Asterisk Security Releases, Security Advisories on December 8, 2011
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced security releases for Asterisk 1.4, 1.6.2 and 1.8. The available security releases are released as versions 1.4.43, 1.6.2.21 and 1.8.7.2.
These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases
The release of Asterisk versions 1.4.43, 1.6.2.21, and 1.8.7.2 resolves an issue with possible remote enumeration of SIP endpoints with differing NAT settings.
The release of Asterisk versions 1.6.2.21 and 1.8.7.2 resolves a remote crash possibility with SIP when the “automon” feature is enabled.
The issues and resolutions are described in the AST-2011-013 and AST-2011-014 security advisories.
For more information about the details of these vulnerabilities, please read the security advisories AST-2011-013 and AST-2011-014, which were released at the same time as this announcement.
For a full list of changes in the current releases, please see the ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-…
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-…
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-…
Security advisory AST-2011-013 is available at:
Security advisory AST-2011-014 is available at:
Thank you for your continued support of Asterisk!
Asterisk 1.8.8.0-rc4 Now Available
Posted by admin in asterisk, Release Candidates on November 17, 2011
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!
Asterisk 1.8.8.0-rc3 Now Available
Posted by admin in asterisk, Release Candidates on November 10, 2011
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:
Thank you for your continued support of Asterisk!
Asterisk 1.8.8.0-rc2 Now Available
Posted by admin in asterisk, Release Candidates on October 19, 2011
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!
Asterisk 1.8.7.1 now available (Security Release)
Posted by admin in asterisk, Asterisk Security Releases, Security Advisories on October 17, 2011
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced a security release for Asterisk 1.8.
The available security release is released as version 1.8.7.1.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases
The release of Asterisk 1.8.7.1 resolves an issue with SIP URI parsing which can lead to a remotely exploitable crash:
Remote Crash Vulnerability in SIP channel driver (AST-2011-012)
The issue and resolution is described in the AST-2011-012 security advisory.
For more information about the details of this vulnerability, please read the security advisory AST-2011-012, which was released at the same time as this announcement.
For a full list of changes in the current release, please see the ChangeLog:
Security advisory AST-2011-012 is available at:
Thank you for your continued support of Asterisk!
Asterisk 1.8.8.0-rc1 Now Available
Posted by admin in asterisk, Release Candidates, sip on October 6, 2011
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:
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 1.8.7.0-rc1 Now Available
Posted by admin in asterisk, Release Candidates on September 8, 2011
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:
Thank you for your continued support of Asterisk!
