Posts Tagged 1.8
Asterisk 1.8.9.3 Now Available
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the release of Asterisk 1.8.9.3.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.8.9.3 resolves several issues reported by the community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
- — Fix ACK routing for non-2xx responses.
(Closes issue ASTERISK-19389. Reported by: Karsten Wemheuer) - — Fix regressions with regards to route-set creation on early dialogs —
(Closes issue ASTERISK-19358. Reported-by: Karsten Wemheuer)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.9.3
Asterisk 1.8.10.0-rc2 Now Available
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team is pleased to announce the second release candidate of Asterisk 1.8.10.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.8.10.0-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 deadlocks occuring in chan_agent.Bad locking order was added to chan_agent to prevent a segfault that occurred due to a race condition. This patch addresses the bad locking order by locking the channel and its private data in the correct order. This stops the deadlock, which occurred when doing multiple interactions with agents. (closes issue ASTERISK-19285) Reported by: Alex Villacis Lasso
- Include iLBC source code for distribution with AsteriskClarification regarding the iLBC source code was provided by Google, and the appropriate licenses have been included in the codecs/ilbc folder.
(closes issue: ASTERISK-18943) Reporter: Leif Madsen - Don’t do a DNS lookup on an outbound REGISTER host if there is an outbound proxy configured.
(closes issue ASTERISK-16550) reported by: Olle Johansson - Create and initialize udptl only when a dialog negotiates for image mediaPrior to this patch, the udptl struct was allocated and initialized when a dialog was associated with a peer that supported T.38, when a new SIP channel was allocated, or when an INVITE request was received. This resulted in any dialog associated with a peer that supported T.38 having udptl support assigned to it, including the UDP ports needed for communication. This patch creates and initializes the udptl structure only when the SDP for a dialog specifies that image media is supported, or when Asterisk indicates that a dialog needs to support T.38.
(closes issue ASTERISK-16698, ASTERISK-16794)
Reported by: under, Elazar; Tested by: Stefan Schmidt - Allow only one thread at time to do Asterisk cleanup/shutdownAdd locking around the really-really-quit part of the core stop/restart part. Previously more than one thread could be called to do cleanup, causing atexit handlers to be run multiple times, in turn causing segfaults. (issue ASTERISK-18883)
Patch by: Walter Doekes - Fix outbound DTMF for inband mode in chan_ooh323 This tells asterisk core to generate DTMF sounds. (Closes issue ASTERISK-19233) Reported by: Matt Behrens Patches: chan_ooh323.c.patch uploaded by Matt Behrens (License #6346)
And much more! For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-…
Thank you for your continued support of Asterisk!
Asterisk 1.8.9.1 Now Available
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the release of Asterisk 1.8.9.1.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.8.9.1 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:
- — Fixes deadlocks occuring in chan_agent —
(Closes issue ASTERISK-19285. Reported by: Alex Villacis Lasso) - — Ensure entering T.38 passthrough does not cause an infinite loop —
(Closes issue ASTERISK-18951. Reported-by: Kristijan Vrban)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.9.1
Thank you for your continued support of Asterisk!
Asterisk 1.8.10.0-rc1 Now Available
Posted by admin in asterisk, iLBC, Release Candidates, t.38 on February 1, 2012
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team is pleased to announce the first release candidate of Asterisk 1.8.10.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.8.10.0-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:
- Include iLBC source code for distribution with AsteriskClarification regarding the iLBC source code was provided by Google, and the appropriate licenses have been included in the codecs/ilbc folder.
(closes issue: ASTERISK-18943) Reporter: Leif Madsen - Don’t do a DNS lookup on an outbound REGISTER host if there is an outbound proxy configured.
(closes issue ASTERISK-16550) reported by: Olle Johansson - Create and initialize udptl only when a dialog negotiates for image mediaPrior to this patch, the udptl struct was allocated and initialized when a dialog was associated with a peer that supported T.38, when a new SIP channel was allocated, or when an INVITE request was received. This resulted in any dialog associated with a peer that supported T.38 having udptl support assigned to it, including the UDP ports needed for communication. This patch creates and initializes the udptl structure only when the SDP for a dialog specifies that image media is supported, or when Asterisk indicates that a dialog needs to support T.38.
(closes issue ASTERISK-16698, ASTERISK-16794)
Reported by: under, Elazar; Tested by: Stefan Schmidt - Allow only one thread at time to do Asterisk cleanup/shutdownAdd locking around the really-really-quit part of the core stop/restart part. Previously more than one thread could be called to do cleanup, causing atexit handlers to be run multiple times, in turn causing segfaults.
(issue ASTERISK-18883)
Patch by: Walter Doekes - Fix outbound DTMF for inband mode in chan_ooh323This tells asterisk core to generate DTMF sounds. (Closes issue ASTERISK-19233) Reported by: Matt Behrens Patches: chan_ooh323.c.patch uploaded by Matt Behrens (License #6346)
And much more! For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-…
Thank you for your continued support of Asterisk!
Asterisk 1.8.9.0 Now Available
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team is pleased to announce the release of Asterisk 1.8.9.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.8.9.0 resolves several issues reported by the community and would have not been possible without your participation.
Thank you!
The following is a sample of the issues resolved in this release:
- AST-2012-001: prevent crash when an SDP offer is received with an encrypted video stream when support for video is disabled and res_srtp is loaded.
(closes issue ASTERISK-19202) Reported by: Catalin Sanda - Handle AST_CONTROL_UPDATE_RTP_PEER frames in local bridge loop.Failing to handle AST_CONTROL_UPDATE_RTP_PEER frames in the local bridge loop causes the loop to exit prematurely. This causes a variety of negative side effects, depending on when the loop exits. This patch handles the frame by essentially swallowing the frame in the local loop, as the current channel drivers expect the RTP bridge to handle the frame, and, in the case of the local bridge loop, no additional action is necessary.
(closes issue ASTERISK-19095) Reported by: Stefan Schmidt Tested by: Matt Jordan - Fix timing source dependency issues with MOH.Prior to this patch, res_musiconhold existed at the same module priority level as the timing sources that it depends on. This would cause a problem when music on hold was reloaded, as the timing source could be changed after res_musiconhold was processed. This patch adds a new module priority level, AST_MODPRI_TIMING, that the various timing modules are now loaded at. This now occurs before loading other resource modules, such that the timing source is guaranteed to be set prior to resolving the timing source dependencies.
(closes issue ASTERISK-17474) Reporter: Luke H Tested by: Luke H, Vladimir Mikhelson, zzsurf, Wes Van Tlghem, elguero, Thomas Arimont
Patched by elguero - Fix RTP reference leak.If a blind transfer were initiated using a REFER without a prior reINVITE to place the call on hold, AND if Asterisk were sending RTCP reports, then there was a reference leak for the RTP instance of the transferrer.
(closes issue ASTERISK-19192) Reported by: Tyuta Vitali - Fix blind transfers from failing if an ‘h’ extension is present.This prevents the ‘h’ extension from being run on the transferee channel when it is transferred via a native transfer mechanism such as SIP REFER. (closes issue ASTERISK-19173) Reported by: Ross Beer Tested by: Kristjan Vrban Patches: ASTERISK-19173 by Mark Michelson (license 5049)
- Restore call progress code for analog ports.Extracting sig_analog from chan_dahdi lost call progress detection functionality. Fix analog ports from considering a call answered immediately after dialing has completed if the callprogress option is enabled. (closes issue ASTERISK-18841) Reported by: Richard Miller Patched by Richard Miller
- Fix regression that ‘rtp/rtcp set debup ip’ only works when a port was also specified.
(closes issue ASTERISK-18693) Reported by: Davide Dal Reviewed by: Walter Doekes
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.9.0
Thank you for your continued support of Asterisk!
Asterisk 1.8.8.0 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!
