Asterisk Development Team
2011-Apr-26 17:04 UTC
[asterisk-users] Asterisk 1.6.2.18 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.6.2.18. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.18 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: * Only offer codecs both sides support for directmedia. (Closes issue #17403. Reported, patched by one47) * Resolution of several DTMF based attended transfer issues. (Closes issue #17999, #17096, #18395, #17273. Reported by iskatel, gelo, shihchuan, grecco. Patched by rmudgett) NOTE: Be sure to read the ChangeLog for more information about these changes. * Resolve deadlocks related to device states in chan_sip (Closes issue #18310. Reported, patched by one47. Patched by jpeeler) * Fix channel redirect out of MeetMe() and other issues with channel softhangup (Closes issue #18585. Reported by oej. Tested by oej, wedhorn, russellb. Patched by russellb) * Fix voicemail sequencing for file based storage. (Closes issue #18498, #18486. Reported by JJCinAZ, bluefox. Patched by jpeeler) * Guard against retransmitting BYEs indefinitely during attended transfers with chan_sip. (Review: https://reviewboard.asterisk.org/r/1077/) In addition to the changes listed above, commits to resolve security issues AST-2011-005 and AST-2011-006 have been merged into this release. More information about AST-2011-005 and AST-2011-006 can be found at: http://downloads.asterisk.org/pub/security/AST-2011-005.pdf http://downloads.asterisk.org/pub/security/AST-2011-006.pdf For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.18 Thank you for your continued support of Asterisk!
Hi, I'm about to deliver a production system based on Debian Squeeze and Asterisk 1.6.2.9-2+squeeze1 from the Debian repositories. Asterisk 1.8 packages for Debian & Ubuntu are available from packages.asterisk.org. Observing some recent discussions on this list, it seems that 1.8 might not yet be ready for production use. Would whoever kindly makes the Asterisk 1.8 packages available also consider doing that for 1.6 releases? If the build environment has been set up for 1.8, I'd imagine it would be easy to set up something similar for 1.6 releases? kind regards, Jan On 27/04/11 05:04, Asterisk Development Team wrote:> The Asterisk Development Team has announced the release of Asterisk 1.6.2.18. > This release is available for immediate download at > http://downloads.asterisk.org/pub/telephony/asterisk/ > > The release of Asterisk 1.6.2.18 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: > > * Only offer codecs both sides support for directmedia. > (Closes issue #17403. Reported, patched by one47) > > * Resolution of several DTMF based attended transfer issues. > (Closes issue #17999, #17096, #18395, #17273. Reported by iskatel, gelo, > shihchuan, grecco. Patched by rmudgett) > NOTE: Be sure to read the ChangeLog for more information about these changes. > > * Resolve deadlocks related to device states in chan_sip > (Closes issue #18310. Reported, patched by one47. Patched by jpeeler) > > * Fix channel redirect out of MeetMe() and other issues with channel softhangup > (Closes issue #18585. Reported by oej. Tested by oej, wedhorn, russellb. > Patched by russellb) > > * Fix voicemail sequencing for file based storage. > (Closes issue #18498, #18486. Reported by JJCinAZ, bluefox. Patched by > jpeeler) > > * Guard against retransmitting BYEs indefinitely during attended transfers with > chan_sip. > (Review: https://reviewboard.asterisk.org/r/1077/) > > In addition to the changes listed above, commits to resolve security issues > AST-2011-005 and AST-2011-006 have been merged into this release. More > information about AST-2011-005 and AST-2011-006 can be found at: > > http://downloads.asterisk.org/pub/security/AST-2011-005.pdf > http://downloads.asterisk.org/pub/security/AST-2011-006.pdf > > For a full list of changes in this release, please see the ChangeLog: > > http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.18 > > Thank you for your continued support of Asterisk! > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >