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!
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