[asterisk-dev] Asterisk 1.6.2.18 Now Available

[Asterisk Development Team]

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!