[asterisk-announce] Asterisk Now Available

Asterisk Development Team asteriskteam at digium.com
Tue Apr 26 12:04:39 CDT 2011

The Asterisk Development Team has announced the release of Asterisk
This release is available for immediate download at

The release of Asterisk 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

 * Guard against retransmitting BYEs indefinitely during attended transfers with
   (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:


For a full list of changes in this release, please see the ChangeLog:


Thank you for your continued support of Asterisk!

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