[asterisk-dev] Asterisk 1.8.4 Now Available

[Asterisk Development Team]

The Asterisk Development Team has announced the release of Asterisk 1.8.4. This
release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.4 resolves several issues reported by the community.
Without your help this release would not have been possible. Thank you!

Below is a sample of the issues resolved in this release:

 * Use SSLv23_client_method instead of old SSLv2 only.
   (Closes issue #19095, #19138. Reported, patched by tzafrir. Tested by russell
   and chazzam.

 * Resolve crash in ast_mutex_init()
   (Patched by twilson)

 * 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)

 * Resolve an issue with the Asterisk manager interface leaking memory when
   disabled.
   (Reported internally by kmorgan. Patched by russellb)

 * Support greetingsfolder as documented in voicemail.conf.sample.
   (Closes issue #17870. Reported by edhorton. Patched by seanbright)

 * 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)

 * Set hangup cause in local_hangup so the proper return code of 486 instead of
   503 when using Local channels when the far sides returns a busy. Also affects
   CCSS in Asterisk 1.8+.
   (Patched by twilson)

 * Fix issues with verbose messages not being output to the console.
   (Closes issue #18580. Reported by pabelanger. Patched by qwell)

 * Fix Deadlock with attended transfer of SIP call
   (Closes issue #18837. Reported, patched by alecdavis. Tested by
   alecdavid, Irontec, ZX81, cmaj)

Includes changes per AST-2011-005 and AST-2011-006
For a full list of changes in this release candidate, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.4

Information about the security releases are available at:

http://downloads.asterisk.org/pub/security/AST-2011-005.pdf
http://downloads.asterisk.org/pub/security/AST-2011-006.pdf

Thank you for your continued support of Asterisk!