[asterisk-announce] Asterisk 1.8.4 Now Available

Asterisk Development Team asteriskteam at digium.com
Tue May 10 09:38:48 CDT 2011

The Asterisk Development Team has announced the release of Asterisk 1.8.4. This
release is available for immediate download at

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
   (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

 * 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:


Information about the security releases are available at:


Thank you for your continued support of Asterisk!

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