| For the interest of our members 
 Asterisk 1.8.4 Now Available.
 
 Asterisk_OSR_ 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/pu ... isk/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!
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