[AsteriskBrasil] Fwd: [asterisk-dev] Asterisk 1.8.4 Now Available

[Denis Galvão - Gmail]

Begin forwarded message:

> From: Asterisk Development Team <asteriskteam em digium.com>
> Date: 10 de maio de 2011 11:38:48 BRT
> To: Asterisk Development Team <asteriskteam em digium.com>
> Subject: [asterisk-dev] Asterisk 1.8.4 Now Available
> Reply-To: Asterisk Developers Mailing List <asterisk-dev em lists.digium.com>
> 
> 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!
> 
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