[AsteriskBrasil] Fwd: [asterisk-dev] Asterisk 1.4.41-rc1 Now Available

[Denis Galvão - Gmail]

Begin forwarded message:

> From: Asterisk Development Team <asteriskteam em digium.com>
> Date: 28 de fevereiro de 2011 14:19:44 BRT
> To: Asterisk Development Team <asteriskteam em digium.com>
> Subject: [asterisk-dev] Asterisk 1.4.41-rc1 Now Available
> Reply-To: Asterisk Developers Mailing List <asterisk-dev em lists.digium.com>
> 
> The Asterisk Development Team has announced the first release candidate of
> Asterisk 1.4.41. This release candidate is available for immediate download at
> http://downloads.asterisk.org/pub/telephony/asterisk/
> 
> The release of Asterisk 1.4.41-rc1 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 candidate:
> 
> * 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.
> 
> * 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)
> 
> * Guard against retransmitting BYEs indefinitely during attended transfers with
>  chan_sip.
>  (Review: https://reviewboard.asterisk.org/r/1077/)
> 
> For a full list of changes in this release candidate, please see the ChangeLog:
> 
> http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.41-rc1
> 
> Thank you for your continued support of Asterisk!
> 
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