Begin forwarded message:
> From: Asterisk Development Team <asteriskt...@digium.com> > Date: 28 de fevereiro de 2011 14:22:07 BRT > To: Asterisk Development Team <asteriskt...@digium.com> > Subject: [asterisk-dev] Asterisk 1.8.4-rc2 Now Available > Reply-To: Asterisk Developers Mailing List <asterisk-...@lists.digium.com> > > The Asterisk Development Team has announced the second release candidate of > Asterisk 1.8.4. This release candidate is available for immediate download at > http://downloads.asterisk.org/pub/telephony/asterisk/ > > The release of Asterisk 1.8.4-rc2 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: > > * 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) > > Asterisk 1.8.4-rc1 was not released due to a blocking issue found prior to > release. An additional fix was merged into Asterisk 1.8.4-rc2: > > * Fix Deadlock with attended transfer of SIP call > (Closes issue #18837. Reported, patched by alecdavis. Tested by > alecdavid, Irontec, ZX81, cmaj) > > > 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-rc2 > > Thank you for your continued support of Asterisk! > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev
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