Release Candidate 1 of Asterisk 1.8.5 Is Ready For Download.

Asterisk 1.8.5 - RC1 http://snapvoip.blogspot.com/
The Asterisk Development Team just announced the first release candidate of Asterisk 1.8.5 branch and is available for download at Asterisk site, immediately.

The developer team thanks the community for the involving and reporting of issues with the previous release.
The following is a sample of the issues resolved in this release candidate:

• Fix Deadlock with attended transfer of SIP call Closes issue #18837. Reported, patched by alecdavis. Tested by Irontec, ZX81,cmaj)
• Fixes thread blocking issue in the sip TCP/TLS implementation.(Closes issue #18497. Reported by vois. Patched by dvossel. Tested by vois, rossbeer, kowalma, Freddi_Fonet)
• Be more tolerant of what URI we accept for call completion PUBLISH requests.(Closes issue #18946. Reported by GeorgeKonopacki.Patched by mmichelson)
• Fix a nasty chanspy bug which was causing a channel leak every time a spied on channel made a call.(Closes issue #18742. Reported by jkister. Tested by jcovert, jrose)
• This patch fixes a bug with MeetMe behavior where the 'P' option for alwaysprompting for a pin is ignored for the first caller.(Closes issue #18070. Reported by mav3rick. Patched by bbryant)
• Fix issue where Asterisk does not hangup a channel after endpoint hangs up. If he call that the dialplan started an AGI script for is hungup while the AGI script is in the middle of a command then the AGI script is not notified of the hangup. (Closes issue #17954, #18492. Reported by mn3250, devmod. Patched by rmudgett)
• Resolve issue where leaving a voicemail, the MWI message is never sent. The same thing happens when checking a voicemail and marking it as read.(Closes issue ASTERISK-18002. Reported by Leif Madsen. Resolved by Richard Mudgett)
• Resolve issue where wait for leader with Music On Hold allows crosstalk between participants. Parenthesis in the wrong position. Regression from issue #14365 when expanding conference flags to use 64 bits. (Closes issue #18418. Reported by MrHanMan. Patched by rmudgett)
• Fix timerfd locking issue.(Closes ASTERISK-17867, ASTERISK-17415. Patched by kobaz)

For a full list of changes in this release candidate, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.5-rc1

Asterisk 1.8.4-rc2 Is Available For Download

http://snapvoip.blogspot.com/
The Asterisk Development Team just announced the release candidate 2 of
Asterisk 1.8.4, Asterisk 1.8.4-rc2, which is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
This Asterisk 1.8.4-rc2 release  resolves several issues reported by the
community. Including but not limited to the following  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