[asterisk-dev] Asterisk 1.8.4-rc2 Now Available

Asterisk Development Team asteriskteam at digium.com
Mon Feb 28 11:22:07 CST 2011


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!



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