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:
Asterisk 1.8.4-rc1 was not released due to a blocking issue found prior to
- 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)
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)
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.4-rc2
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