Rilasciato Asterisk 1.8.4.1
Il giorno 24 maggio, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Asterisk 1.8.4.1
Dal post originale:
The release of Asterisk 1.8.4.1 resolves several issues reported by the
community. Without your help this release would not have been possible.
Thank you!
Below is a list of issues resolved in this release:
Fix our compliance with RFC 3261 section 18.2.2. (aka Cisco phone fix)
(Closes issue #18951. Reported by jmls. Patched by wdoekes)
Resolve a change in IPv6 header parsing due to the Cisco phone fix issue.
This issue was found and reported by the Asterisk test suite.
(Closes issue #18951. Patched by mnicholson)
Resolve potential crash when using SIP TLS support.
(Closes issue #19192. Reported by stknob. Patched by Chainsaw. Tested by
vois, Chainsaw)
Improve reliability when using SIP TLS.
(Closes issue #19182. Reported by st. Patched by mnicholson)
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.4.1
Rilasciato Asterisk 1.8.4-rc3
Il giorno 26 aprile, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Asterisk 1.8.4-rc3
Dal post originale:
The release of Asterisk 1.8.4-rc3 resolves a couple of issues since the last
release candidate, including two security related issues (AST-2011-005 and
AST-2011-006).
Use SSLv23_client_method instead of old SSLv2 only.
(Closes issue #19095, #19138. Reported, patched by tzafrir. Tested by russell
and chazzam.
Resolve crash in ast_mutex_init()
(Patched by twilson)
Includes changes per AST-2011-005 and AST-2011-006
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-rc3
Information about the security releases are available at:
http://downloads.asterisk.org/pub/security/AST-2011-005.pdf
http://downloads.asterisk.org/pub/security/AST-2011-006.pdf
Rilasciato Asterisk 1.6.2.18
Il giorno 26 aprile, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Asterisk 1.6.2.18
Dal post originale:
The release of Asterisk 1.6.2.18 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:
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.
Resolve deadlocks related to device states in chan_sip
(Closes issue #18310. Reported, patched by one47. Patched by jpeeler)
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/)
In addition to the changes listed above, commits to resolve security issues
AST-2011-005 and AST-2011-006 have been merged into this release. More
information about AST-2011-005 and AST-2011-006 can be found at:
http://downloads.asterisk.org/pub/security/AST-2011-005.pdf
http://downloads.asterisk.org/pub/security/AST-2011-006.pdf
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.18
Rilasciato Asterisk 1.4.41
Il giorno 26 aprile, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Asterisk 1.4.41
Dal post originale:
The release of Asterisk 1.4.41 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:
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/)
In addition to the changes listed above, commits to resolve security issues
AST-2011-005 and AST-2011-006 have been merged into this release. More
information about AST-2011-005 and AST-2011-006 can be found at:
http://downloads.asterisk.org/pub/security/AST-2011-005.pdf
http://downloads.asterisk.org/pub/security/AST-2011-006.pdf
After the initial release of AST-2011-006, a regression was found and then
resolved. This release contains the correct change.
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.41