Asterisk 1.8.0-rc2 Now Available
Il Team di Sviluppo di Asterisk ha annunciato il rilascio della seconda beta della 1.8.0.
Dal post originale:
Asterisk 1.8.0-rc1 was not released due to an issue found prior to release.
* Make AMI honor enabled=no
(Closes issue #18040. Reported by: twilson
Review: https://reviewboard.asterisk.org/r/938/)
All interested users of Asterisk are encouraged to participate in the 1.8
testing process. Please report any issues found to the issue tracker,
https://issues.asterisk.org/. It is also very useful to see successful test
reports. Please post those to the asterisk-dev mailing list.
This release candidate contains fixes since the last beta release as reported by
the community. A sampling of the changes in this release candidate include:
* Add slin16 support for format_wav (new wav16 file extension)
(Closes issue #15029. Reported, patched by andrew. Tested by Qwell)
* Fixes a bug in manager.c where the default configuration values weren't reset
when the manager configuration was reloaded.
(Closes issue #17917. Reported by lmadsen. Patched by bbryant)
* Various fixes for the calendar modules.
(Patched by Jan Kalab.
Reviewboard: https://reviewboard.asterisk.org/r/880/
Closes issue #17877. Review: https://reviewboard.asterisk.org/r/916/
Closes issue #17776. Review: https://reviewboard.asterisk.org/r/921/)
* Add CHANNEL(checkhangup) to check whether a channel is in the process of
being hung up.
(Closes issue #17652. Reported, patched by kobaz)
* Fix a bug with MeetMe where after announcing the amount of time left in a
conference, if music on hold was playing, it doesn't restart.
(Closes issue #17408, Reported, patched by sysreq)
* Fix interoperability problems with session timer behavior in Asterisk.
(Closes issue #17005. Reported by alexcarey. Patched by dvossel)
* Rate limit calls to fsync() to 1 per second after astdb updates. Astdb was
determined to be one of the most significant bottlenecks in SIP registration
processing. This patch improved the speed of an astdb load test by 50000%
(yes, Fifty-Thousand Percent). On this particular load test setup, this
doubled the number of SIP registrations the server could handle.
(Review: https://reviewboard.asterisk.org/r/825/)
* Don't clear the username from a realtime database when a registration
expires. Non-realtime chan_sip does not clear the username from memory when a
registration expiries so realtime probably shouldn't either.
(Closes issue #17551. Reported, patched by: ricardolandim. Patched by
mnicholson)
* Don't hang up a call on an SRTP unprotect failure. Also make it more obvious
when there is an issue en/decrypting.
(Closes issue #17563. Reported by Alexcr. Patched by sfritsch. Tested by
twilson)
* Many more issues. This is a significant upgrade over Asterisk 1.8.0 beta 5!
A short list of available features includes:
* Secure RTP
* IPv6 Support in the SIP channel driver
* Connected Party Identification Support
* Calendaring Integration
* A new call logging system, Channel Event Logging (CEL)
* Distributed Device State using Jabber/XMPP PubSub
* Call Completion Supplementary Services support
* Advice of Charge support
* Much, much more!
A full list of new features can be found in the CHANGES file.
http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=checkout
For a full list of changes in the current release candidate, please see the
ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc2
Asterisk 1.6.2.14-rc1 Now Available
Il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 1.6.2.14-rc1
Dal post originale:
The release of Asterisk 1.6.2.14-rc1 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:
* Fix issue where session timers would be advertised as supported even when
session-timers=refuse was set in sip.conf. Also fix interoperability
problems with session timer behavior in Asterisk.
(Closes issue #17005. Reported by alexcarey. Patched by dvossel)
* Fix issue with decoding ^-escaped characters in realtime (res_pgsql).
(Closes issue #17790. Reported by denzs. Patched by Qwell)
* Parse all "Accept" headers for SIP SUBSCRIBE requests.
(Closes issue #17758. Reported by ibc. Patched by dvossel)
* Fix issue where queue stats would be reset on reload.
(Closes issue #17535. Reported by raarts. Patched by tilghman)
* Fix issue where MoH files were no longer rescanned on during a reload.
(Closes issue #16744. Reported by pj. Patched by Qwell)
* Fix issue with dialplan pattern matching where the specificity for pattern
ranges and pattern characters was inconsistent.
(Closes issue #16903. Reported, patched by Nick_Lewis)
Questo il ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.14-rc1
Asterisk 1.4.37-rc1 Now Available
Il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 1.4.37-rc1
Dal post originale:
The release of Asterisk 1.4.37-rc1 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:
* Fix issue with decoding ^-escaped characters in realtime (res_pgsql)
(Closes issue #17790. Reported denzs. Patched by Qwell)
* Don't send a devstate change on poke_noanswer if the state did not change.
(Closes issue #17741. Reported, patched by schmidts)
* Transmit silence when reading DTMF in ast_readstring. Otherwise you could get
issues with DTMF timeouts causing hangups.
(Closes issue #17370. Reported, patched by makoto)
* Fix to SIP extension state update (deadlock issues)
(Closes issue #17888. Reported by zerohalo. Patched by dvossel)
* Fix issue with MoH where it doesn't recover cleanly when it can't play a file
and would just stop, instead of continuing to find the next playable file in
the MoH class.
(Closes issue #17807. Reported by kshumard. Patched by bbryant)
Questo il ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.37-rc1
Asterisk 1.6.2.13 Now Available (1.6.2.12 Re-Release)
Il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 1.6.2.13
Dal post originale:
This release resolves an issue where the .version and ChangeLog files were not
updated for 1.6.2.12. Asterisk 1.6.2.13 has no additional changes from 1.6.2.12
other than the .version, ChangeLog and summary files.
Questo il hangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.13
Asterisk 1.6.2.12 Now Available
Il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 1.6.2.12
Dal post originale:
The release of Asterisk 1.6.2.12 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:
* Fix issue where DNID does not get cleared on a new call when using
immediate=yes with ISDN signaling.
(Closes issue #17568. Reported by wuwu. Patched by rmudgett)
* Several updates to res_config_ldap.
(Closes issue #13573. Reported by navkumar. Patched by navkumar, bencer.
Tested by suretec)
* Prevent loss of Caller ID information set on local channel after masquerade.
(Closes issue #17138. Reported by kobaz, patched by jpeeler)
* Fix SIP peers memory leak.
(Closes issue #17774. Reported, patched by kkm)
* Add Danish support to say.conf.sample
(Closes issue #17836. Reported, patched by RoadKill)
* Ensure SSRC is changed when media source is changed to resolve audio delay.
(Closes issue #17404. Reported, tested by sdolloff. Patched by jpeeler)
* Only do magic pickup when notifycid is enabled.
A new way of doing BLF pickup was introduced into 1.6.2. This feature adds a
call-id value into the XML of a SIP_NOTIFY message sent to alert a subscriber
that a device is ringing. This option should only be enabled when the new
'notifycid' option is set, but this was not the case. Instead the call-id
value was included for every RINGING Notify message, which caused a
regression for people who used other methods for call pickup.
(Closes issue #17633. Reported, patched by urosh. Patched by dvossel.
Tested by: dvossel, urosh, okrief, alecdavis)
Questo il ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.12
Asterisk 1.4.36 Now Available
Il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 1.4.36
Dal post originale:
The release of Asterisk 1.4.36 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:
* Fix issue where DNID does not get cleared on a new call when using
immediate=yes with ISDN signaling.
(Closes issue #17568. Reported by wuwu. Patched by rmudgett)
* Fix issue where SIP promiscuous redirect could fail to dial the
redirect (app_queue).
* Fixes issue with translator frame not getting freed. This issue prevented
G.729 licenses from being freed up.
(Closes issue #17630. Reported by manvirr. Patched by dvossel)
* Ensure SSRC is changed when media source is changed to resolve audio delay.
(Closes issue #17404. Reported, tested by sdolloff. Patched by jpeeler)
* Q931 - Sending PROGRESS after sending ALERTING is a protocol error.
(Closes issue #17874. Reported, patched by nic_bellamy)
Questo il ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.36