Posts Tagged 1.4
Asterisk 1.4.35-Rc1 and Asterisk 1.6.2.11-Rc1 Now Available
Posted by admin in Release Candidates, asterisk on July 23, 2010
The Asterisk Development Team has announced release candidates of Asterisk for versions 1.4.35 and 1.6.2.11. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
These release candidates address issues that were reported by the community and resolved since the last round of bug fix releases.
The following is a sampling of issues resolved in these release candidates:
- Send DialPlanComplete as a response, not as a separate event.
(Closes issue #17504. Reported, patched by rrb3942) - Ensure channel placed into MeetMe in ringing state is properly hung up.
(Closes issue #15871. Reported, patched by Ivan) - Fix a problem with RFC 2833 DTMF not being accepted.
(Closes issue #17571. Reported by mdeneen. Tested by richardf, maxochoa, JJCinAZ) - Add option to not do a call forward on a 482 Loop Detected.
(Reviewboard: https://reviewboard.asterisk.org/r/764/ ) - Fix logging message for a stale nonce.
(Closes issue #17582. Reported, patched by kenner) - Fix some issues related to dynamic feature groups in features.conf
(Closes issue #17589. Reported,tested by lmadsen. Patched,tested by russell) - cdr_pgsql does not detect when a table is found. Add an ERROR message letting you know when a failure to get the columns from the database exists.
(Closes issue #17478. Reported, patched by kobaz) - Delete IMAP messages in reverse order to ensure reordering after each expunge does not cause deletion of the wrong message.
(Closes issue #16350. Reported by noahisaac. Patched by tilghman)
For a full list of changes in the current release candidates, please see the ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.35-rc1
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.11-rc1
Issues found in any of these release candidates should be reported to the Asterisk issue tracker at https://issues.asterisk.org
Thank you for your continued support of Asterisk!
Asterisk 1.4.34 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.4.34.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.34 resolves several issues reported by the community and would have not been possible without your participation.
Thank you!
The following are a few of the issues resolved by community developers:
- Allow users to specify a port for DUNDi peers.
(Closes issue #17056. Reported, patched by klaus3000) - Decrease the module ref count in sip_hangup when SIP_DEFER_BYE_ON_TRANSFER is set.
(Closes issue #16815. Reported, patched by rain) - First caller into a dynamic conference new enters the pin once.
(Closes issue #15878. Reported, patched by pabelanger) - Send AgentComplete manager events in the event of blind and attended transfers.
(Closes issue #16819. Reported, patched by elbriga)
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.34
Thank you for your continued support of Asterisk!
Asterisk 1.4.34-Rc2 and Asterisk 1.6.2.10-Rc2 Now Available
Posted by admin in Release Candidates, asterisk on July 14, 2010
The Asterisk Development Team has announced the 2nd set of release candidates of Asterisk for versions 1.4.34 and 1.6.2.10. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
These release candidates address issues that were reported by the community and resolved since the last round of bug fix releases.
The following is are issues resolved in these release candidates:
- Fix problem with RFC 2833 DTMF not being accepted.A recent check was added to ensure that we did not erroneously detect duplicate DTMF when we received packets out of order.
The problem was that the check did not account for the fact that the seqno of an RTP stream will roll over back to 0 after hitting 65535. Now, we have a secondary check that will ensure that the
seqno rolling over will not cause us to stop accepting DTMF.
(Closes issue #17571) - Clear the AST_CDR_FLAG_DIALED flag for channels going into the pbx via the G option in app_dial
(Closes issue #17592)
For a full list of changes in the current release candidates, please see the ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.34-rc2
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.10-rc2
Issues found in any of these release candidates should be reported to the Asterisk issue tracker at https://issues.asterisk.org
Thank you for your continued support of Asterisk!
Asterisk 1.4.34-Rc1 And Asterisk 1.6.2.10-Rc1 Now Available
Posted by admin in Release Candidates, asterisk on June 29, 2010
The Asterisk Development Team has announced release candidates of Asterisk for versions 1.4.34 and 1.6.2.10. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
These release candidates address issues that were reported by the community and resolved since the last round of bug fix releases.
The following is a sampling of issues resolved in these release candidates:
- If MeetMe is configured to use dynamic conference numbers, then the first caller (which creates the conference) had to enter the PIN number twice.
(Closes issue #15878. Reported by shawkris. Patched by pabelanger) - Send AgentComplete manager events in the event of blind and attended transfers.
(Closes issue #16819. Reported and patched by elbriga) - Correct manager variable ‘EventList’ case.
(Closes issue #17520. Reported and patched by kobaz) - If there is realtime configuration, it does not get re-read on reload unless the config file also changes.
(Closes issue #16982. Reported and patched by dmitri. Tested by atis)
For a full list of changes in the current release candidates, please see the ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.34-rc1
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.10-rc1
Issues found in any of these release candidates should be reported to the Asterisk issue tracker at https://issues.asterisk.org
Thank you for your continued support of Asterisk!
Asterisk 1.4.33.1 Released
The Asterisk Development Team has announced the release of Asterisk 1.4.33.1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.33.1 resolves a regression involving the use of FXO signaling in chan_dahdi where a channel could continue ringing after it has been answered.
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.33.1
Thank you for your continued support of Asterisk!
Asterisk 1.4.33 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.4.33.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.33 resolves several issues reported by the community, and would have not been possible without your participation. Thank you!
The following are a few of the issues resolved by community developers:
- Remove arbitrary size limitation for hints
(Closes issue #17257. Reported, patched by tim_ringenbach) - Fix incorrectly typed indications for [nz] stutter and dialrecall
(Closes issue #17359. Reported, patched by alecdavis) - Make AgentComplete message more consistent
(Closes issue #15638. Reported, patched by elbriga) - Missing fallback to audio fax feature when T.38 re-INVITE failed
(Closes issue #16692. Reported, patched by vrban) - Don’t hang up on a queue caller if the file we attempt to play does not exist
(Closes issue #17061. Reported by RoadKill)
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.33
Thank you for your continued support of Asterisk!
Asterisk 1.4.33-rc2 and Asterisk 1.6.2.9-rc3 Now Available
Posted by admin in Release Candidates, asterisk on June 10, 2010
The Asterisk Development Team has announced new release candidates of Asterisk for versions 1.4.33 and 1.6.2.9. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
These releases resolve two issues reported by the community:
- Fix Debian init script to not use -c.When using the init script as-is currently, it could cause issues on Debian such as high CPU usage. This fix has worked for several people so I’m implementing the change. We now handle color displays properly.
(Closes issue #16784. Reported, tested by pabelanger. Patched by tilghman) - Ensure signals are not blocked inside other signal handlers.This eliminates the annoying on the console.
(Closes issue #17477. Reported by jvandal. Patched by tilghman)
For a full list of changes in the current release candidates, please see the ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.33-rc2
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.9-rc3
Issues found in these release candidates should be reported to the Asterisk issue tracker at https://issues.asterisk.org
Thank you for your continued support of Asterisk!
Asterisk 1.4.32 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.4.32. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.32 resolves several issues reported by the community, and would have not been possible without your participation.
Thank you!
The following are a few of the issues resolved by community developers:
- Make the mixmonitor thread process audio frames faster.
(Closes issue #17078. Reported, tested by: geoff2010. Patched by dhubbard) - When StopMonitor is called, ensure that it will not be restarted by a channel event.
(Closes issue #16590. Reported, patched by: kkm) - Fix up hidecallerid feature in chan_dahdi.
(Closes issue #17143, #7321. Reported, patched by djenson99) - Resolve deadlocks in chan_local.
(Closes issue #17185. Reported, tested by schmoozecom, GameGamer43) - Ensure channel state is not incorrectly set in the case of a very early answer.
(Closes issue #17067. Reported, patched by tzafrir) - Registration fix for SIP realtime.
(Closes issue #17266. Reported, patched by Nick_Lewis. Tested by sberney)
For a full list of changes in the current release, please see the
ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.32
Thank you for your continued support of Asterisk!
Asterisk 1.4.32-Rc1, Asterisk 1.6.0.28-Rc1, Asterisk 1.6.1.20-Rc1 and Asterisk 1.6.2.7-Rc1 Now Available
Posted by admin in Release Candidates, asterisk on May 6, 2010
The Asterisk Development Team has announced release candidates of Asterisk for versions 1.4.32, 1.6.0.28, 1.6.1.20, and 1.6.2.8. These release candidates are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
Note that support for the 1.6.0 and 1.6.1 branches are moving to security fixes only, scheduled for the first half of May 2010. The Asterisk development team recommends that all users of Asterisk 1.6.0 and 1.6.1 series move to the 1.6.2 series for continued bug fix support.
Asterisk releases for 1.6.0.28 and 1.6.1.20 are the last scheduled maintenance releases for Asterisk branches 1.6.0 and 1.6.1 and will move to security maintenance only after their full release.
More information about the changes to maintenance support can be found at:
http://www.asterisk.org/node/49924
Information about the Asterisk maintenance schedule is available at:
http://www.asterisk.org/asterisk-versions
These release candidates resolve many issues reported by the community since the last set of releases.
For a full list of changes in the current release candidates, please see the ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.32-rc1
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.0.28-rc1
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.1.20-rc1
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.8-rc1
Issues found in any of these release candidates should be reported to the Asterisk issue tracker at https://issues.asterisk.org
Thank you for your continued support of Asterisk!
Asterisk 1.4.29 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.4.29.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.29 resolved several issues reported by the community, and would have not been possible without your participation. Thank you!
- Fix to Monitor which previously assumed the file to write to did not contain pathing.
(Closes issue #16377, #16376. Reported by bcnit. Patched by dant. - Propertly set T.38 attributes and don’t return before T.38 ports are configured when T.38 is found but no audio stream is found.
(Closes issue #16318. Reported by bird_of_Luck. Tested by vrban, mihaill. Patched by vrban, mnicholson.) - Avoid crashes with large numbers of MeetMe conferences.
(Closes issue #16509. Reported by Kashif Raza. Tested, Patched by seanbright.) - Change in ‘sip show channels’ display format allowing more digits for CID.
(Closes issue #16459. Reported, Patched by Rzadzins. - Revise documentation on disposition values to the actual values used.
(Closes issue #16289. Reported by wdoekes.)
A summary of changes in this release can be found in the release summary:
http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.4.29-summary.txt
For a full list of changes in this releases, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.29
Thank you for your continued support of Asterisk!
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