Showing posts with label Asterisk 1.6.. Show all posts
Showing posts with label Asterisk 1.6.. Show all posts

Friday, July 1, 2011

Final Maintenance Release Of Asterisk 1.6.2.19 Now Available.

Asterisk 1.6.2.19 http://snapvoip.blogspot.com/
Asterisk developer team on 29th June announced the release of final maintenance release of Asterisk 1.6.2.9, taking care of some security issues and fixes. The release is available for download at Asterisk site.
The Asterisk 1.6.2.19 is the final maintenance release from the1.6.2 branch. Support for security related issues will continue until April 21,2012. More information on support for various Asterisk branches could be found here.

The following is a sample of the issues resolved in this release:
  • Don't broadcast FullyBooted to every AMI connection The FullyBooted event should not be sent to every AMI connection every time someone connects via AMI. It should only be sent to the user who just connected. (Closes issue #18168. Reported, patched by FeyFre)
  • Fix thread blocking issue in the sip TCP/TLS implementation. (Closes issue #18497. Reported by vois. Tested by vois, rossbeer, kowalma, Freddi_Fonet. Patched by dvossel)
  • Don't delay DTMF in core bridge while listening for DTMF features. (Closes issue #15642, #16625. Reported by jasonshugart, sharvanek. Tested by globalnetinc, jde. Patched by oej, twilson)
  • Fix chan_local crashs in local_fixup() Thanks OEJ for tracking down the issue and submitting the patch. (Closes issue #19053. Reported, patched by oej)
  • Don't offer video to directmedia callee unless caller offered it as well (Closes issue #19195. Reported, patched by one47)

Additionally security announcements AST-2011-008, AST-2011-010, and AST-2011-011 have been resolved in this release.

For a full list of changes in this release, please see the ChangeLog:


Friday, May 21, 2010

The Asterisk Development Team Releases Asterisk 1.6.0.28 and 1.6.1.20

 Asterisk 1.6.0.28 and Asterisk 1.6.1.20 http://snapvoip.blogspot.com/

The Asterisk Development Team has announced the final maintenance releases of Asterisk branches 1.6.0 and 1.6.1 as versions 1.6.0.28 and 1.6.1.20. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The Asterisk releases for 1.6.0.28 and 1.6.1.20 are the last maintenance releases for Asterisk branches 1.6.0 and 1.6.1 and have now moved to security maintenance only.

The releases of Asterisk 1.6.0.28 and 1.6.1.20 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:

    * Fix issue where MixMonitor() recordings would be shorter than total duration.
      (Closes issue #17078. Reported,tested by geoff2010. Patched by dhubbard)
    * When StopMonitor() is called, ensure it will not be restarted by a channel event.
      (Closes issue #16590. Reported, patched by kkm)
    * Allow hidecalleridname feature to work.
      (Closes issue #17143. Reported, patched by djensen99)
    * 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 by chan_dahdi.
      (Closes issue #17067. Reported, patched by tzafrir)
    * Registration fix for SIP realtime. Make sure realtime fields are not empty.
      (Closes issue #17266. Reported, patched by Nick_Lewis. Tested by sberney)

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

For a full list of changes in the current release candidates, please see the
ChangeLogs:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.0.28
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.1.20

Wednesday, October 7, 2009

Distributed Call Centers With Asterisk 1.6, (Astricon 2009)

Leif Madsen At Astricon 2009 snapvoip.blogspot.com
I have seen and listened to Leif Madsen at Astricon many a times and have come away with brand new ideas. This year is not going to be different at Astricon 2009, as he will be presenting at Carrier/Call Center track on October 14th morning, 10:00 to 10:35.
He has been building a distributed call center based on Asterisk 1.4 but says that Asterisk 1.6 will help to things differently, easier and better, with new tools and features.
He is promising to be concentrated on technical aspects of the task, given the short time he has and I think it will be a great way for me to start the day.
You can read more about the talk at "Building a Distributed Call Center" at Astricon site and his blog post on the same subject at "AstriCon 2009, Building a Distributed Call Center"
Via Asterisk News

Friday, July 17, 2009

Elastix 2.0, An In Depth Look

Preview of Elastix 2.0
I was pretty excited about Elastix 2.0, the beta I am trying and working on, I went and voted for Elastix at SF Community Choice Awards. I really like what they do with the project. Even though I tried to write about Elastix 2.0,there is a very nice writeup on the Elastix site.
It explains how; Elastix 2.0 and its major components like;
  • Asterisk 1.6
  • Elastix Operator Panel release
  • Meeting / Presentation sharing
  • Advanced reports
  • Improvements to the Call Center Module
  • Better integration between all components
play major roles in the upcoming release. Enjoy.
A Preview Of Elastix 2.0

Friday, March 27, 2009

Trixbox 2.8.Beta Released

Trixbox 2.8
Fonality has released a beta of upcoming Trixbox 2.8. The blog post mentions that Asterisk 1.6 will be a part of the new release and the poster offers everything but flowers to Digium and the Asterisk team for the reliability and capability of Asterisk 1.6.
The current 2.8 beta is basically Trixbox 2.6 with Asterisk 1.6 and currently carries no other new features. But they will be added in the future.
If you are an Trixbox fan, you can help out the new product by testing the beta. It is available for download at sourceforge, from old Asterisk@home project.
Blog post on Fonality.