ASTERWEB Blog

16Nov/100

Asterisk 1.6.2.14 Released

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Il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 1.6.2.14.

The Asterisk Development Team has announced the release of Asterisk 1.6.2.14. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.6.2.14 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 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)

* 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)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.14

Thank you for your continued support of Asterisk!

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16Nov/100

Asterisk 1.4.37 Released

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Il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 1.4.37.

The release of Asterisk 1.4.37 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 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)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.37

Thank you for your continued support of Asterisk!

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4Nov/100

Elastix: 1.000.000 di donwloads

Elastix

Elastix


Nato nel 2006, Elastix, il progetto della PaoloSanto Solutions, ha avuto sempre più riscontro nel mercato delle distro basate su Asterisk e come annunciato dalla stessa comunità, ha raggiunto il milione di download.

Questo il link del progetto: ELASTIX

3Nov/100

Asterisk 1.8.0 Released

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Il giorno 26 ottobre, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 1.8.0.

Dal post originale:

Asterisk 1.8 is the next major release series of Asterisk. It will be a Long Term Support (LTS) release, similar to Asterisk 1.4. For more information about support time lines for Asterisk releases, see the Asterisk versions page.

http://www.asterisk.org/asterisk-versions

The release of Asterisk 1.8.0 would not have been possible without the support and contributions of the community. Since Asterisk 1.6.2, we've had over 500 reporters, more than 300 testers and greater than 200 developers contributed to
this release.

You can find a summary of the work involved with the 1.8.0 release in the sumary:

http://svn.asterisk.org/svn/asterisk/tags/1.8.0/asterisk-1.8.0-summary.txt

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=markup

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

Thank you for your continued support of Asterisk!

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