ASTERWEB Blog

Rilasciato Asterisk 12.0.0-alpha1

Il giorno 31 agosto, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Asterisk 12.0.0-alpha1.

Dal post originale:
The Asterisk Development Team is pleased to announce the first alpha release of
Asterisk 12.0.0. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases

All interested users of Asterisk are encouraged to participate in the
Asterisk 12 testing process. Please report any issues found to the issue
tracker, https://issues.asterisk.org/jira. All Asterisk users are invited to
participate in the #asterisk-bugs channel to help communicate issues found to
the Asterisk developers. It is also very useful to see successful test reports.
Please post those to the asterisk-dev mailing list (http://lists.digium.com).

The first preliminary test release of Asterisk 12 is an alpha release, not a
beta release. Due to the size and scope of the changes in Asterisk 12, both an
alpha test cycle and a beta test cycle will be performed. While users are
encouraged to participate in both test cycles, users who choose to participate
in the alpha release testing should understand that an alpha release has not
undergone all of the community testing that a beta release goes through.

Asterisk 12 is the next major release series of Asterisk. It will be a Standard
release, similar to Asterisk 10. For more information about
support time lines for Asterisk releases, see the Asterisk versions page:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

For important information regarding upgrading to Asterisk 12, please see the
Asterisk wiki:

https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+12

A short list of some of the new major features includes:

* A new SIP channel driver and accompanying SIP stack named chan_pjsip has been
added. This new channel driver is based on the PJSIP SIP stack by Teluu. It
includes support for the vast majority of features currently in chan_sip,
as well as numerous architectural improvements that alleviate pain points
present in the legacy SIP channel driver. Users who wish to use the new SIP
channel driver are encouraged to read the instructions on installing and
configuring PJSIP for Asterisk on the Asterisk wiki at
https://wiki.asterisk.org/wiki/x/J4GLAQ. Detailed instructions on configuring
the new SIP stack in Asterisk can be found on the Asterisk wiki as well, at
https://wiki.asterisk.org/wiki/x/hYCLAQ. Test reports of successful use of
chan_pjsip, with endpoint details, in addition to bug reports, are most
welcome.

* The Asterisk RESTful Interface (ARI) has been added. This interface lets
external systems harness the telephony primitives within Asterisk to develop
their own communications applications. Communication with Asterisk is done
through a REST interface, while asynchronous events from Asterisk are
encoded in JSON and sent via a WebSocket. More information on ARI can be found
at https://wiki.asterisk.org/wiki/x/lYBbAQ

* Major standardization of the Asterisk Manager Interface and its events have
occurred within this version. In particular, the names of Asterisk channels
no longer change and are stable throughout the lifetime of the channel.
More information on the changes in AMI can be seen in the AMI 1.4
Specification at https://wiki.asterisk.org/wiki/x/dAFRAQ

* All bridging within Asterisk is now performed using the Asterisk Bridging API,
which previously was only used by the ConfBridge application. This affords
Asterisk users greater stability, and has resulted in the abstraction of
channel masquerades, renaming, and other internal implementation details. It
also allows for the seamless transition between two-party and multi-party
bridges using core features.

And much more!

More information about the new features can be found on the Asterisk wiki:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Documentation

A full list of all new features can also be found in the CHANGES file.

http://svnview.digium.com/svn/asterisk/branches/12/CHANGES

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

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.0.0-alpha1

Rilasciato Asterisk 1.8.23.0

Il giorno 15 luglio, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Asterisk 1.8.23.0.

Dal post originale:
The release of Asterisk 1.8.23.0 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 a memory copying bug in slinfactory which was causing
mixmonitor issues.
(Closes issue ASTERISK-21799. Reported by Michael Walton)

* --- IAX2: fix race condition with nativebridge transfers.
(Closes issue ASTERISK-21409. Reported by alecdavis)

* --- Fix crash in chan_sip when a core initiated op occurs at the
same time as a BYE
(Closes issue ASTERISK-20225. Reported by Jeff Hoppe)

* --- Fix The Payload Being Set On CN Packets And Do Not Set Marker
Bit
(Closes issue ASTERISK-21246. Reported by Peter Katzmann)

* --- chan_sip: Session-Expires: Set timer to correctly expire at
(~2/3) of the interval when not the refresher
(Closes issue ASTERISK-21742. Reported by alecdavis)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.23.0

Rilasciato Asterisk 11.5.0

Il giorno 15 luglio, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Asterisk 11.5.0.

Dal post originale:
The release of Asterisk 11.5.0 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 Segfault In app_queue When "persistentmembers" Is Enabled
And Using Realtime
(Closes issue ASTERISK-21738. Reported by JoshE)

* --- IAX2: fix race condition with nativebridge transfers.
(Closes issue ASTERISK-21409. Reported by alecdavis)

* --- Fix The Payload Being Set On CN Packets And Do Not Set Marker
Bit
(Closes issue ASTERISK-21246. Reported by Peter Katzmann)

* --- Fix One-Way Audio With auto_* NAT Settings When SIP Calls
Initiated By PBX
(Closes issue ASTERISK-21374. Reported by Michael L. Young)

* --- chan_sip: NOTIFYs for BLF start queuing up and fail to be sent
out after retries fail
(Closes issue ASTERISK-21677. Reported by Dan Martens)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.5.0

PDF relativo al webinar “Zimbra: il nuovo business!”

In allegato al presente post il pdf relativo al webinar effettuato ieri 07/06/2013 dal titolo "webinar-zimbra-nuovo-business".

Troverete spunti per meglio comprendere ler potenzialità commerciali del prodotto.

Il nostro staff tecnico e commerciale è a disposizione per qualsiasi chiarimento o informazione aggiuntiva.

SCARICA IL PDF

Logo Asterweb

Telefono: 06.92946573

E-mail: commerciale@promente.it

Rilasciato DAHDI-Linux and DAHDI-Tools 2.7.0

Il giorno 29 maggio, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione DAHDI-Linux e DAHDI-Tools 2.7.0-rc1.

Dal post originale:
This release is available for immediate download at:
http://downloads.asterisk.org/pub/telephony/dahdi-linux
http://downloads.asterisk.org/pub/telephony/dahdi-tools
http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete

Rilasciato DAHDI-Linux e DAHDI-Tools 2.7.0-rc1

Il giorno 29 maggio, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione DAHDI-Linux e DAHDI-Tools 2.7.0-rc1.

Dal post originale:
http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=shortlog;h=refs/tags/v2.7.0-rc1

http://git.asterisk.org/gitweb/?p=dahdi/tools.git;a=shortlog;h=refs/tags/v2.7.0-rc1

Rilasciato Asterisk 11.4.0

Il giorno 17 maggio, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Asterisk 11.4.0.

Dal post originale:
The release of Asterisk 11.4.0 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 Sorting Order For Parking Lots Stored In Static Realtime
(Closes issue ASTERISK-21035. Reported by Alex Epshteyn)

* --- Fix StopMixMonitor Hanging Up When Unable To Stop MixMonitor On
A Channel
(Closes issue ASTERISK-21294. Reported by daroz)

* --- When a session timer expires during a T.38 call, re-invite with
correct SDP
(Closes issue ASTERISK-21232. Reported by Nitesh Bansal)

* --- Fix white noise on SRTP decryption
(Closes issue ASTERISK-21323. Reported by andrea)

* --- Fix reload skinny with active devices.
(Closes issue ASTERISK-16610. Reported by wedhorn)

For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.4.0

Rilasciato Asterisk 1.8.22.0

Il giorno 17 maggio, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Asterisk 1.8.22.0.

Dal post originale:
The release of Asterisk 1.8.22.0 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 Sorting Order For Parking Lots Stored In Static Realtime
(Closes issue ASTERISK-21035. Reported by Alex Epshteyn)

* --- Make ParkAndAnnounce return to priority + 1 when return context
is not defined
(Closes issue ASTERISK-20113. Reported by serginuez)

* --- When a session timer expires during a T.38 call, re-invite with
correct SDP
(Closes issue ASTERISK-21232. Reported by Nitesh Bansal)

* --- Fix several unreleased mutex locks that cause problem with
processing calls
(Closes issue ASTERISK-21119. Reported by Daniel Bohling)

* --- Fix crash when AMI redirect action redirects two channels out of
a bridge.
(Closes issue ASTERISK-21356. Reported by William luke)

For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.22.0

Rilasciato Asterisk 1.8.19.0

Il giorno 11 dicembre, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Asterisk 1.8.19.0.

Dal post originale:
The release of Asterisk 1.8.19.0 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:

* --- Prevent resetting of NATted realtime peer address on reload.
(Closes issue ASTERISK-18203. Reported by daren ferreira)

* --- Do not use a FILE handle when doing SIP TCP reads.
(Closes issue ASTERISK-20212. Reported by Phil Ciccone)

* --- Fix execution of 'i' extension due to uninitialized variable.
(Closes issue ASTERISK-20455. Reported by Richard Miller)

* --- Ensure that the Queue application tracks busy members in off nominal situations
(Closes issue ASTERISK-20623. Reported by Bryan Walters)

* --- Properly extract the Body information of an EWS calendar item
(Closes issue ASTERISK-19738. Reported by Dmitry Burilov)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.19.0

Rilasciato Asterisk 11.1.0

Il giorno 11 dicembre, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Asterisk 11.1.0.

Dal post originale:
The release of Asterisk 11.1.0 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 execution of 'i' extension due to uninitialized variable.
(Closes issue ASTERISK-20455. Reported by Richard Miller)

* --- Prevent resetting of NATted realtime peer address on reload.
(Closes issue ASTERISK-18203. Reported by daren ferreira)

* --- Fix ConfBridge crash if no timing module loaded.
(Closes issue ASTERISK-19448. Reported by feyfre)

* --- Fix the Park 'r' option when a channel parks itself.
(Closes issue ASTERISK-19382. Reported by James Stocks)

* --- Fix an issue where outgoing calls would fail to establish audio due to ICE negotiation failures.
(Closes issue ASTERISK-20554. Reported by mmichelson)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.1.0

Rilasciato Asterisk 10.11.0

Il giorno 11 dicembre, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Asterisk 10.11.0.

Dal post originale:
The release of Asterisk 10.11.0 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:

* --- Prevent resetting of NATted realtime peer address on reload.
(Closes issue ASTERISK-18203. Reported by daren ferreira)

* --- Do not use a FILE handle when doing SIP TCP reads.
(Closes issue ASTERISK-20212. Reported by Phil Ciccone)

* --- Fix ConfBridge crash if no timing module loaded.
(Closes issue ASTERISK-19448. Reported by feyfre)

* --- confbridge: Fix a bug which made conferences not record with AMI/CLI commands
(Closes issue ASTERISK-20601. Reported by Vilius)

* --- Fix execution of 'i' extension due to uninitialized variable.
(Closes issue ASTERISK-20455. Reported by Richard Miller)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.11.0

Rilasciato Asterisk 11

Il giorno 31 ottobre, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione stabile Asterisk.
La versione 11 di Asterisk è una LTS (Long Term Support) come la 1.8.

Dal post originale:
Asterisk 11 includes a number of major new features including:

WebRTC Support with WebSocket transport over SIP.
DTLS-SRTP – A secure transport for RTP media streams used by WebRTC and SIP endpoints.
ICE, STUN and TURN – A set of related technologies for establishing live media streams between software agents running behind network address translators (NATs) and firewalls. ICE, STUN and TURN have been incorporated into the Asterisk RTP engine as part of the effort to support WebRTC.
Motif – A new channel driver for supporting the Jingle protocol and Google Talk. Motif combines functions previously spread across multiple channels, and makes use of a new and more standards-compliant XMPP implementation.
More information about the new features can be found on the Asterisk 11 Documentation wiki page and a full list of all new features can be found in the CHANGES file.