Posts Tagged XMPP

Asterisk 10.0.0 Is Released

Asterisk The Open Source PBX & Telephony Platform

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team is proud to announce the release of Asterisk 10.0.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

Asterisk 10 is the next major release series of Asterisk. It will be a Standard support release, similar to Asterisk 1.6.2. For more information about support time lines for Asterisk releases, see the Asterisk versions page:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

With the release of the Asterisk 10 branch, the preceding ’1.’ has been removed from the version number per the blog post available at

http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-a…

The release of Asterisk 10 would not have been possible without the support and contributions of the community.

You can find an overview of the work involved with the 10.0.0 release in the summary:

http://svn.asterisk.org/svn/asterisk/tags/10.0.0/asterisk-10.0.0-summary…

A short list of available features includes:

  • T.38 gateway functionality has been added to res_fax.
  • Protocol independent out-of-call messaging support. Text messages not associated with an active call can now be routed through the Asterisk dialplan. SIP and XMPP are supported so far.
  • New highly optimized and customizable ConfBridge application capable of mixing audio at sample rates ranging from 8kHz-192kHz
  • Addition of video_mode option in confbridge.conf to provide basic video conferencing in the ConfBridge() dialplan application.
  • Support for defining hints has been added to pbx_lua.
  • Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB).
  • Much, much more!

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

http://svn.asterisk.org/svn/asterisk/branches/10/CHANGES

Also, when upgrading a system between major versions, it is imperative that you read and understand the contents of the UPGRADE.txt file, which is located at:

http://svn.asterisk.org/svn/asterisk/branches/10/UPGRADE.txt

Thank you for your continued support of Asterisk!

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Asterisk 10.0.0-beta2 Now Available

Asterisk The Open Source PBX & Telephony Platform

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team is pleased to announce the second beta release of Asterisk 10.0.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

With the release of the Asterisk 10 branch, the preceding ’1.’ has been removed from the version number per the blog post available at
http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-a…

All interested users of Asterisk are encouraged to participate in the Asterisk 10 testing process. Please report any issues found to the issue tracker, https://issues.asterisk.org/jira. It is also very useful to see successful test reports. Please post those to the asterisk-dev mailing list.

All Asterisk users are invited to participate in the #asterisk-testing channel on IRC to work together in testing the many parts of Asterisk.

Asterisk 10 is the next major release series of Asterisk. It will be a Standard support release, similar to Asterisk 1.6.2. For more information about support time lines for Asterisk releases, see the Asterisk
versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

A short list of features includes:

  • T.38 gateway functionality has been added to res_fax.
  • Protocol independent out-of-call messaging support. Text messages not associated with an active call can now be routed through the Asterisk dialplan. SIP and XMPP are supported so far.
  • New highly optimized and customizable ConfBridge application capable of mixing audio at sample rates ranging from 8kHz-192kHz
  • Addition of video_mode option in confbridge.conf to provide basic video conferencing in the ConfBridge() dialplan application.
  • Support for defining hints has been added to pbx_lua.
  • Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB).
  • Much, much more!

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

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

Thank you for your continued support of Asterisk!

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Asterisk 1.8.7.0 Now Available

Asterisk The Open Source PBX & Telephony Platform

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team announces the release of Asterisk 1.8.7.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.7.0 resolves several issues reported by the community and would have not been possible without your participation.
Thank you!

Please note that a significant numbers of changes and fixes have gone into features.c in this release (call parking, built-in transfers, call pickup, etc.).

NOTE:

Recently, we were notified that the mechanism included in our Asterisk source code releases to download and build support for the iLBC codec had stopped working correctly; a little investigation revealed that this occurred because of some changes on the ilbcfreeware.org website. These changes occurred as a result of Google’s acquisition of GIPS, who produced (and provided licenses for) the
iLBC codec.

If you are a user of Asterisk and iLBC together, and you’ve already executed a license agreement with GIPS, we believe you can continue using iLBC with Asterisk. If you are a user of Asterisk and iLBC together, but you had not executed a license agreement with GIPS, we encourage you to research the situation and consult with your own legal representatives to determine what actions you may want to take (or avoid taking).

More information is available on the Asterisk blog:

http://blogs.asterisk.org/2011/09/19/ilbc-support-in-asterisk-after-goog…

The following is a sample of the issues resolved in this release:

  • Added the ‘storesipcause’ option to sip.conf to allow the user to disable the setting of HASH(SIP_CAUSE,) on the channel. Having chan_sip set HASH(SIP_CAUSE,) on the channel carries a significant performance penalty because of the usage of the MASTER_CHANNEL() dialplan function.We’ve decided to disable this feature by default in future 1.8 versions. This would be an unexpected behavior change for anyone depending on that SIP_CAUSE update in their dialplan. Please refer to the asterisk-dev mailing list more information:http://lists.digium.com/pipermail/asterisk-dev/2011-August/050626.html
  • Significant fixes and improvements to parking lots.
    (Closes issues ASTERISK-17183, ASTERISK-17870, ASTERISK-17430, ASTERISK-17452, ASTERISK-17452, ASTERISK-15792. Reported by: David Cabrejos, Remi Quezada, Philippe Lindheimer, David Woolley, Mat Murdock. Patched by: rmudgett)
  • Numerous issues have been reported for deadlocks that are caused by a blocking read in res_timing_timerfd on a file descriptor that will never be written to. A change to Asterisk adds some checks to make sure that the timerfd is both valid and armed before calling read(). Should fix: ASTERISK-18142, ASTERISK-18197, ASTERISK-18166 and possibly others.
    (In essence, this change should make res_timing_timerfd usable.)
  • Resolve segfault when publishing device states via XMPP and not connected.
    (Closes issue ASTERISK-18078. Reported, patched by: Michael L. Young. Tested by Jonathan Rose)
  • Refresh peer address if DNS unavailable at peer creation.
    (Closes issue ASTERISK-18000)
  • Fix the missing DAHDI channels when using the newer chan_dahdi.conf sections for channel configuration.
    (Closes issue ASTERISK-18496. Reported by Sean Darcy. Patched by Richard Mudgett)
  • Remove unnecessary libpri dependency checks in the configure script.
    (Closes issue ASTERISK-18535. Reported by Michael Keuter. Patched by Richard Mudgett)
  • Update get_ilbc_source.sh script to work again.
    (Closes issue ASTERISK-18412)

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

Thank you for your continued support of Asterisk!

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Asterisk 1.8.2-rc1 Now Available

Asterisk The Open Source PBX & Telephony Platform

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team has announced the first release candidate of Asterisk 1.8.2. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.2-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:

  • ‘sip notify clear-mwi’ needs terminating CRLF.
    (Closes issue #18275. Reported, patched by klaus3000)
  • Patch for deadlock from ordering issue between channel/queue locks in app_queue (set_queue_variables).
    (Closes issue #18031. Reported by rain. Patched by bbryant)
  • Fix cache of device state changes for multiple servers.
    (Closes issue #18284, #18280. Reported, tested by klaus3000. Patched, tested by russellb)
  • Resolve issue where channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call.
    (Closes issue #18171. Reported by: SantaFox)
    (Closes issue #18185. Reported by: kwemheuer)
    (Closes issue #18211. Reported by: zahir_koradia)
    (Closes issue #18230. Reported by: vmarrone)
    (Closes issue #18299. Reported by: mbrevda)
    (Closes issue #18322. Reported by: nerbos)
  • Fix reloading of peer when a user is requested. Prevent peer reloading from causing multiple MWI subscriptions to be created when using realtime.
    (Closes issue #18342. Reported, patched by nivek.)
  • Fix XMPP PubSub-based distributed device state. Initialize pubsubflags to 0 so res_jabber doesn’t think there is already an XMPP connection sending device state. Also clean up CLI commands a bit.
    (Closes issue #18272. Reported by klaus3000. Patched by Marquis42)
  • Don’t crash after Set(CDR(userfield)=…) in ast_bridge_call. Instead of setting peer->cdr = NULL, set it to not post.
    (Closes issue #18415. Reported by macbrody. Patched, tested by jsolares)
  • Fixes issue with outbound google voice calls not working. Thanks to az1234 and nevermind_quack for their input in helping debug the issue.
    (Closes issue #18412. Reported by nevermind_quack. Patched by dvossel)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.2-rc1

Thank you for your continued support of Asterisk!

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Asterisk 1.8.0 Now Available!

Asterisk The Open Source PBX & Telephony Platform

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team is proud to announce the release of Asterisk 1.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

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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Asterisk 1.8.0-Beta2 Now Available

Asterisk The Open Source PBX & Telephony Platform

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta2.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any issues found to the issue tracker, http://issues.asterisk.org/. It is also very useful to see successful test reports. Please post those to the asterisk-dev mailing list.

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

This release contains fixes since the last beta release as reported by the community. Some of the changes include:

  • Remove duplicate -c flag when using $(INSTALL)
    (Closes issue #17695. Reported, patched by pabelanger)
  • Don’t re-register CDR module on reload.
    (Closes issue #17304. Reported, tested by jnemeth. Patched by tilghman)
  • Don’t assume qlog is open.
    (Closes issue #17704. Reported, tested by vrban. Patched by pabelanger)
  • Expand the correct value within AST_OPTION_ONLY.
    (Closes issue #17703. Reported by stuarth. Patched by seanbright)
  • Allow for systems without locale support to be usable.
    (Closes issue #17697. Reported, patched by pprindeville. Tested by mmichelson)
  • Fixes for sounds/Makefile to install on systems using older GNU make.
    (Closes issue #17716. Reported by farisraouf. Patched by tilghman, qwell, seanbright)
  • Update logger.conf.sample to include documentation about new ‘fax’ logger level.
    (Closes issue #17715. Reported, tested by vrban. Patched by pabelanger)

Asterisk 1.8 contains many new features over previous releases of Asterisk.
A short list of included features includes:

  • Secure RTP
  • IPv6 Support
  • 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, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-beta2

Thank you for your continued support of Asterisk!

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Asterisk 1.8.0-Beta1 Now Available

Asterisk The Open Source PBX & Telephony Platform

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta1.
This release marks the beginning of the testing process for the eventual release of Asterisk 1.8.0.

This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any issues found to the issue tracker, http://issues.asterisk.org/. It is also very useful to hear successful test reports. Please post those to the asterisk-dev mailing list.

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

Asterisk 1.8 contains many new features over previous releases of Asterisk.
A short list of included features includes:

  • Secure RTP
  • IPv6 Support
  • 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, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-beta1

Thank you for your continued support of Asterisk!

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