Archive for category codecs

Asterisk 10.2.0-rc2 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 release candidate of Asterisk 10.2.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 10.2.0-rc2 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 candidate:

  • Fix deadlocks occuring in chan_agent.Bad locking order was added to chan_agent to prevent a segfault that occurred due to a race condition. This patch addresses the bad locking order by locking
    the channel and its private data in the correct order. This stops the deadlock, which occurred when doing multiple interactions with agents. (closes issue ASTERISK-19285) Reported by: Alex Villacis Lasso
  • Include iLBC source code for distribution with AsteriskClarification regarding the iLBC source code was provided by Google, and the appropriate licenses have been included in the codecs/ilbc folder.
    (closes issue: ASTERISK-18943) Reporter: Leif Madsen
  • Fix crash from bridge channel hangup race condition in ConfBridgeThis patch addresses two issues in ConfBridge and the channel bridge layer:
    1. It fixes a race condition wherein the bridge channel could be hung up
    2. It removes the deadlock avoidance from the bridging layer and makes the bridge_pvt an ao2 ref counted object
    (issue ASTERISK-18988, ASTERISK-18885, ASTERISK-19100)
    Reported by: Dmitry Melekhov, Alexander Akimov
  • Don’t do a DNS lookup on an outbound REGISTER host if there is an outbound proxy configured.
    (closes issue ASTERISK-16550) reported by: Olle Johansson
  • Create and initialize udptl only when a dialog negotiates for image mediaPrior to this patch, the udptl struct was allocated and initialized when a dialog was associated with a peer that supported T.38, when a new SIP channel was allocated, or when an INVITE request was received. This resulted in any dialog associated with a peer that supported T.38 having udptl support assigned to it, including the UDP ports needed for communication. This patch creates and initializes the udptl structure only when the SDP for a dialog specifies that image media is supported, or when Asterisk indicates that a dialog needs to support T.38.
    (closes issue ASTERISK-16698, ASTERISK-16794)
    Reported by: under, Elazar; Tested by: Stefan Schmidt
  • Allow only one thread at time to do Asterisk cleanup/shutdownAdd locking around the really-really-quit part of the core stop/restart part. Previously more than one thread could be called to do cleanup, causing atexit handlers to be run multiple times, in turn causing segfaults. (issue ASTERISK-18883)
    Patch by: Walter Doekes
  • Fix outbound DTMF for inband mode in chan_ooh323This tells asterisk core to generate DTMF sounds. (Closes issue ASTERISK-19233) Reported by: Matt Behrens Patches:
    chan_ooh323.c.patch uploaded by Matt Behrens (License #6346)

And much more! For a full list of changes in the current release, please see the ChangeLog:

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

Thank you for your continued support of Asterisk!

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Asterisk 1.8.10.0-rc2 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 release candidate of Asterisk 1.8.10.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.10.0-rc2 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 candidate:

  • Fix deadlocks occuring in chan_agent.Bad locking order was added to chan_agent to prevent a segfault that occurred due to a race condition. This patch addresses the bad locking order by locking the channel and its private data in the correct order. This stops the deadlock, which occurred when doing multiple interactions with agents. (closes issue ASTERISK-19285) Reported by: Alex Villacis Lasso
  • Include iLBC source code for distribution with AsteriskClarification regarding the iLBC source code was provided by Google, and the appropriate licenses have been included in the codecs/ilbc folder.
    (closes issue: ASTERISK-18943) Reporter: Leif Madsen
  • Don’t do a DNS lookup on an outbound REGISTER host if there is an outbound proxy configured.
    (closes issue ASTERISK-16550) reported by: Olle Johansson
  • Create and initialize udptl only when a dialog negotiates for image mediaPrior to this patch, the udptl struct was allocated and initialized when a dialog was associated with a peer that supported T.38, when a new SIP channel was allocated, or when an INVITE request was received. This resulted in any dialog associated with a peer that supported T.38 having udptl support assigned to it, including the UDP ports needed for communication. This patch creates and initializes the udptl structure only when the SDP for a dialog specifies that image media is supported, or when Asterisk indicates that a dialog needs to support T.38.
    (closes issue ASTERISK-16698, ASTERISK-16794)
    Reported by: under, Elazar; Tested by: Stefan Schmidt
  • Allow only one thread at time to do Asterisk cleanup/shutdownAdd locking around the really-really-quit part of the core stop/restart part. Previously more than one thread could be called to do cleanup, causing atexit handlers to be run multiple times, in turn causing segfaults. (issue ASTERISK-18883)
    Patch by: Walter Doekes
  • Fix outbound DTMF for inband mode in chan_ooh323 This tells asterisk core to generate DTMF sounds. (Closes issue ASTERISK-19233) Reported by: Matt Behrens Patches: chan_ooh323.c.patch uploaded by Matt Behrens (License #6346)

And much more! For a full list of changes in the current release, please see the ChangeLog:

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

Thank you for your continued support of Asterisk!

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Asterisk 10.2.0-rc1 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 first release candidate of Asterisk 10.2.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 10.2.0-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 candidate:

  • Include iLBC source code for distribution with AsteriskClarification regarding the iLBC source code was provided by Google, and the appropriate licenses have been included in the codecs/ilbc folder.
    (closes issue: ASTERISK-18943) Reporter: Leif Madsen
  • Fix crash from bridge channel hangup race condition in ConfBridgeThis patch addresses two issues in ConfBridge and the channel bridge layer:
    1. It fixes a race condition wherein the bridge channel could be hung up
    2. It removes the deadlock avoidance from the bridging layer and makes the bridge_pvt an ao2 ref counted object
    (issue ASTERISK-18988, ASTERISK-18885, ASTERISK-19100)
    Reported by: Dmitry Melekhov, Alexander Akimov
  • Don’t do a DNS lookup on an outbound REGISTER host if there is an outbound proxy configured.
    (closes issue ASTERISK-16550) reported by: Olle Johansson
  • Create and initialize udptl only when a dialog negotiates for image mediaPrior to this patch, the udptl struct was allocated and initialized when a dialog was associated with a peer that supported T.38, when a new SIP channel was allocated, or when an INVITE request was received. This resulted in any dialog associated with a peer that supported T.38 having udptl support assigned to it, including the UDP ports needed for communication. This patch creates and initializes the udptl structure only when the SDP for a dialog specifies that image media is supported, or when Asterisk indicates that a dialog needs to support T.38.
    (closes issue ASTERISK-16698, ASTERISK-16794)
    Reported by: under, Elazar; Tested by: Stefan Schmidt
  • Allow only one thread at time to do Asterisk cleanup/shutdownAdd locking around the really-really-quit part of the core stop/restart part.
    Previously more than one thread could be called to do cleanup, causing atexit handlers to be run multiple times, in turn causing segfaults.
    (issue ASTERISK-18883)
    Patch by: Walter Doekes
  • Fix outbound DTMF for inband mode in chan_ooh323This tells asterisk core to generate DTMF sounds. (Closes issue ASTERISK-19233) Reported by: Matt Behrens Patches:
    chan_ooh323.c.patch uploaded by Matt Behrens (License #6346)

And much more! For a full list of changes in the current release, please see the ChangeLog:

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

Thank you for your continued support of Asterisk!

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Asterisk 1.8.10.0-rc1 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 first release candidate of Asterisk 1.8.10.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.10.0-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 candidate:

  • Include iLBC source code for distribution with AsteriskClarification regarding the iLBC source code was provided by Google, and the appropriate licenses have been included in the codecs/ilbc folder.
    (closes issue: ASTERISK-18943) Reporter: Leif Madsen
  • Don’t do a DNS lookup on an outbound REGISTER host if there is an outbound proxy configured.
    (closes issue ASTERISK-16550) reported by: Olle Johansson
  • Create and initialize udptl only when a dialog negotiates for image mediaPrior to this patch, the udptl struct was allocated and initialized when a dialog was associated with a peer that supported T.38, when a new SIP channel was allocated, or when an INVITE request was received. This resulted in any dialog associated with a peer that supported T.38 having udptl support assigned to it, including the UDP ports needed for communication. This patch creates and initializes the udptl structure only when the SDP for a dialog specifies that image media is supported, or when Asterisk indicates that a dialog needs to support T.38.
    (closes issue ASTERISK-16698, ASTERISK-16794)
    Reported by: under, Elazar; Tested by: Stefan Schmidt
  • Allow only one thread at time to do Asterisk cleanup/shutdownAdd locking around the really-really-quit part of the core stop/restart part. Previously more than one thread could be called to do cleanup, causing atexit handlers to be run multiple times, in turn causing segfaults.
    (issue ASTERISK-18883)
    Patch by: Walter Doekes
  • Fix outbound DTMF for inband mode in chan_ooh323This tells asterisk core to generate DTMF sounds. (Closes issue ASTERISK-19233) Reported by: Matt Behrens Patches: chan_ooh323.c.patch uploaded by Matt Behrens (License #6346)

And much more! For a full list of changes in the current release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/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 1.8.7.0-rc2 Now Available

Asterisk The Open Source PBX & Telephony Platform

Asterisk The Open Source PBX & Telephony Platform

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

The release of Asterisk 1.8.7.0-rc2 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.) and a testing focus on those particular areas would be appreciated.

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 candidate:

  • 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)

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

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

Thank you for your continued support of Asterisk!

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iLBC support in Asterisk after Google’s acquisition of GIPS

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 result of Google’s acquisition of GIPS, who produced (and provided licenses for) the iLBC codec.

We’ve determined that the change necessary to fix Asterisk’s iLBC build process is rather trivial, and so we’re planning to make that change in Asterisk 1.8.7.0-rc2, and subsequently in 1.8.7.0. We are not planning on making new releases of Asterisk 1.4 and Asterisk 1.6.2, since they are in security-maintenance mode and this is not a security issue. Users who wish to make the same change on their own to their copies of those versions are of course welcome to do so.

As part of the process of determining what had broken here, we also became aware that the ilbcfreeware.org website no longer offers the iLBC license agreement it used to offer; this agreement was required by the iLBC licensors (GIPS) in order for users to safely distribute and use iLBC (and this is why the Asterisk project does not include the iLBC source code directly with Asterisk). The removal of this license agreement also occurred as a result of the Google acquisition, but as of this moment no alternative has been made available for those who wish to use the iLBC source code published in RFC 3951 (which Asterisk uses).

Google does have an alternative implementation of iLBC available as part of the WebRTC project, with a license that is compatible with Asterisk (and does not require written agreements from end users), but the codec_ilbc module in Asterisk cannot be built against the WebRTC implementation of iLBC. Until such time as we have an improved version of codec_ilbc, Asterisk users will have to continue using the RFC 3951 iLBC source code.

Unfortunately, that leaves Asterisk users in a bit of a bind; if they had already signed and sent in the GIPS iLBC license agreement, we believe they can continue to safely use the existing iLBC implementation. New users, though, do not have the option of agreeing to a license agreement that would allow them to use the RFC 3951 iLBC source code, as there is no mechanism to do that currently available.

We’ve contacted Google and they are aware of the dilemma, and have said that they will address it, but we don’t have a timeframe for when an alternative license mechanism will be available.
In summary, 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).

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AsteriskNOW 1.7.1 now available with add-on installer module for FreePBX

Asterisk NOW

AsteriskNOW 1.7.1 has been released with a new module for FreePBX that allows installation of Digium add-on software from within the web-based interface. Now there’s no command-line work to be done to get Digium’s G729 codec, Fax for Asterisk, HPEC, or Skype for Asterisk. We’ve also made some changes to make AsteriskNOW even friendlier for newcomers to the Asterisk community.

Together with the DAHDI configuration module that began shipping with 1.7.0, these modules make Asterisk administration even easier.  Download AsteriskNOW today, and burn it to a CD or start it in a virtual machine. In minutes you can turn your computer into your next phone system!

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Asterisk 1.8.0-Beta3 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-beta3.
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. A sampling of the changes in this release include:

  • Fix a regression where HTTP would always be enabled regardless of setting.
    (Closes issue #17708. Reported, patched by pabelanger)
  • ACL errors displayed on screen when using dynamic_exclude_static in sip.conf
    (Closes issue #17717. Reported by Dennis DeDonatis. Patched by mmichelson)
  • Support “channels” in addition to “channel” in chan_dahdi.conf.
    (https://reviewboard.asterisk.org/r/804)
  • Fix parsing error in sip_sipredirect(). The code was written in a way that did a bad job of parsing the port out of a URI. Specifically, it would do badly when dealing with an IPv6 address.
    (Closes issue #17661. Reported by oej. Patched by mmichelson)
  • Fix inband DTMF detection on outgoing ISDN calls.
    (Patched by russellb and rmudgett)
  • Fixes issue with translator frame not getting freed. This issue prevented g729 licenses from being freed up.
    (Closes issue #17630. Reported by manvirr. Patched by dvossel)
  • Fixed IPv6-related SIP parsing bugs and updated documention.
    (Reported by oej. Patched by sperreault)
  • Add new, self-contained feature FIELDNUM(). Returns a 1-based index into a list of a specified item. Matches up with FIELDQTY() and CUT().
    (Closes #17713. Reported, patched by gareth. Tested by tilghman)

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

  • Secure RTP
  • IPv6 Support in the SIP Channel
  • 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-beta3

Thank you for your continued support of Asterisk!

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Fax For Asterisk 1.1.6 Release Announcement

T.38 fax for Asterisk

T.38 fax for Asterisk

Digium is pleased to announce the release of version 1.1.6 of its Fax For Asterisk product, a commercial grade FAX add-on module for open source Asterisk.

This release contains a number of significant improvements, including:

  • Support for 64-bit Linux installations.
  • Reduction in resource consumption, and increase in performance, of T.38 session handling.
  • Simplification of session handling during transition from G.711 to T.38 mode.
  • Adoption of the latest Asterisk T.38 negotiation API, ensuring interoperability with a wide range of T.38 endpoints.

Version 1.1.6 of Fax For Asterisk is available for immediate download at http://www.digium.com/en/docs/FAX/faa-download.php. Note that because this release uses the very latest T.38 negotiation mechanism in Asterisk, it is not compatible with all released versions of Asterisk. The Fax For Asterisk download selector lists the Asterisk versions supported by this release.

For more information about Fax For Asterisk, please visit the product page.

Thank you for your support!
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