Latest News About Asterisk » asterisk

Asterisk 10.1.3 Now Available

admin — Thu, 23 Feb 2012 14:25:55 +0000

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team has announced the release of Asterisk 10.1.3.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 10.1.3 resolves several issues reported by the community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

— Fix ACK routing for non-2xx responses.
(Closes issue ASTERISK-19389. Reported by: Karsten Wemheuer)
— Fix regressions with regards to route-set creation on early dialogs —
(Closes issue ASTERISK-19358. Reported-by: Karsten Wemheuer)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.1.3

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Asterisk 1.8.9.3 Now Available

admin — Thu, 23 Feb 2012 14:22:56 +0000

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team has announced the release of Asterisk 1.8.9.3.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.9.3 resolves several issues reported by the community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

— Fix ACK routing for non-2xx responses.
(Closes issue ASTERISK-19389. Reported by: Karsten Wemheuer)
— Fix regressions with regards to route-set creation on early dialogs —
(Closes issue ASTERISK-19358. Reported-by: Karsten Wemheuer)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.9.3

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Asterisk 10.2.0-rc2 Now Available

admin — Fri, 10 Feb 2012 11:52:29 +0000

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!

Share and Enjoy:

Asterisk 1.8.10.0-rc2 Now Available

admin — Fri, 10 Feb 2012 11:51:11 +0000

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!

Share and Enjoy:

Asterisk 10.1.2 Now Available

admin — Thu, 09 Feb 2012 11:38:20 +0000

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team has announced the release of Asterisk 10.1.2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 10.1.2 resolves several issues reported by the community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

— Fix SIP INFO DTMF handling for non-numeric codes —
(Closes issue ASTERISK-19290. Reported by: Ira Emus)
— Fix crash in ParkAndAnnounce —
(Closes issue ASTERISK-19311. Reported-by: tootai)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.1.2

Thank you for your continued support of Asterisk!

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Asterisk 1.8.9.2 Now Available

admin — Thu, 09 Feb 2012 11:37:50 +0000

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team has announced the release of Asterisk 1.8.9.2.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.9.2 resolves several issues reported by the community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

— Fix SIP INFO DTMF handling for non-numeric codes —
(Closes issue ASTERISK-19290. Reported by: Ira Emus)
— Fix crash in ParkAndAnnounce —
(Closes issue ASTERISK-19311. Reported-by: tootai)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.9.2

Thank you for your continued support of Asterisk!

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Asterisk 1.8.9.1 Now Available

admin — Tue, 07 Feb 2012 10:26:24 +0000

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team has announced the release of Asterisk 1.8.9.1.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.9.1 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:

— Fixes deadlocks occuring in chan_agent —
(Closes issue ASTERISK-19285. Reported by: Alex Villacis Lasso)
— Ensure entering T.38 passthrough does not cause an infinite loop —
(Closes issue ASTERISK-18951. Reported-by: Kristijan Vrban)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.9.1

Thank you for your continued support of Asterisk!

Share and Enjoy:

Asterisk 10.1.1 Now Available

admin — Tue, 07 Feb 2012 10:24:36 +0000

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team has announced the release of Asterisk 10.1.1. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 10.1.1 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:

— Fixes deadlocks occuring in chan_agent —
(Closes issue ASTERISK-19285. Reported by: Alex Villacis Lasso)
— Ensure entering T.38 passthrough does not cause an infinite loop —
(Closes issue ASTERISK-18951. Reported-by: Kristijan Vrban)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.1.1

Thank you for your continued support of Asterisk!

Share and Enjoy:

Asterisk 10.2.0-rc1 Now Available

admin — Wed, 01 Feb 2012 12:10:19 +0000

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!

Share and Enjoy:

Asterisk 1.8.10.0-rc1 Now Available

admin — Wed, 01 Feb 2012 12:08:17 +0000

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!

Share and Enjoy:

Asterisk 10.1.0 Now Available

admin — Fri, 27 Jan 2012 12:59:01 +0000

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team is pleased to announce the release of Asterisk 10.1.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 10.1.0 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:

AST-2012-001: prevent crash when an SDP offer is received with an encrypted video stream when support for video is disabled and res_srtp is loaded.
(closes issue ASTERISK-19202) Reported by: Catalin Sanda
Allow playback of formats that don’t support seeking.ast_streamfile previously did unconditional seeking on files that broke playback of formats that don’t support that functionality. This patch avoids the seek that was causing the problem.
(closes issue ASTERISK-18994) Patched by: Timo Teras
Add pjmedia probation concepts to res_rtp_asterisk’s learning mode.In order to better handle RTP sources with strictrtp enabled (which is the default setting in 10) using the learning mode to figure out new sources when they change is handled by checking for a number of consecutive (by sequence number) packets received to an rtp struct based on a new configurable value called ‘probation’. Also, during learning mode instead
of liberally accepting all packets received, we now reject packets until a clear source has been determined.
Handle AST_CONTROL_UPDATE_RTP_PEER frames in local bridge loop.Failing to handle AST_CONTROL_UPDATE_RTP_PEER frames in the local bridge loop causes the loop to exit prematurely. This causes a variety of negative side effects, depending on when the loop exits. This patch handles the frame by essentially swallowing the frame in the local loop, as the current channel drivers expect the RTP bridge to handle the frame, and, in the case of the local bridge loop, no additional action is necessary. (closes issue ASTERISK-19095) Reported by: Stefan Schmidt Tested by: Matt Jordan
Fix timing source dependency issues with MOH.Prior to this patch, res_musiconhold existed at the same module priority level as the timing sources that it depends on. This would cause a problem when music on hold was reloaded, as the timing source could be changed after res_musiconhold was processed. This patch adds a new module priority level, AST_MODPRI_TIMING, that the various timing modules are now loaded at. This now occurs before loading other resource modules, such that the timing source is guaranteed to be set prior to resolving the timing source dependencies.
(closes issue ASTERISK-17474) Reporter: Luke H Tested by: Luke H, Vladimir Mikhelson, zzsurf, Wes Van Tlghem, elguero, Thomas Arimont Patched by elguero
Fix RTP reference leak.If a blind transfer were initiated using a REFER without a prior reINVITE to place the call on hold, AND if Asterisk were sending RTCP reports, then there was a reference leak for the RTP instance of the transferrer.
(closes issue ASTERISK-19192) Reported by: Tyuta Vitali
Fix blind transfers from failing if an ‘h’ extension is present.This prevents the ‘h’ extension from being run on the transferee channel when it is transferred via a native transfer mechanism such as SIP REFER. (closes issue ASTERISK-19173) Reported by: Ross Beer Tested by: Kristjan Vrban Patches: ASTERISK-19173 by Mark Michelson (license 5049)
Restore call progress code for analog ports.Extracting sig_analog from chan_dahdi lost call progress detection functionality. Fix analog ports from considering a call answered immediately after dialing has completed if the callprogress option is enabled. (closes issue ASTERISK-18841)
Reported by: Richard Miller Patched by Richard Miller
Fix regression that ‘rtp/rtcp set debup ip’ only works when a port was also specified.
(closes issue ASTERISK-18693) Reported by: Davide Dal Reviewed by: Walter Doekes

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.1.0

Thank you for your continued support of Asterisk!

Share and Enjoy:

Asterisk 1.8.9.0 Now Available

admin — Fri, 27 Jan 2012 12:54:01 +0000

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team is pleased to announce the release of Asterisk 1.8.9.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.9.0 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:

AST-2012-001: prevent crash when an SDP offer is received with an encrypted video stream when support for video is disabled and res_srtp is loaded.
(closes issue ASTERISK-19202) Reported by: Catalin Sanda
Handle AST_CONTROL_UPDATE_RTP_PEER frames in local bridge loop.Failing to handle AST_CONTROL_UPDATE_RTP_PEER frames in the local bridge loop causes the loop to exit prematurely. This causes a variety of negative side effects, depending on when the loop exits. This patch handles the frame by essentially swallowing the frame in the local loop, as the current channel drivers expect the RTP bridge to handle the frame, and, in the case of the local bridge loop, no additional action is necessary.
(closes issue ASTERISK-19095) Reported by: Stefan Schmidt Tested by: Matt Jordan
Fix timing source dependency issues with MOH.Prior to this patch, res_musiconhold existed at the same module priority level as the timing sources that it depends on. This would cause a problem when music on hold was reloaded, as the timing source could be changed after res_musiconhold was processed. This patch adds a new module priority level, AST_MODPRI_TIMING, that the various timing modules are now loaded at. This now occurs before loading other resource modules, such that the timing source is guaranteed to be set prior to resolving the timing source dependencies.
(closes issue ASTERISK-17474) Reporter: Luke H Tested by: Luke H, Vladimir Mikhelson, zzsurf, Wes Van Tlghem, elguero, Thomas Arimont
Patched by elguero
Fix RTP reference leak.If a blind transfer were initiated using a REFER without a prior reINVITE to place the call on hold, AND if Asterisk were sending RTCP reports, then there was a reference leak for the RTP instance of the transferrer.
(closes issue ASTERISK-19192) Reported by: Tyuta Vitali
Fix blind transfers from failing if an ‘h’ extension is present.This prevents the ‘h’ extension from being run on the transferee channel when it is transferred via a native transfer mechanism such as SIP REFER. (closes issue ASTERISK-19173) Reported by: Ross Beer Tested by: Kristjan Vrban Patches: ASTERISK-19173 by Mark Michelson (license 5049)
Restore call progress code for analog ports.Extracting sig_analog from chan_dahdi lost call progress detection functionality. Fix analog ports from considering a call answered immediately after dialing has completed if the callprogress option is enabled. (closes issue ASTERISK-18841) Reported by: Richard Miller Patched by Richard Miller
Fix regression that ‘rtp/rtcp set debup ip’ only works when a port was also specified.
(closes issue ASTERISK-18693) Reported by: Davide Dal Reviewed by: Walter Doekes

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.9.0

Thank you for your continued support of Asterisk!

Share and Enjoy:

Asterisk 1.6.2.22 Now Available

admin — Mon, 19 Dec 2011 11:09:26 +0000

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team has announced the release of Asterisk 1.6.2.22.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.6.2.22 corrects two flaws in sip.conf.sample related to AST-2011-013:

The sample file listed *two* values for the ‘nat’ option as being the default. Only ‘yes’ is the default.
The warning about having differing ‘nat’ settings confusingly referred to both peers and users.

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

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

Thank you for your continued support of Asterisk!

Share and Enjoy:

Asterisk 10.0.0 Is Released

admin — Thu, 15 Dec 2011 11:48:54 +0000

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!

Share and Enjoy:

Asterisk 1.8.8.0 Now Available

admin — Thu, 15 Dec 2011 11:43:03 +0000

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team is pleased to announce the release of Asterisk 1.8.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.8.0 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:

Updated SIP 484 handling; added Incomplete control frame
When a SIP phone uses the dial application and receives a 484 Address Incomplete response, if overlapped dialing is enabled for SIP, then the 484 Address Incomplete is forwarded back to the SIP phone and the HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete application dialplan logic was automatically triggered; now, explicit dialplan usage of the application is required.
(Closes ASTERISK-17288. Reported by: Mikael Carlsson Tested by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/1416/)
Prevent IAX2 from getting IPv6 addresses via DNS IAX2 does not support IPv6 and getting such addresses from DNS can cause error messages on the remote end involving bad IPv4 address casts in the presence of IPv6/IPv4 tunnels.
(Closes issue ASTERISK-18090. Patched by Kinsey Moore)
Fix bad RTP media bridges in directmedia calls on peers separated by multiple Asterisk nodes.
(Closes issue ASTERISK-18340. Reported by: Thomas Arimont. Closes issue ASTERISK-17725. Reported by: kwk. Tested by: twilson, jrose)
Fix crashes in ast_rtcp_write()
(Closes issue ASTERISK-18570)
Related issues that look like they are the same problem:
(Issue ASTERISK-17560, ASTERISK-15406, ASTERISK-15257, ASTERISK-13334, ASTERISK-9977, ASTERISK-9716)
Review: https://reviewboard.asterisk.org/r/1444/
Patched by: Russell Bryant
Fix for incorrect voicemail duration in external notifications.
This patch fixes an issue where the voicemail duration was being reported with a duration significantly less than the actual sound file duration.
(Closes ASTERISK-16981. Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1443)
Prevent segfault if call arrives before Asterisk is fully booted.
(Patched by alecdavis. https://reviewboard.asterisk.org/r/1407/)
Fix remote Crash Vulnerability in SIP channel driver (AST-2011-012)
http://downloads.asterisk.org/pub/security/AST-2011-012.pdf
Fix locking order in app_queue.c which caused deadlocks
(Closes issue ASTERISK-18101. Reported by Paul Rolfe, patched by Gregory Nietsky)
(Closes issue ASTERISK-18487. Reported by Jason Legault, patched by Gregory Nietsky)
Fix regression in configure script for libpri capability checks
(Closes issue ASTERISK-18687. Reported by norbert, patched by Richard Mudgett)
Prevent BLF subscriptions from causing deadlocks.
(Closes issue ASTERISK-18663)
Review: https://reviewboard.asterisk.org/r/1563/
Fix deadlock if peer is destroyed while sending MWI notice.
(Closes issue ASTERISK-18747)
Reported by: Gregory Hinton Nietsky
Fix issue with setting defaultenabled on categories that are already enabled by default.
(Closes issue ASTERISK-18738)
Reported by: Paul Belanger
Don’t crash on INFO automon request with no channel AST-2011-014. When automon was enabled in features.conf, it was possible to crash Asterisk by sending an INFO request if no channel had been created yet.
Fixed crash from orphaned MWI subscriptions in chan_sip This patch resolves the issue where MWI subscriptions are orphaned by subsequent SIP SUBSCRIBE messages.
Default to nat=yes; warn when nat in general and peer differ AST-2011-013. It is possible to enumerate SIP usernames when the general and user/peer nat settings differ in whether to respond to the port a request is sent from or the port listed for responses in the Via header. In 1.4 and

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.8.0

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