Posts Tagged dtmf
Asterisk 10.2.0-rc2 Now Available
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!
Asterisk 1.8.10.0-rc2 Now Available
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!
Asterisk 10.1.2 Now Available
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!
Asterisk 1.8.9.2 Now Available
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!
Asterisk 10.2.0-rc1 Now Available
Posted by admin in asterisk, iLBC, Release Candidates on February 1, 2012
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!
Asterisk 1.8.10.0-rc1 Now Available
Posted by admin in asterisk, iLBC, Release Candidates, t.38 on February 1, 2012
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!
Asterisk 10, Beta 1
Posted by admin in digium, Uncategorized on July 23, 2011
On the heels of Kevin Fleming’s announcement yesterday discussing the changes in the Asterisk versioning scheme, we’d like to formally announce that Asterisk 10, Beta 1 is now available for community testing. Asterisk 10, a Standard Support release, will be the next major release of Asterisk and follows the release of Asterisk 1.8 LTS, a Long Term Support release. For more information on the different releases of Asterisk, check out the Asterisk Versionspage on the Wiki.
Let’s talk about some of its new capabilities.
A major focus of the Asterisk 10 development cycle was Asterisk’s support for media types. In versions of Asterisk 1.8 and prior, Asterisk supported a rather limited number of codecs due to some architectural limitations. Plumbing was ripped out, kitchens were remodeled, girders were swapped, and Asterisk 10 now has a media architecture that’s capable of handling both a nearly unlimited number of codecs as well as codecs with more complex parameters. What does this mean for users? First, it means that Asterisk now comes with some additional codecs, including the 32kHz variant of the Speex codec (previous versions of Asterisk only supported the 8kHz or 16kHz variants), Skype’s Superwideband SILK codec, and pass-through support for the 44.1kHz and 48kHz variants of the CELT format.
Astute readers will note that earlier versions of Asterisk were only capable of operating on 8kHz and 16kHz sampled audio, and that the aforementioned newly-supported codecs operate at rates other than these. You’re absolutely correct. In order to support these new codecs, Asterisk 10 has also been provided with support for a variety of super and ultra-wideband sampling rates, all of which are supported as file format types for file playback or recording.
Asterisk 1.4.42 Now Available (Final Maintenance Release)
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the final maintenance release of Asterisk, version 1.4.42. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
Please note that Asterisk 1.4.42 is the final maintenance release from the 1.4 branch. Support for security related issues will continue until April 21, 2012. For more information about support of the various Asterisk branches, see https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
The release of Asterisk 1.4.42 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:
- Resolve regression with ring groups in the Dial() application
(Closes issue ASTERISK-17874. Reported by mspuhler. Patched by elguero) - Resolve deadlock when using tab completion on the ‘meetme kick’ CLI command when an invalid (non-existent) conference room is specified.
(Closes issue ASTERISK-17771. Reported, patched by zvision) - Resolve issue where voice frames could be dropped when checking for T.38 during early media.
(Closes issue ASTERISK-17705. Reported, patched by oej) - Resolve issue where DYNAMIC_FEATURES would not activate after a recent DTMF fix.
(Closes issue ASTERISK-17914. Reported by vrban. Patched by twilson)
Additionally security announcements AST-2011-010, and AST-2011-011 have been resolved in this release.
For a full list of changes in this release, please see the ChangeLog:
Thank you for your continued support of Asterisk!
Asterisk 1.6.2.19-rc1 Now Available
Posted by admin in asterisk, Release Candidates on June 24, 2011
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the first release candidate of Asterisk 1.6.2.19. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
Please note that Asterisk 1.6.2.19 will be the final maintenance release from the 1.6.2 branch. Support for security related issues will continue for one additional year. For more information about support of the various Asterisk branches, see https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
The release of Asterisk 1.6.2.19-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:
- Don’t broadcast FullyBooted to every AMI connection The FullyBooted event should not be sent to every AMI connection every time someone connects via AMI. It should only be sent to the user who just connected.
(Closes issue #18168. Reported, patched by FeyFre) - Fix thread blocking issue in the sip TCP/TLS implementation.
(Closes issue #18497. Reported by vois. Tested by vois, rossbeer, kowalma, Freddi_Fonet. Patched by dvossel) - Don’t delay DTMF in core bridge while listening for DTMF features.
(Closes issue #15642, #16625. Reported by jasonshugart, sharvanek. Tested by globalnetinc, jde. Patched by oej, twilson) - Fix chan_local crashs in local_fixup()
Thanks OEJ for tracking down the issue and submitting the patch.
(Closes issue #19053. Reported, patched by oej) - Don’t offer video to directmedia callee unless caller offered it as well
(Closes issue #19195. Reported, patched by one47)
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.19-rc1
Thank you for your continued support of Asterisk!
Asterisk 1.8.4-rc2 Now Available
Posted by admin in asterisk, Release Candidates, sip on February 28, 2011
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the second release candidate of Asterisk 1.8.4. This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.8.4-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:
- Resolution of several DTMF based attended transfer issues.
(Closes issue #17999, #17096, #18395, #17273. Reported by iskatel, gelo, shihchuan, grecco. Patched by rmudgett)
NOTE: Be sure to read the ChangeLog for more information about these changes. - Resolve deadlocks related to device states in chan_sip
(Closes issue #18310. Reported, patched by one47. Patched by jpeeler) - Resolve an issue with the Asterisk manager interface leaking memory when disabled.
(Reported internally by kmorgan. Patched by russellb) - Support greetingsfolder as documented in voicemail.conf.sample.
(Closes issue #17870. Reported by edhorton. Patched by seanbright) - Fix channel redirect out of MeetMe() and other issues with channel softhangup
(Closes issue #18585. Reported by oej. Tested by oej, wedhorn, russellb. Patched by russellb) - Fix voicemail sequencing for file based storage.
(Closes issue #18498, #18486. Reported by JJCinAZ, bluefox. Patched by jpeeler) - Set hangup cause in local_hangup so the proper return code of 486 instead of 503 when using Local channels when the far sides returns a busy. Also affects CCSS in Asterisk 1.8+.
(Patched by twilson) - Fix issues with verbose messages not being output to the console.
(Closes issue #18580. Reported by pabelanger. Patched by qwell)
Asterisk 1.8.4-rc1 was not released due to a blocking issue found prior to release. An additional fix was merged into Asterisk 1.8.4-rc2:
- Fix Deadlock with attended transfer of SIP call
(Closes issue #18837. Reported, patched by alecdavis. Tested by alecdavid, Irontec, ZX81, cmaj)
For a full list of changes in this release candidate, please see the ChangeLog:
Thank you for your continued support of Asterisk!
