Posts Tagged dtmf
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!
Asterisk 1.6.2.18-rc1 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 first release candidate of Asterisk 1.6.2.18. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.2.18-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:
- Only offer codecs both sides support for directmedia.
(Closes issue #17403. Reported, patched by one47) - 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) - 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) - Guard against retransmitting BYEs indefinitely during attended transfers with chan_sip.
(Review: https://reviewboard.asterisk.org/r/1077/)
For a full list of changes in this release candidate, please see the ChangeLog:
Thank you for your continued support of Asterisk!
Asterisk 1.6.2.17 Now Available
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the release of Asterisk 1.6.2.17.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.2.17 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 duplicated data in the AstDB when using DIALGROUP()
(Closes issue #18091. Reported by bunny. Patched by tilghman) - Correct issue where res_config_odbc could populate fields with invalid data.
(Closes issue #18251, #18279. Reported by bcnit, zerohalo. Tested by trev, jthurman, elguero, zerohalo. Patched by tilghman) - When using cdr_pgsql the billsec field was not populated correctly on unanswered calls.
(Closes issue #18406. Reported by joscas. Patched by tilghman) - Resolve issue where re-transmissions of SUBSCRIBE could break presence.
(Closes issue #18075. Reported by mdu113. Patched by twilson) - Fix regression causing forwarding voicemails to not work with file storage.
(Closes issue #18358. Reported by cabal95. Patched by jpeeler) - This version of Asterisk includes the new Compiler Flags option BETTER_BACKTRACES which uses libbfd to search for better symbol information
within both the Asterisk binary, as well as loaded modules, to assist when using inline backtraces to track down problems.
(Patched by tilghman) - Resolve several issues with DTMF based attended transfers.
(Closes issues #17999, #17096, #18395, #17273. Reported by iskatel, gelo, shihchaun, grecco. Patched by rmudgett).
NOTE: Be sure to read the ChangeLog for more information about these changes. - Resolve issue where no Music On Hold may be triggered when using res_timing_dahdi.
(Closes issues #18262. Reported by francesco_r. Patched by cjacobson. Tested by francesco_r, rfrantik, one47) - Fix regression that changed behavior of queues when ringing a queue member.
(Closes issue #18747, #18733. Reported by vrban. Patched by qwell.)
Additionally, this release has the changes related to security bulletin AST-2011-002 which can be found at http://downloads.asterisk.org/pub/security/AST-2011-002.pdf
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.17
Thank you for your continued support of Asterisk!
Asterisk 1.4.40 Now Available
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the release of Asterisk 1.4.40. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.40 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:
- Correct issue where res_config_odbc could populate fields with invalid data.
(Closes issue #18251, #18279. Reported by bcnit, zerohalo. Tested by trev, jthurman, elguero, zerohalo. Patched by tilghman) - Resolve issue where re-transmissions of SUBSCRIBE could break presence.
(Closes issue #18075. Reported by mdu113. Patched by twilson) - Resolve issue in res_odbc where it may crash when a query fails.
(Closes issue #18243. Reported, patched by ks3) - Fix CPU spike when pressing DTMF after agent login.
(Closes issue #18130. Reported by rgj. Patched by jpeeler) - Fix cross-compiling issue.
(Closes issue #18301. Reported, patched by abelbeck) - This version of Asterisk includes the new Compiler Flags option BETTER_BACKTRACES which uses libbfd to search for better symbol information within both the Asterisk binary, as well as loaded modules, to assist when using inline backtraces to track down problems.
(Patched by tilghman) - Resolve several issues with DTMF based attended transfers.
(Closes issues #17999, #17096, #18395, #17273. Reported by iskatel, gelo, shihchaun, grecco. Patched by rmudgett).
NOTE: Be sure to read the ChangeLog for more information about these changes. - Fix regression that changed behavior of queues when ringing a queue member.
(Closes issue #18747, #18733. Reported by vrban. Patched by qwell.)
Additionally, this release has the changes related to security bulletin AST-2011-002 which can be found at http://downloads.asterisk.org/pub/security/AST-2011-002.pdf
For a full list of changes in this release, please see the ChangeLog:
Thank you for your continued support of Asterisk!
Asterisk 1.6.2.17-rc2 Now Available
Posted by admin in asterisk, Release Candidates on January 26, 2011
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the second release candidate of Asterisk 1.6.2.17. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.2.17-rc2 resolves the following issues in addition to those included in 1.6.2.17-rc1:
- Resolve several issues with DTMF based attended transfers.
(Closes issues #17999, #17096, #18395, #17273. Reported by iskatel, gelo, shihchaun, grecco. Patched by rmudgett).
NOTE: Be sure to read the ChangeLog for more information about these changes. - Resolve issue where no Music On Hold may be triggered when using res_timing_dahdi.
(Closes issues #18262. Reported by francesco_r. Patched by cjacobson. Tested by francesco_r, rfrantik, one47)
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.17-rc2
Thank you for your continued support of Asterisk!
Asterisk 1.4.40-rc2 Now Available
Posted by admin in asterisk, Release Candidates on January 26, 2011
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the second release candidate of Asterisk 1.4.40. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.40-rc2 resolves the following issues in addition to those included in 1.4.40-rc1:
- Resolve several issues with DTMF based attended transfers.
(Closes issues #17999, #17096, #18395, #17273. Reported by iskatel, gelo, shihchaun, grecco. Patched by rmudgett).
NOTE: Be sure to read the ChangeLog for more information about these changes.
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.40-rc2
Thank you for your continued support of Asterisk!
Asterisk 1.6.2.16-rc1 Now Available
Posted by admin in asterisk, Release Candidates on December 15, 2010
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the first release candidate of Asterisk 1.6.2.16. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.2.16-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:
- 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) - Linux and *BSD disagree on the elements within the ucred structure. Detect which one is in use on the system.
(Closes issue #18384. Reported, patched, tested by bjm, tilghman) - app_followme: Don’t create a Local channel if the target extension does not exist.
(Closes issue #18126. Reported, patched by junky) - Revert code that changed SSRC for DTMF.
(Closes issue #17404, #18189, #18352. Reported by sdolloff, marcbou. rsw686.
Tested by cmbaker82) - Resolve issue where REGISTER request with a Call-ID matching an existing transaction is received it was possible that the REGISTER request would overwrite the initreq of the private structure.
(Closes issue #18051. Reported by eeman. Patched, tested by twilson)
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.16-rc1
Thank you for your continued support of Asterisk!
