Posts Tagged dtmf
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!
Asterisk 1.4.39-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 1.4.39. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.39-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:
- Resolve issue where channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call.
(Closes issue #18171. Reported by: SantaFox)
(Closes issue #18185. Reported by: kwemheuer)
(Closes issue #18211. Reported by: zahir_koradia)
(Closes issue #18230. Reported by: vmarrone)
(Closes issue #18299. Reported by: mbrevda)
(Closes issue #18322. Reported by: nerbos) - Fix bugs in saying numbers using the Swedish language syntax
(Closes issue #18355. Reported, patched by oej) - Fix not stopping MOH when transfered local channel queue member is answered. The problem here is only present when local channels are used with the MOH passthru option as well as no optimization (/nm). Patched by jpeeler.
- Improve handling of REGISTER requests with multiple contact headers. Patched by jpeeler.
- 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-1.4.39-rc1
Thank you for your continued support of Asterisk!
Asterisk 1.8.0-Beta3 Now Available
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!
Asterisk 1.4.35-Rc1 and Asterisk 1.6.2.11-Rc1 Now Available
Posted by admin in asterisk, Release Candidates on July 23, 2010
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced release candidates of Asterisk for versions 1.4.35 and 1.6.2.11. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
These release candidates address issues that were reported by the community and resolved since the last round of bug fix releases.
The following is a sampling of issues resolved in these release candidates:
- Send DialPlanComplete as a response, not as a separate event.
(Closes issue #17504. Reported, patched by rrb3942) - Ensure channel placed into MeetMe in ringing state is properly hung up.
(Closes issue #15871. Reported, patched by Ivan) - Fix a problem with RFC 2833 DTMF not being accepted.
(Closes issue #17571. Reported by mdeneen. Tested by richardf, maxochoa, JJCinAZ) - Add option to not do a call forward on a 482 Loop Detected.
(Reviewboard: https://reviewboard.asterisk.org/r/764/ ) - Fix logging message for a stale nonce.
(Closes issue #17582. Reported, patched by kenner) - Fix some issues related to dynamic feature groups in features.conf
(Closes issue #17589. Reported,tested by lmadsen. Patched,tested by russell) - cdr_pgsql does not detect when a table is found. Add an ERROR message letting you know when a failure to get the columns from the database exists.
(Closes issue #17478. Reported, patched by kobaz) - Delete IMAP messages in reverse order to ensure reordering after each expunge does not cause deletion of the wrong message.
(Closes issue #16350. Reported by noahisaac. Patched by tilghman)
For a full list of changes in the current release candidates, please see the ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.35-rc1
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.11-rc1
Issues found in any of these release candidates should be reported to the Asterisk issue tracker at https://issues.asterisk.org
Thank you for your continued support of Asterisk!
Asterisk 1.4.34-Rc2 and Asterisk 1.6.2.10-Rc2 Now Available
Posted by admin in asterisk, Release Candidates on July 14, 2010
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the 2nd set of release candidates of Asterisk for versions 1.4.34 and 1.6.2.10. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
These release candidates address issues that were reported by the community and resolved since the last round of bug fix releases.
The following is are issues resolved in these release candidates:
- Fix problem with RFC 2833 DTMF not being accepted.A recent check was added to ensure that we did not erroneously detect duplicate DTMF when we received packets out of order.
The problem was that the check did not account for the fact that the seqno of an RTP stream will roll over back to 0 after hitting 65535. Now, we have a secondary check that will ensure that the
seqno rolling over will not cause us to stop accepting DTMF.
(Closes issue #17571) - Clear the AST_CDR_FLAG_DIALED flag for channels going into the pbx via the G option in app_dial
(Closes issue #17592)
For a full list of changes in the current release candidates, please see the ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.34-rc2
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.10-rc2
Issues found in any of these release candidates should be reported to the Asterisk issue tracker at https://issues.asterisk.org
Thank you for your continued support of Asterisk!
