Archive for category 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 1.6.2.11 Now Available
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the release of Asterisk 1.6.2.11.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.2.11 resolves several issues reported by the community and would have not been possible without your participation.
Thank you!
The following are a few of the issues resolved by community developers:
- Send DialPlanComplete as a response, not as a separate event. Otherwise, it goes to all manager sessions and may exclude the current session, if the Events mask excludes it.
(Closes issue #17504. Reported, patched by rrb3942) - Allow the “useragent” value to be restored into memory from the realtime backend. This value is purely informational. It does not alter configuration at all.
(Closes issue #16029. Reported, patched by Guggemand) - Fix rt(c)p set debug ip taking wrong argument Also clean up some coding errors.
(Closes issue #17469. Reported, patched by wdoekes) - Ensure channel placed in meetme in ringing state is properly hung up. An outgoing channel placed in meetme while still ringing which was then hung up would not exit meetme and the channel was not properly destroyed.
(Closes issue #15871. Reported, patched by Ivan) - Correct how 100, 200, 300, etc. is said. Also add the crazy British numbers.
(Closes issue #16102. Reported, patched by Delvar) - cdr_pgsql does not detect when a table is found. This change adds an ERROR message to let you know when a failure exists to get the columns from the pgsql database, which typically means that the table does not exist.
(Closes issue #17478. Reported, patched by kobaz) - Avoid crashing when installing a duplicate translation path with a lower cost.
(Closes issue #17092. Reported, patched by moy) - Add missing handling for ringing state for use with queue empty options.
(Closes issue #17471. Reported, patched by jazzy) - Fix reporting estimated queue hold time. Just say the number of seconds (after minutes) rather than doing some incorrect calculation with respect to minutes.
(Closes issue #17498. Reported, patched by corruptor)
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.11
Thank you for your continued support of Asterisk!
Asterisk 1.4.35 Now Available
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the release of Asterisk 1.4.35.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.35 resolves several issues reported by the community and would have not been possible without your participation.
Thank you!
The following are a few of the issues resolved by community developers:
- Ensure channel placed in meetme in ringing state is properly hung up.
(Closes issue #15871. Reported, patched by Ivan) - If all members are paused, the wrong status is indicated.
(Closes issue #17576. Reported, patched by ramonpeek) - Fix logging message for stale nonce.
(Closes issue #17582. Reported, patched by kenner) - Interpret device state AST_DEVICE_UNKNOWN as extension state AST_EXTENSION_NOT_INUSE.
(Closes issue #16035. Reported by francesco_r. Patched by viniciusfontes) - Resolve T.38 negotiation regression.
(Closes issue #16705. Reported by mpiazzatnetbug. Patched by ebroad)
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.35
Thank you for your continued support of Asterisk!
Asterisk 1.8.0-Beta2 Now Available
Posted by admin in Release Candidates, asterisk on July 27, 2010
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta2.
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. Some of the changes include:
- Remove duplicate -c flag when using $(INSTALL)
(Closes issue #17695. Reported, patched by pabelanger) - Don’t re-register CDR module on reload.
(Closes issue #17304. Reported, tested by jnemeth. Patched by tilghman) - Don’t assume qlog is open.
(Closes issue #17704. Reported, tested by vrban. Patched by pabelanger) - Expand the correct value within AST_OPTION_ONLY.
(Closes issue #17703. Reported by stuarth. Patched by seanbright) - Allow for systems without locale support to be usable.
(Closes issue #17697. Reported, patched by pprindeville. Tested by mmichelson) - Fixes for sounds/Makefile to install on systems using older GNU make.
(Closes issue #17716. Reported by farisraouf. Patched by tilghman, qwell, seanbright) - Update logger.conf.sample to include documentation about new ‘fax’ logger level.
(Closes issue #17715. Reported, tested by vrban. Patched by pabelanger)
Asterisk 1.8 contains many new features over previous releases of Asterisk.
A short list of included features includes:
- Secure RTP
- IPv6 Support
- 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-beta2
Thank you for your continued support of Asterisk!
Asterisk 1.6.2.11-Rc2 Now Available
Posted by admin in Release Candidates, asterisk on July 27, 2010
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the next release candidate of Asterisk for version 1.6.2.11. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
This release candidate resolves an issue with sounds/Makefile on older versions of GNU make (such as 3.80 shipped with CentOS 4.8). Some additional changes were done to cleanup the sounds/Makefile and to help avoid additional issues in the future.
For a full list of changes in the current release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.11-rc2
Issues found in this release candidate should be reported to the Asterisk issue tracker at https://issues.asterisk.org
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 Release Candidates, asterisk 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 1.6.2.10 Now Available
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the release of Asterisk 1.6.2.10.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.2.10 resolves several issues reported by the community and would have not been possible without your participation.
Thank you!
The following are a few of the issues resolved by community developers:
- Allow users to specify a port for DUNDI peers.
(Closes issue #17056. Reported, patched by klaus3000) - Decrease the module ref count in sip_hangup when SIP_DEFER_BYE_ON_TRANSFER is set.
(Closes issue #16815. Reported, patched by rain) - If there is realtime configuration, it does not get re-read on reload unless the config file also changes.
(Closes issue #16982. Reported, patched by dmitri) - Send AgentComplete manager event for attended transfers.
(Closes issue #16819. Reported, patched by elbriga) - Correct manager variable ‘EventList’ case.
(Closes issue #17520. Reported, patched by kobaz)
In addition, changes to res_timing_pthread that should make it more stable have also been implemented.
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.10
Thank you for your continued support of Asterisk!
Asterisk 1.4.34 Now Available
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the release of Asterisk 1.4.34.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.34 resolves several issues reported by the community and would have not been possible without your participation.
Thank you!
The following are a few of the issues resolved by community developers:
- Allow users to specify a port for DUNDi peers.
(Closes issue #17056. Reported, patched by klaus3000) - Decrease the module ref count in sip_hangup when SIP_DEFER_BYE_ON_TRANSFER is set.
(Closes issue #16815. Reported, patched by rain) - First caller into a dynamic conference new enters the pin once.
(Closes issue #15878. Reported, patched by pabelanger) - Send AgentComplete manager events in the event of blind and attended transfers.
(Closes issue #16819. Reported, patched by elbriga)
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.34
Thank you for your continued support of Asterisk!
Asterisk 1.8.0-Beta1 Now Available
Posted by admin in Release Candidates, asterisk on July 23, 2010
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta1.
This release marks the beginning of the testing process for the eventual release of Asterisk 1.8.0.
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 hear 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
Asterisk 1.8 contains many new features over previous releases of Asterisk.
A short list of included features includes:
- Secure RTP
- IPv6 Support
- 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-beta1
Thank you for your continued support of Asterisk!
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