Posts Tagged Releases
Asterisk 1.6.2.11 Now Available
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
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 1.6.2.10 Now Available
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
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 1.4.33.1 Released
The Asterisk Development Team has announced the release of Asterisk 1.4.33.1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.33.1 resolves a regression involving the use of FXO signaling in chan_dahdi where a channel could continue ringing after it has been answered.
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.33.1
Thank you for your continued support of Asterisk!
Asterisk 1.4.33 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.4.33.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.33 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:
- Remove arbitrary size limitation for hints
(Closes issue #17257. Reported, patched by tim_ringenbach) - Fix incorrectly typed indications for [nz] stutter and dialrecall
(Closes issue #17359. Reported, patched by alecdavis) - Make AgentComplete message more consistent
(Closes issue #15638. Reported, patched by elbriga) - Missing fallback to audio fax feature when T.38 re-INVITE failed
(Closes issue #16692. Reported, patched by vrban) - Don’t hang up on a queue caller if the file we attempt to play does not exist
(Closes issue #17061. Reported by RoadKill)
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.33
Thank you for your continued support of Asterisk!
Asterisk 1.6.2.9 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.6.2.9.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.2.9 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:
- Fix the PickupChan() application
(Closes issue #16863. Reported, patched by schern. Patched by cjacobsen.
Tested by Graber, cjacobsen, lathama, rickead2000, dvossel) - Improve logging by displaying line number
(Closes issue #16303. Reported by dant. Patched by pabelanger. Tested by
dant, pabelanger, lmadsen) - Notify CLI when modules are loaded/unloaded
(Closes issue #17308. Reported, patched by pabelanger. Tested by russell) - Make the Makefile logic more explicit and move the Snow Leopard logic down to
where it’s not executed on non-Darwin systems
(Closes issue #17028. Reported by pabelanger. Patched by seanbright,
tilghman. Tested by pabelanger) - Manager cookies are not compatible with RFC2109. Make that no longer true.
(Closes issue #17231. Reported, patched by ecarruda) - With IMAP backend, messages in INBOX were counted twice for MWI
(Closes issue #17135. Reported by edhorton. Patched by ebroad, tilghman) - Fix possible segfault when logging
(Closes issue #17331. Reported, patched by under. Patched by dvossel) - Fix memory hogging behavior of app_queue
(Closes issue #17081. Reported by wliegel. Patched by mmichelson) - Allow type=user SIP endpoints to be loaded properly from realtime
(Closes issue #16021. Reported, patched by Guggemand)
Additionally, the following issue may be of interest:
- Fix transcode_via_sln option with SIP calls and improve PLC usage
(Review: https://reviewboard.asterisk.org/r/622/)
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.9
Thank you for your continued support of Asterisk!
Libpri 1.4.11.1 Now Available
The Asterisk Development Team has announced the release of version 1.4.11.1 of libpri. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri/
This release fixes a regression in multi component FACILITY messages and includes a minor bug fix for BRI spans:
- Multi component FACILITY messages only process the first component. The code was only processing the first ROSE component in the facility message.
(Closes issue #17428. Reported, tested by: patrol-cz. Patched by rmudgett) - Inband disconnect setting does nothing on BRI spans. The acceptinbanddisconnect flag is not inherited when creating a new TEI and thus rendering the setting (and its respective equivalent in Asterisk) a no-op on BRI setups.
(Closes issue #15265. Reported, patched, tested by: paravoid)
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/libpri/releases/ChangeLog-1.4.11.1
Thank you for your continued support of Asterisk!
Asterisk 1.6.2.8 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.6.2.8. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.2.8 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:
- Enable auto complete for CLI command ‘logger set level’.
(Closes issue #17152. Reported, patched by pabelanger) - Make the mixmonitor thread process audio frames faster.
(Closes issue #17078. Reported, tested by geoff2010. Patched by dhubbard) - Add missing ‘useragent’ field to sip-friends.sql file.
(Closes issue #17171. Reported, patched by thehar) - Add example dialplan for dialing ISN numbers (http://www.freenum.org)
(Closes issue #17058. Reported, patched by pprindeville) - Fix issue with double “sip:” in header field.
(Closes issue #15847. Reported, patched by ebroad) - Add ability to generate ASCII documentation from the TeX files by running ‘make asterisk.txt’.
(Closes issue #17220. Reported by lmadsen. Tested, patched by pabelanger) - When StopMonitor() is called, ensure that it will not be restarted by a channel event.
- Small error in the T.140 RTP port verbose log.
(Closes issue #16998. Reported, patched by frawd. Tested by russell)
(Closes issue #16590. Reported, patched by kkm)
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.8
Thank you for your continued support of Asterisk!
Asterisk 1.4.32 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.4.32. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.32 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:
- Make the mixmonitor thread process audio frames faster.
(Closes issue #17078. Reported, tested by: geoff2010. Patched by dhubbard) - When StopMonitor is called, ensure that it will not be restarted by a channel event.
(Closes issue #16590. Reported, patched by: kkm) - Fix up hidecallerid feature in chan_dahdi.
(Closes issue #17143, #7321. Reported, patched by djenson99) - Resolve deadlocks in chan_local.
(Closes issue #17185. Reported, tested by schmoozecom, GameGamer43) - Ensure channel state is not incorrectly set in the case of a very early answer.
(Closes issue #17067. Reported, patched by tzafrir) - Registration fix for SIP realtime.
(Closes issue #17266. Reported, patched by Nick_Lewis. Tested by sberney)
For a full list of changes in the current release, please see the
ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.32
Thank you for your continued support of Asterisk!
