Posts Tagged 1.6

Asterisk 1.6.2.11 Now Available

Asterisk The Open Source PBX & Telephony Platform

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!

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Asterisk 1.6.2.11-Rc2 Now Available

Asterisk The Open Source PBX & Telephony Platform

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!

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Asterisk 1.4.35-Rc1 and Asterisk 1.6.2.11-Rc1 Now Available

Asterisk The Open Source PBX & Telephony Platform

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:

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!

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Asterisk 1.6.2.10 Now Available

Asterisk The Open Source PBX & Telephony Platform

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!

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Asterisk 1.4.34-Rc2 and Asterisk 1.6.2.10-Rc2 Now Available

Asterisk The Open Source PBX & Telephony Platform

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:

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!

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Asterisk 1.4.34-Rc1 And Asterisk 1.6.2.10-Rc1 Now Available

Asterisk The Open Source PBX & Telephony Platform

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team has announced 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 a sampling of issues resolved in these release candidates:

  • If MeetMe is configured to use dynamic conference numbers, then the first caller (which creates the conference) had to enter the PIN number twice.
    (Closes issue #15878. Reported by shawkris. Patched by pabelanger)
  • Send AgentComplete manager events in the event of blind and attended transfers.
    (Closes issue #16819. Reported and patched by elbriga)
  • Correct manager variable ‘EventList’ case.
    (Closes issue #17520. Reported and patched by kobaz)
  • If there is realtime configuration, it does not get re-read on reload unless the config file also changes.
    (Closes issue #16982. Reported and patched by dmitri. Tested by atis)

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-rc1
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.10-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!

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Asterisk 1.6.2.9 Now Available

Asterisk The Open Source PBX & Telephony Platform

Asterisk The Open Source PBX & Telephony Platform

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:

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!

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Asterisk 1.4.33-rc2 and Asterisk 1.6.2.9-rc3 Now Available

Asterisk The Open Source PBX & Telephony Platform

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team has announced new release candidates of Asterisk for versions 1.4.33 and 1.6.2.9. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

These releases resolve two issues reported by the community:

  • Fix Debian init script to not use -c.When using the init script as-is currently, it could cause issues on Debian such as high CPU usage. This fix has worked for several people so I’m implementing the change. We now handle color displays properly.
    (Closes issue #16784. Reported, tested by pabelanger. Patched by tilghman)
  • Ensure signals are not blocked inside other signal handlers.This eliminates the annoying on the console.
    (Closes issue #17477. Reported by jvandal. 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-1.4.33-rc2
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.9-rc3

Issues found in these release candidates should be reported to the Asterisk issue tracker at https://issues.asterisk.org

Thank you for your continued support of Asterisk!

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Asterisk 1.6.2.9-Rc2 Now Available

Asterisk The Open Source PBX & Telephony Platform

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team has announced a new release candidate of Asterisk for version 1.6.2.9. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

This release candidate resolves a crash in DTMF detection. See issues #17371 and #17474 for more information.

For a full list of changes in the current release candidate, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.9-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!

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No Comments

Asterisk 1.6.2.8 Now Available

Asterisk The Open Source PBX & Telephony Platform

Asterisk The Open Source PBX & Telephony Platform

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.
  • (Closes issue #16590. Reported, patched by kkm)

  • Small error in the T.140 RTP port verbose log.
    (Closes issue #16998. Reported, patched by frawd. Tested by russell)

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!

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