Posts Tagged MeetMe
Asterisk 1.8.6.0-rc1 Now Available
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team announces the first release candidate of Asterisk 1.8.6.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.8.6.0-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:
- Fix an issue with Music on Hold classes losing files in playlist when realtime is used.
(Closes issue ASTERISK-17875. Reported by David Cunningham. Patched by Igor Goncharovsky) - Resolve a potential crash in chan_sip when utilizing auth= and performing a ‘sip reload’ from the console.
(Closes issue ASTERISK-17939. Reported by wdoekes. Patched by Richard Mudgett) - Address some improper sql statements in res_odbc that would cause an update to fail on realtime peers due to trying to set as “(NULL)” rather than an actual NULL.
(Closes issue ASTERISK-17791. Reported by marcelloceschia. Patched by Tilghman Lesher) - Resolve issue where 403 Forbidden would always be sent maximum number of times regardless to receipt of ACK.
(Patched by Richard Mudgett) - Updated chan_gtalk to work with changes made by Google.
(Closes issue ASTERISK-18804. Patched by Terry Wilson) - Resolve issue where if a call to MeetMe includes both the dynamic(D) and always request PIN(P) options, MeetMe will ask for the PIN two times: once for creating the conference and once for entering the conference.
(Patched by Kinsey Moore) - Fix New Zealand indications profile based on http://www.telepermit.co.nz/TNA102.pdf
(Closes issue ASTERISK-16263. Reported, Patched by richardf)
For a full list of changes in this release candidate, please see the ChangeLog:
Thank you for your continued support of Asterisk!
Asterisk 1.8.5-rc1 Now Available
Posted by admin in asterisk, Release Candidates, sip on June 29, 2011
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team announces the first release candidate of Asterisk 1.8.5. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.8.5-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:
- Fix Deadlock with attended transfer of SIP call
(Closes issue #18837. Reported, patched by alecdavis. Tested by Irontec, ZX81, cmaj) - Fixes thread blocking issue in the sip TCP/TLS implementation.
(Closes issue #18497. Reported by vois. Patched by dvossel. Tested by vois, rossbeer, kowalma, Freddi_Fonet) - Be more tolerant of what URI we accept for call completion PUBLISH requests.
(Closes issue #18946. Reported by GeorgeKonopacki. Patched by mmichelson) - Fix a nasty chanspy bug which was causing a channel leak every time a spied on channel made a call.
(Closes issue #18742. Reported by jkister. Tested by jcovert, jrose) - This patch fixes a bug with MeetMe behavior where the ‘P’ option for always prompting for a pin is ignored for the first caller.
(Closes issue #18070. Reported by mav3rick. Patched by bbryant) - Fix issue where Asterisk does not hangup a channel after endpoint hangs up. If the call that the dialplan started an AGI script for is hungup while the AGI script is in the middle of a command then the AGI script is not notified of the hangup.
(Closes issue #17954, #18492. Reported by mn3250, devmod. Patched by rmudgett) - Resolve issue where leaving a voicemail, the MWI message is never sent. The same thing happens when checking a voicemail and marking it as read.
(Closes issue ASTERISK-18002. Reported by Leif Madsen. Resolved by Richard Mudgett) - Resolve issue where wait for leader with Music On Hold allows crosstalk between participants. Parenthesis in the wrong position. Regression from issue #14365 when expanding conference flags to use 64 bits.
(Closes issue #18418. Reported by MrHanMan. Patched by rmudgett) - Fix timerfd locking issue.
(Closes ASTERISK-17867, ASTERISK-17415. Patched by kobaz)
For a full list of changes in this release candidate, please see the ChangeLog:
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 1.4.41-rc1 Now Available
Posted by admin in asterisk, Release Candidates on February 28, 2011
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the first release candidate of Asterisk 1.4.41. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.41-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. - 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 1.4.35-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-Rc1 And Asterisk 1.6.2.10-Rc1 Now Available
Posted by admin in asterisk, Release Candidates on June 29, 2010
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!
Asterisk 1.4.29 Now Available
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the release of Asterisk 1.4.29.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.29 resolved several issues reported by the community, and would have not been possible without your participation. Thank you!
- Fix to Monitor which previously assumed the file to write to did not contain pathing.
(Closes issue #16377, #16376. Reported by bcnit. Patched by dant. - Propertly set T.38 attributes and don’t return before T.38 ports are configured when T.38 is found but no audio stream is found.
(Closes issue #16318. Reported by bird_of_Luck. Tested by vrban, mihaill. Patched by vrban, mnicholson.) - Avoid crashes with large numbers of MeetMe conferences.
(Closes issue #16509. Reported by Kashif Raza. Tested, Patched by seanbright.) - Change in ‘sip show channels’ display format allowing more digits for CID.
(Closes issue #16459. Reported, Patched by Rzadzins. - Revise documentation on disposition values to the actual values used.
(Closes issue #16289. Reported by wdoekes.)
A summary of changes in this release can be found in the release summary:
http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.4.29-summary.txt
For a full list of changes in this releases, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.29
Thank you for your continued support of Asterisk!
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