Posts Tagged mwi
Asterisk 1.8.8.0 Now Available
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team is pleased to announce the release of Asterisk 1.8.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.8.8.0 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:
- Updated SIP 484 handling; added Incomplete control frame
When a SIP phone uses the dial application and receives a 484 Address Incomplete response, if overlapped dialing is enabled for SIP, then the 484 Address Incomplete is forwarded back to the SIP phone and the HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete application dialplan logic was automatically triggered; now, explicit dialplan usage of the application is required.
(Closes ASTERISK-17288. Reported by: Mikael Carlsson Tested by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/1416/) - Prevent IAX2 from getting IPv6 addresses via DNS IAX2 does not support IPv6 and getting such addresses from DNS can cause error messages on the remote end involving bad IPv4 address casts in the presence of IPv6/IPv4 tunnels.
(Closes issue ASTERISK-18090. Patched by Kinsey Moore) - Fix bad RTP media bridges in directmedia calls on peers separated by multiple Asterisk nodes.
(Closes issue ASTERISK-18340. Reported by: Thomas Arimont. Closes issue ASTERISK-17725. Reported by: kwk. Tested by: twilson, jrose) - Fix crashes in ast_rtcp_write()
(Closes issue ASTERISK-18570)
Related issues that look like they are the same problem:
(Issue ASTERISK-17560, ASTERISK-15406, ASTERISK-15257, ASTERISK-13334, ASTERISK-9977, ASTERISK-9716)
Review: https://reviewboard.asterisk.org/r/1444/
Patched by: Russell Bryant - Fix for incorrect voicemail duration in external notifications.
This patch fixes an issue where the voicemail duration was being reported with a duration significantly less than the actual sound file duration.
(Closes ASTERISK-16981. Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1443) - Prevent segfault if call arrives before Asterisk is fully booted.
(Patched by alecdavis. https://reviewboard.asterisk.org/r/1407/) - Fix remote Crash Vulnerability in SIP channel driver (AST-2011-012)
http://downloads.asterisk.org/pub/security/AST-2011-012.pdf
- Fix locking order in app_queue.c which caused deadlocks
(Closes issue ASTERISK-18101. Reported by Paul Rolfe, patched by Gregory Nietsky)
(Closes issue ASTERISK-18487. Reported by Jason Legault, patched by Gregory Nietsky) - Fix regression in configure script for libpri capability checks
(Closes issue ASTERISK-18687. Reported by norbert, patched by Richard Mudgett) - Prevent BLF subscriptions from causing deadlocks.
(Closes issue ASTERISK-18663)
Review: https://reviewboard.asterisk.org/r/1563/ - Fix deadlock if peer is destroyed while sending MWI notice.
(Closes issue ASTERISK-18747)
Reported by: Gregory Hinton Nietsky - Fix issue with setting defaultenabled on categories that are already enabled by default.
(Closes issue ASTERISK-18738)
Reported by: Paul Belanger - Don’t crash on INFO automon request with no channel AST-2011-014. When automon was enabled in features.conf, it was possible to crash Asterisk by sending an INFO request if no channel had been created yet.
- Fixed crash from orphaned MWI subscriptions in chan_sip This patch resolves the issue where MWI subscriptions are orphaned by subsequent SIP SUBSCRIBE messages.
- Default to nat=yes; warn when nat in general and peer differ AST-2011-013. It is possible to enumerate SIP usernames when the general and user/peer nat settings differ in whether to respond to the port a request is sent from or the port listed for responses in the Via header. In 1.4 and
For a full list of changes in this release, please see the ChangeLog:
Thank you for your continued support of Asterisk!
Asterisk 1.8.8.0-rc5 Now Available
Posted by admin in asterisk, Release Candidates on December 9, 2011
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the fifth release candidate of Asterisk 1.8.8.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.8.8.0-rc5 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:
- Don’t crash on INFO automon request with no channel
AST-2011-014. When automon was enabled in features.conf, it was possible to crash Asterisk by sending an INFO request if no channel had been created yet. - Fixed crash from orphaned MWI subscriptions in chan_sip
This patch resolves the issue where MWI subscriptions are orphaned by subsequent SIP SUBSCRIBE messages. - Default to nat=yes; warn when nat in general and peer differ
AST-2011-013. It is possible to enumerate SIP usernames when the general and user/peer nat settings differ in whether to respond to the port a request is sent from or the port listed for responses in the Via header. In 1.4 and 1.6.2, this would mean if one setting was nat=yes or nat=route and the other was either nat=no or nat=never. In 1.8 and 10, this would mean when one was nat=force_rport and the other was nat=no.In order to address this problem, it was decided to switch the default behavior to nat=yes/force_rport as it is the most commonly used option and to strongly discourage setting nat per-peer/user when at all possible.
For a full list of changes in this release candidate, please see the ChangeLog:
Thank you for your continued support of Asterisk!
Asterisk 1.8.3 Now Available
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the release of Asterisk 1.8.3. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.8.3 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) - Ensure the ipaddr field in realtime is large enough to handle IPv6 addresses.
(Closes issue #18464. Reported, patched by IgorG) - Reworking parsing of mwi => lines to resolve a segfault. Also add a set of unit tests for the function that does the parsing.
(Closes issue #18350. Reported by gbour. Patched by Marquis) - When using cdr_pgsql the billsec field was not populated correctly on unanswered calls.
(Closes issue #18406. Reported by joscas. Patched by tilghman) - Resolve memory leak in iCalendar and Exchange calendaring modules.
(Closes issue #18521. Reported, patched by pitel. Tested by cervajs) - 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 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) - Resolve a memory leak when the Asterisk Manager Interface is disabled.
(Reported internally by kmorgan. Patched by russellb) - Reimplemented fax session reservation to reverse the ABI breakage introduced in r297486.
(Reported internally. Patched by mnicholson) - Fix regression that changed behavior of queues when ringing a queue member.
(Closes issue #18747, #18733. Reported by vrban. Patched by qwell.) - Resolve deadlock involving REFER.
(Closes issue #18403. Reported, tested by jthurman. Patched by jpeeler.)
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 1.4.26 released
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team is pleased to announce the release of Asterisk 1.4.26.
Asterisk 1.4.26 is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
This release resolves a large assortment of issues reported by the community.
For a summary of the changes in this release, please see the release summary:
http://svn.asterisk.org/svn/asterisk/tags/1.4.26/asterisk-1.4.26-summary.txt
For a full list of changes in this release, please see the ChangeLog:
http://svn.asterisk.org/svn/asterisk/tags/1.4.26/ChangeLog
The following list of issues were resolved with the participation of the community, and this release would not have been possible without your help!
- Fix handling of the ‘state_interface’ option of the ‘queue add member’ CLI command.
(closes issue #15181. Reported and tested by loloski. Patch by seanbright) - Fix a possible crash in pbx_spool.
(closes issue #15072. Reported, and patched by garlew) - MWI NOTIFY contains a wrong URI if Asterisk listens to non-standard port and transport.
(closes issue #14659. Reported, patch, and testing by klaus3000) - Don’t fast forward past the end of a message.
(closes issue #14554. Reported and patched by lacoursj) - Prevent phantom calls to queue members.
(closes issue #14631. Reported and patched by latinsud) - No audio on calls from Asterisk to various ISDN devices until DTMF sent by caller.
(closes issues #15420, #15416, #15389, #15205. Reported by scottbmilne, avinoash, alecdavis, vinsik. Tested by scottbmilne, alecdavis. Patched by alecdavis)
Thank you for your continued support of Asterisk!
