Asterisk News

Asterisk Releases

Asterisk 12.8.0-rc2 Now Available

Dec 10, 2014

The Asterisk Development Team has announced the second release candidate of Asterisk 12.8.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 12.8.0-rc2 resolves an issue reported by the community and would have not been possible without your participation.

Thank you!

The following is the issue resolved in this release candidate:

  • res_http_websocket: Fix crash due to double freeing memory when receiving a payload length of zero.
    (Closes issue ASTERISK-24472. Reported by Badalian Vyacheslav)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-12.8.0-rc2

Thank you for your continued support of Asterisk!


Asterisk 13.1.0-rc2 Now Available

Dec 10, 2014

The Asterisk Development Team has announced the second release candidate of Asterisk 13.1.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.1.0-rc2 resolves an issue reported by the community and would have not been possible without your participation.

Thank you!

The following is the issue resolved in this release candidate:

  • res_http_websocket: Fix crash due to double freeing memory when receiving a payload length of zero.
    (Closes issue ASTERISK-24472. Reported by Badalian Vyacheslav)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.1.0-rc2

Thank you for your continued support of Asterisk!


Asterisk 13.1.0-rc1 Now Available

Dec 8, 2014

The Asterisk Development Team has announced the first release candidate of Asterisk 13.1.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.1.0-rc1 resolves several issues reported by the community and would have not been possible without your participation.

Thank you!

The following are the issues resolved in this release candidate:

Bug

  • [ASTERISK-15242] - transmit_refer leaks sip_refer structures
  • [ASTERISK-20127] - [Regression] Config.c config_text_file_load() unescapes semicolons ("\;" -> ";") turning them into comments (corruption) on rewrite of a config file
  • [ASTERISK-21721] - SIP Failed to parse multiple Supported: headers
  • [ASTERISK-23651] - Reloading some modules that are loaded already, results in 'No such module' before a successful reload
  • [ASTERISK-24190] - IMAP voicemail causes segfault
  • [ASTERISK-24250] - [patch] Voicemail with multi-recipients To: header fix
  • [ASTERISK-24257] - agent must dial acceptdtmf twice to bridge to queue caller
  • [ASTERISK-24304] - asterisk crashing randomly because of unistim channel
  • [ASTERISK-24307] - Unintentional memory retention in stringfields
  • [ASTERISK-24336] - PJSIP timer_min_se value under 90 causes crash
  • [ASTERISK-24411] - [patch] Status of outbound registration is not changed upon unregistering.
  • [ASTERISK-24430] - missing letter "p" in word response in OriginateResponse event documentation
  • [ASTERISK-24432] - Install refcounter.py when REF_DEBUG is enabled
  • [ASTERISK-24436] - Missing header in res/res_srtp.c when compiling against libsrtp-1.5.0
  • [ASTERISK-24437] - Review implementation of ast_bridge_impart for leaks and document proper usage
  • [ASTERISK-24438] - res_pjsip_multihomed.so blocks Asterisk reload when DNS settings invalid
  • [ASTERISK-24444] - PBX: Crash when generating extension for pattern matching hint
  • [ASTERISK-24447] - Bridge DTMF hooks: Audio doesn't pass when waiting for more matching digits.
  • [ASTERISK-24453] - manager: acl_change_sub leaks
  • [ASTERISK-24454] - app_queue: ao2_iterator not destroyed, causing leak
  • [ASTERISK-24455] - func_cdr: CDR_PROP leaks payload
  • [ASTERISK-24457] - res_fax: fax gateway frames leak
  • [ASTERISK-24458] - chan_phone fails to build on big endian systems
  • [ASTERISK-24462] - res_pjsip: Stale qualify statistics after disablementation
  • [ASTERISK-24465] - audiohooks list leaks reference to formats
  • [ASTERISK-24466] - app_queue: fix a couple leaks to struct call_queue
  • [ASTERISK-24468] - Incoming UCS2 encoded SMS truncated if SMS length exceeds 50 (roughly) national symbols
  • [ASTERISK-24469] - Security Vulnerability: Mixed IPv4/IPv6 ACLs allow blocked addresses through
  • [ASTERISK-24471] - Crash - assert_fail in libc in pjmedia_sdp_neg_negotiate from /usr/local/lib/libpjmedia.so.2
  • [ASTERISK-24476] - main/app.c / app_voicemail: ast_writestream leaks
  • [ASTERISK-24480] - res_http_websockets: Module reference decrease below zero
  • [ASTERISK-24482] - func_talkdetect: Fix stasis message leak in audiohook callback
  • [ASTERISK-24487] - configuration: sections should be loadable as template even when not marked
  • [ASTERISK-24489] - Crash: Asterisk crashes when converting RTCP packet to JSON for res_hep_rtcp and report blocks are greater than 1
  • [ASTERISK-24491] - Memory leak in res_hep
  • [ASTERISK-24492] - main/file.c: ast_filestream sometimes causes extra calls to ast_module_unref
  • [ASTERISK-24498] - Segmentation fault in res_hep_rtcp on attended transfer
  • [ASTERISK-24499] - Need more explicit debug when PJSIP dialstring is invalid
  • [ASTERISK-24500] - Regression introduced in chan_mgcp by SVN revision r227276
  • [ASTERISK-24501] - ARI: Moving a channel between bridges followed by a hangup can cause an ARI client to not receive an expected ChannelLeftBridge event before StasisEnd
  • [ASTERISK-24502] - Build fails when dev-mode, dont optimize and coverage are enabled
  • [ASTERISK-24504] - chan_console: Fix reference leaks to pvt
  • [ASTERISK-24505] - manager: http connections leak references
  • [ASTERISK-24508] - pjsip - REFER request from SNOM is rejected with "400 bad request" - DEBUG shows "Received a REFER without a parseable Refer-To"
  • [ASTERISK-24516] - [patch]Asterisk segfaults when playing back voicemail under high concurrency with an IMAP backend
  • [ASTERISK-24522] - ConfBridge: delay occurs between kicking all endmarked users when last marked user leaves
  • [ASTERISK-24528] - res_pjsip_refer: Sending INVITE with Replaces in-dialog with invalid target causes crash
  • [ASTERISK-24531] - res_pjsip_acl: ACLs not applied on initial module load
  • [ASTERISK-24533] - 2 threads created per chan_sip entry
  • [ASTERISK-24535] - stringfields: Fix regression from fix for unintentional memory retention and another issue exposed by the fix
  • [ASTERISK-24537] - Stasis: StasisStart/StasisEnd events are not reliably transmitted during transfers
  • [ASTERISK-24542] - [patch]Failure showing codecs via 'core show channeltype <tech>'
  • [ASTERISK-24556] - Asterisk 13 core dumps when calling from pjsip extension to another pjsip extension
  • [ASTERISK-24572] - [patch]App_meetme is loaded without its defaults when the configuration file is missing
  • [ASTERISK-24573] - [patch]Out of sync conversation recording when divided in multiple recordings

Improvement

  • [ASTERISK-24279] - Documentation: Clarify the behaviour of the CDR property 'unanswered'
  • [ASTERISK-24283] - [patch]Microseconds precision in the eventtime column in the cel_odbc module
  • [ASTERISK-24530] - [patch] app_record stripping 1/4 second from recordings
  • [ASTERISK-24577] - Speed up loopback switches by avoiding unneeded lookups

New Feature

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.1.0-rc1

Thank you for your continued support of Asterisk!


Asterisk 12.8.0-rc1 Now Available

Dec 8, 2014

The Asterisk Development Team has announced the first release candidate of Asterisk 12.8.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 12.8.0-rc1 resolves several issues reported by the community and would have not been possible without your participation.

Thank you!

The following are the issues resolved in this release candidate:

Bug

  • [ASTERISK-15242] - transmit_refer leaks sip_refer structures
  • [ASTERISK-20127] - [Regression] Config.c config_text_file_load() unescapes semicolons ("\;" -> ";") turning them into comments (corruption) on rewrite of a config file
  • [ASTERISK-23651] - Reloading some modules that are loaded already, results in 'No such module' before a successful reload
  • [ASTERISK-24257] - agent must dial acceptdtmf twice to bridge to queue caller
  • [ASTERISK-24307] - Unintentional memory retention in stringfields
  • [ASTERISK-24336] - PJSIP timer_min_se value under 90 causes crash
  • [ASTERISK-24438] - res_pjsip_multihomed.so blocks Asterisk reload when DNS settings invalid
  • [ASTERISK-24444] - PBX: Crash when generating extension for pattern matching hint
  • [ASTERISK-24447] - Bridge DTMF hooks: Audio doesn't pass when waiting for more matching digits.
  • [ASTERISK-24468] - Incoming UCS2 encoded SMS truncated if SMS length exceeds 50 (roughly) national symbols
  • [ASTERISK-24469] - Security Vulnerability: Mixed IPv4/IPv6 ACLs allow blocked addresses through
  • [ASTERISK-24471] - Crash - assert_fail in libc in pjmedia_sdp_neg_negotiate from /usr/local/lib/libpjmedia.so.2
  • [ASTERISK-24480] - res_http_websockets: Module reference decrease below zero
  • [ASTERISK-24482] - func_talkdetect: Fix stasis message leak in audiohook callback
  • [ASTERISK-24487] - configuration: sections should be loadable as template even when not marked
  • [ASTERISK-24489] - Crash: Asterisk crashes when converting RTCP packet to JSON for res_hep_rtcp and report blocks are greater than 1
  • [ASTERISK-24491] - Memory leak in res_hep
  • [ASTERISK-24492] - main/file.c: ast_filestream sometimes causes extra calls to ast_module_unref
  • [ASTERISK-24498] - Segmentation fault in res_hep_rtcp on attended transfer
  • [ASTERISK-24499] - Need more explicit debug when PJSIP dialstring is invalid
  • [ASTERISK-24500] - Regression introduced in chan_mgcp by SVN revision r227276
  • [ASTERISK-24501] - ARI: Moving a channel between bridges followed by a hangup can cause an ARI client to not receive an expected ChannelLeftBridge event before StasisEnd
  • [ASTERISK-24502] - Build fails when dev-mode, dont optimize and coverage are enabled
  • [ASTERISK-24504] - chan_console: Fix reference leaks to pvt
  • [ASTERISK-24505] - manager: http connections leak references
  • [ASTERISK-24508] - pjsip - REFER request from SNOM is rejected with "400 bad request" - DEBUG shows "Received a REFER without a parseable Refer-To"
  • [ASTERISK-24516] - [patch]Asterisk segfaults when playing back voicemail under high concurrency with an IMAP backend
  • [ASTERISK-24522] - ConfBridge: delay occurs between kicking all endmarked users when last marked user leaves
  • [ASTERISK-24528] - res_pjsip_refer: Sending INVITE with Replaces in-dialog with invalid target causes crash
  • [ASTERISK-24531] - res_pjsip_acl: ACLs not applied on initial module load
  • [ASTERISK-24533] - 2 threads created per chan_sip entry
  • [ASTERISK-24535] - stringfields: Fix regression from fix for unintentional memory retention and another issue exposed by the fix
  • [ASTERISK-24537] - Stasis: StasisStart/StasisEnd events are not reliably transmitted during transfers
  • [ASTERISK-24572] - [patch]App_meetme is loaded without its defaults when the configuration file is missing
  • [ASTERISK-24573] - [patch]Out of sync conversation recording when divided in multiple recordings

Improvement

  • [ASTERISK-24279] - Documentation: Clarify the behaviour of the CDR property 'unanswered'
  • [ASTERISK-24283] - [patch]Microseconds precision in the eventtime column in the cel_odbc module
  • [ASTERISK-24530] - [patch] app_record stripping 1/4 second from recordings
  • [ASTERISK-24577] - Speed up loopback switches by avoiding unneeded lookups

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-12.8.0-rc1

Thank you for your continued support of Asterisk!


Asterisk 11.15.0-rc1 Now Available

Dec 8, 2014

The Asterisk Development Team has announced the first release candidate of Asterisk 11.15.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 11.15.0-rc1 resolves several issues reported by the community and would have not been possible without your participation.

Thank you!

The following are the issues resolved in this release candidate:

Bug

  • [ASTERISK-15242] - transmit_refer leaks sip_refer structures
  • [ASTERISK-20127] - [Regression] Config.c config_text_file_load() unescapes semicolons ("\;" -> ";") turning them into comments (corruption) on rewrite of a config file
  • [ASTERISK-20402] - Unable to cancel (features.conf) attended transfer
  • [ASTERISK-24307] - Unintentional memory retention in stringfields
  • [ASTERISK-24440] - Call leak in Confbridge
  • [ASTERISK-24444] - PBX: Crash when generating extension for pattern matching hint
  • [ASTERISK-24468] - Incoming UCS2 encoded SMS truncated if SMS length exceeds 50 (roughly) national symbols
  • [ASTERISK-24469] - Security Vulnerability: Mixed IPv4/IPv6 ACLs allow blocked addresses through
  • [ASTERISK-24492] - main/file.c: ast_filestream sometimes causes extra calls to ast_module_unref
  • [ASTERISK-24500] - Regression introduced in chan_mgcp by SVN revision r227276
  • [ASTERISK-24502] - Build fails when dev-mode, dont optimize and coverage are enabled
  • [ASTERISK-24504] - chan_console: Fix reference leaks to pvt
  • [ASTERISK-24505] - manager: http connections leak references
  • [ASTERISK-24516] - [patch]Asterisk segfaults when playing back voicemail under high concurrency with an IMAP backend
  • [ASTERISK-24522] - ConfBridge: delay occurs between kicking all endmarked users when last marked user leaves
  • [ASTERISK-24572] - [patch]App_meetme is loaded without its defaults when the configuration file is missing
  • [ASTERISK-24573] - [patch]Out of sync conversation recording when divided in multiple recordings

Improvement

  • [ASTERISK-24283] - [patch]Microseconds precision in the eventtime column in the cel_odbc module
  • [ASTERISK-24530] - [patch] app_record stripping 1/4 second from recordings
  • [ASTERISK-24577] - Speed up loopback switches by avoiding unneeded lookups

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.15.0-rc1

Thank you for your continued support of Asterisk!


Pages

Subscribe to