The Asterisk Development Team has announced the release of Asterisk 11.21.0.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 11.21.0 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:
Bug
- [ASTERISK-7803] – Update the maximum packetization values in frame.c
- [ASTERISK-24146] – No audio on WebRtc caller side when answer waiting time is more than ~7sec
- [ASTERISK-25135] – RTP Timeout hangup cause code missing
- [ASTERISK-25364] – Issue a TCP connection(kernel) and thread of asterisk is not released
- [ASTERISK-25373] – add documentation for CALLERID(pres) and also the CONNECTEDLINE and REDIRECTING variants
- [ASTERISK-25391] – AMI GetConfigJSON returns invalid JSON
- [ASTERISK-25400] – Hints broken when “CustomPresence” doesn’t exist in AstDB
- [ASTERISK-25434] – Compiler flags not reported in ‘core show settings’ despite usage during compilation
- [ASTERISK-25438] – res_rtp_asterisk: ICE role message even when ICE is not enabled
- [ASTERISK-25443] – IPv6 – Potential issue in via header parsing
- [ASTERISK-25449] – main/sched: Regression introduced by 5c713fdf18f causes erroneous duplicate RTCP messages; other potential scheduling issues in chan_sip/chan_skinny
- [ASTERISK-25455] – Deadlock of PJSIP realtime over res_config_pgsql
- [ASTERISK-25461] – Nested dialplan #includes don’t work as expected.
- [ASTERISK-25476] – chan_sip loses registrations after a while
- [ASTERISK-25494] – build: GCC 5.1.x catches some new const, array bounds and missing paren issues
- [ASTERISK-25498] – Asterisk crashes when negotiating g729 without that module installed
- [ASTERISK-25527] – Quirky xmldoc description wrapping
- [ASTERISK-25537] – format-attribute module: RFC or internal defaults?
- [ASTERISK-25552] – hashtab: Improve NULL tolerance
- [ASTERISK-25569] – app_meetme: Audio quality issues
- [ASTERISK-25585] – rasterisk never hits most of main(), but it’s assumed to
- [ASTERISK-25593] – fastagi: record file closed after sending result
- [ASTERISK-25599] – SLIN Resampling Codec only 80 msec
- [ASTERISK-25609] – Asterisk may crash when calling ast_channel_get_t38_state(c)
- [ASTERISK-25610] – Asterisk crash during “sip reload”
- [ASTERISK-25616] – Warning with a Codec Module which supports PLC with FEC
- [ASTERISK-25640] – pbx: Deadlock on features reload and state change hint.
Improvement
- [ASTERISK-24718] – Add inital support of “sanitize” to configure
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
Thank you for your continued support of Asterisk!