The Asterisk Development Team has announced the release of Asterisk 11.6.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 11.6.0 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:
- [ASTERISK-21036] – Jitter Buffer log file creation doesn’t account for multiple slashes in DAHDI channel names
- [ASTERISK-21064] – Crash when handling ACK on dialog that has no channel
- [ASTERISK-21117] – Bad interpretation of the file chan_dahdi.conf when using open r2 parameters
- [ASTERISK-21789] – ast_http_get_cookies() fails in the presence of RFC2965 Cookie2 header
- [ASTERISK-21825] – websocket segmentation fault on certain invalid input
- [ASTERISK-21847] – Segfault due to dahdi_restart and round robin
- [ASTERISK-21859] – Confbridge doesn’t tear down an empty conference bridge when all users were kicked via end_marked=yes. Also, side effect crashes.
- [ASTERISK-21903] – Return proper result upon error when running some AGI commands
- [ASTERISK-21907] – Crash – segfault – When executing a MeetMeAdmin command that requires a member, without specifying a member
- [ASTERISK-21964] – SIP TLS Register statement fails if sip.conf register directive uses peer name.
- [ASTERISK-21980] – Error message for QUEUE_MEMBER when member is not in queue is unclear
- [ASTERISK-21997] – – Incorrect Ring tone for Malaysia
- [ASTERISK-22007] – chan_sip: segfault with invalid sdp
- [ASTERISK-22071] – chan_sip doesn’t respect Via ..completely
- [ASTERISK-22179] – Update copyright headers – they’re so last year
- [ASTERISK-22185] – Crash in sip_msg_send in chan_sip.c when from parameter is malformed
- [ASTERISK-22189] – Wrap up time is ignored for queue members who are members in multiple queues
- [ASTERISK-22221] – The masquerade super-test fails on all Asterisk versions
- [ASTERISK-22237] – http_shutdown incomplete
- [ASTERISK-22238] – astfd and threadstorage debug cli commands are not unregistered
- [ASTERISK-22248] – test_sip_rtpqos corrupts dialogs container
- [ASTERISK-22249] – xmldoc.c leaks an attribute
- [ASTERISK-22259] – cel segfault on invalid cel.conf
- [ASTERISK-22269] – app_meetme: wrong bit value for CONFFLAG_DONT_DENOISE
- [ASTERISK-22275] – T.38 Passthrough broken if peer doen’t report T38MaxBitRate
- [ASTERISK-22276] – Test test_hashtab_thrash fails on 32-bit machines when compiled without DEBUG_THREADS
- [ASTERISK-22308] – Documentation – chan_dahdi, waitfordialtone is not boolean, it’s time in milliseconds
- [ASTERISK-22347] – res_xmpp timeout when google sends blank responses ‘ ‘
- [ASTERISK-22349] – Behavioral change in Asterisk 11: udptl now requires config file to function
- [ASTERISK-22365] – chan_h323 can’t be compiled
- [ASTERISK-22368] – mixmonitor_free leaks filename
- [ASTERISK-22395] – manager.c and res_agi.c leak results from ast_xmldoc_printable
- [ASTERISK-22413] – features.c TEST_FRAMEWORK leaks channel reference, preventing graceful shutdown
- [ASTERISK-22414] – voicemail and test_voicemail_api leaks
- [ASTERISK-22416] – Segmentation fault (in process_applicationmap_line, at features.c) when using improper feature mapping syntax
- [ASTERISK-22424] – bridge_native_rtp: Asterisk 12 attempts to remotely bridge on 200OK response to invite when the 200 lacks SDP
- [ASTERISK-22435] – jabber/xmpp MWI distributed pubsub issue where the mailbox and context get swapped at the remote end
- [ASTERISK-22450] – No CLI response from database show
- [ASTERISK-22455] – Asterisk 12 on Ubuntu Lucid deadlocks with DEBUG_THREADS+OPTIONAL_API enabled
- [ASTERISK-22504] – chan_iax2: wrong expiry time in astdb
- [ASTERISK-22536] – MALLOC_DEBUG causes /tmp/refs to be written, even if REF_DEBUG is not defined
- [ASTERISK-22562] – Open blockers for 11.6.0
- [ASTERISK-20969] – Fix func_channel documentation for sip/iax2/dadhi
- [ASTERISK-21021] – SQL script to create queue_log table in PostgreSQL
- [ASTERISK-21717] – – Documentation for PASSTHRU function is unclear
- [ASTERISK-21953] – connectedline parameter not documented
- [ASTERISK-22263] – ‘queue add member …’ help text update
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.6.0
Thank you for your continued support of Asterisk!