The Asterisk Development Team has announced the first release candidate of Asterisk 12.4.0. This release candidate is available for immediate
download at http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 12.4.0-rc1 resolves several issues reported by the community and would have not been possible without your participation.
The following are the issues resolved in this release candidate:
- [ASTERISK-18230] – sometimes dialplan switches disappear when merging contexts between pbx_lua and pbx_config
- [ASTERISK-21965] – Bug-fixed version of safe_asterisk not installed over old version
- [ASTERISK-22551] – Session timer : UAS (Asterisk) starts counting at Invite, UAC starts counting at 200 OK.
- [ASTERISK-23035] – ConfBridge with name longer than max (32 chars) results in several bridges with same conf_name
- [ASTERISK-23489] – Vulnerability in res_pjsip_pubsub: unauthenticated remote crash in during MWI unsubscribe without being subscribed
- [ASTERISK-23499] – app_agent_pool: Interval hook prevents channel from being hung up
- [ASTERISK-23541] – Asterisk 12.1.0 Not respecting directmedia=no and issuing REINVITE
- [ASTERISK-23582] – Inconsistent column length in *odbc
- [ASTERISK-23609] – Security: AMI action MixMonitor allows arbitrary programs to be run
- [ASTERISK-23673] – Security: DOS by consuming the number of allowed HTTP connections.
- [ASTERISK-23683] – #includes – wildcard character in a path more than one directory deep – results in no config parsing on module reload
- [ASTERISK-23718] – res_pjsip_incoming_blind_request: crash with NULL session channel
- [ASTERISK-23721] – Calls to PJSIP endpoints with video enabled result in leaked RTP ports
- [ASTERISK-23766] – Specify timeout for database write in SQLite
- [ASTERISK-23790] – – SIP From headers longer than 256 characters result in dropped call and ‘No closing bracket’ warnings.
- [ASTERISK-23792] – Mutex left locked in chan_unistim.c
- [ASTERISK-23802] – Security: Deadlock in res_pjsip_pubsub on transaction timeout
- [ASTERISK-23803] – AMI action UpdateConfig EmptyCat clears all categories but the requested one
- [ASTERISK-23814] – No call started after peer dialed
- [ASTERISK-23818] – PBX_Lua: after asterisk startup module is loaded, but dialplan not available
- [ASTERISK-23824] – ConfBridge: Users cannot be muted via CLI or AMI when waiting to enter a conference
- [ASTERISK-23827] – autoservice thread doesn’t exit at shutdown
- [ASTERISK-23834] – res_rtp_asterisk debug message gives wrong length if ICE
- [ASTERISK-23844] – Load of pbx_lua fails on sample extensions.lua with Lua 5.2 or greater due to addition of goto statement
- [ASTERISK-23897] – Change in SETUP ACK handling (checking PI) in revision 413765 breaks working environments
- [ASTERISK-23908] – When using FEC error correction, asterisk tries considers negative sequence numbers as missing
- [ASTERISK-23916] – SIP/SDP fmtp line may include whitespace between attributes
- [ASTERISK-23917] – res_http_websocket: Delay in client processing large streams of data causes disconnect and stuck socket
- [ASTERISK-23921] – refcounter.py uses excessive ram for large refs files
- [ASTERISK-23922] – ao2_container nodes are inconsistent REF_DEBUG
- [ASTERISK-23947] – ActionID missing from AMI PJSIP events (PJSIPShowEndpoints, etc.)
- [ASTERISK-23948] – REF_DEBUG fails to record ao2_ref against objects that were already freed
- [ASTERISK-23984] – Infinite loop possible in ast_careful_fwrite()
- [ASTERISK-24001] – res_rtp_asterisk fails to load module due to undefined symbol ‘dtls_perform_handshake’ when PJPROJECT is not installed
- [ASTERISK-22961] – DTLS-SRTP not working with SHA-256
- [ASTERISK-23492] – Add option to safe_asterisk to disable backgrounding
- [ASTERISK-23552] – http: support persistent connections
- [ASTERISK-23654] – Add ‘pjsip reload’ to default cli_aliases.conf
- [ASTERISK-23811] – Improve performance of Asterisk by reducing the number of channel snapshots created
- [ASTERISK-23939] – ARI: Allow for channel subscriptions on originate
- [ASTERISK-23975] – Description of variables field for userEvent operation missing details.
- [ASTERISK-21443] – New SIP Channel Driver – Create a state provider for dialog-info+xml
- [ASTERISK-23786] – TALK_DETECT: A dialplan function that emits talking start/stop events for AMI/ARI
For a full list of changes in this release candidate, please see the ChangeLog:
Thank you for your continued support of Asterisk!