The Asterisk Development Team has announced the release of Asterisk 12.4.0. This release is available for immediate
download at http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 12.4.0 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 candidate:
Bug
- [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
Improvement
- [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.
New Feature
- [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, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-12.4.0
Thank you for your continued support of Asterisk!