The Asterisk Development Team has announced the release of Asterisk 13.7.0.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.7.0 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:
- [ASTERISK-7803] – Update the maximum packetization values in frame.c
- [ASTERISK-24106] – WebSockets Automatically decides what driver it will use
- [ASTERISK-24146] – No audio on WebRtc caller side when answer waiting time is more than ~7sec
- [ASTERISK-24543] – Asterisk 13 responds to SIP Invite with all possible codecs configured for peer as opposed to intersection of configured codecs and offered codecs
- [ASTERISK-24779] – Passthrough OPUS codec not working with chan_pjsip
- [ASTERISK-24958] – Forwarding loop detection inhibits certain desirable scenarios
- [ASTERISK-25116] – res_pjsip: Two PeerStatus AMI messages are sent for every status change
- [ASTERISK-25135] – RTP Timeout hangup cause code missing
- [ASTERISK-25160] – Opus Codec: SIP/SDP line fmtp missing when called internally
- [ASTERISK-25165] – Testsuite – Sorcery memory cache leaks
- [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-25404] – segfault/crash in chan_pjsip_hangup … at chan_pjsip.c
- [ASTERISK-25434] – Compiler flags not reported in ‘core show settings’ despite usage during compilation
- [ASTERISK-25435] – Asterisk periodically hangs. UDP Recv-Q greatly exceeds zero.
- [ASTERISK-25438] – res_rtp_asterisk: ICE role message even when ICE is not enabled
- [ASTERISK-25441] – Deadlock in res_sorcery_memory_cache.
- [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-25451] – Broken video – erased rtp marker bit
- [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-25484] – autoframing=yes has no effect
- [ASTERISK-25485] – res_pjsip_outbound_registration: registration stops due to 400 response
- [ASTERISK-25486] – res_pjsip: Fix deadlock when validating URIs
- [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-25505] – res_pjsip_pubsub: Crash on off-nominal when UAS dialog can’t be created
- [ASTERISK-25513] – Crash: malloc failed with high load of subscriptions.
- [ASTERISK-25522] – ARI: Crash when creating channel via ARI originate with requesting channel
- [ASTERISK-25527] – Quirky xmldoc description wrapping
- [ASTERISK-25533] – buffer for ast_format_cap_get_names only 64 bytes
- [ASTERISK-25535] – format creation on module load instead of cache
- [ASTERISK-25537] – format-attribute module: RFC or internal defaults?
- [ASTERISK-25545] – translation module gets cached not joint format
- [ASTERISK-25546] – threadpool: Race condition between idle timeout and activation
- [ASTERISK-25552] – hashtab: Improve NULL tolerance
- [ASTERISK-25561] – app_queue.c line 6503 (try_calling): mutex ‘qe->chan’ freed more times than we’ve locked!
- [ASTERISK-25569] – app_meetme: Audio quality issues
- [ASTERISK-25573] – H.264 format attribute module: resets whole SDP
- [ASTERISK-25575] – res_pjsip: Dynamic outbound registrations created via ARI are not loaded into memory on Asterisk start/restart
- [ASTERISK-25582] – Testsuite: Reactor timeout error in tests/fax/pjsip/directmedia_reinvite_t38
- [ASTERISK-25583] – format-attribute module: RFC 7587 (Opus Codec)
- [ASTERISK-25584] – format-attribute module: VP8 missing
- [ASTERISK-25585] – rasterisk never hits most of main(), but it’s assumed to
- [ASTERISK-25590] – CLI Usage info for ‘pjsip send notify’ references incorrect config
- [ASTERISK-25593] – fastagi: record file closed after sending result
- [ASTERISK-25595] – Unescaped : in messge sent to statsd
- [ASTERISK-25598] – res_pjsip: Contact status messages are printing a hash instead of the uri
- [ASTERISK-25599] – SLIN Resampling Codec only 80 msec
- [ASTERISK-25600] – bridging: Inconsistency in BRIDGEPEER
- [ASTERISK-25601] – json: Audit reference usage and thread safety
- [ASTERISK-25608] – res_pjsip/contacts/statsd: Lifecycle events aren’t consistent
- [ASTERISK-25609] – Asterisk may crash when calling ast_channel_get_t38_state(c)
- [ASTERISK-25610] – Asterisk crash during “sip reload”
- [ASTERISK-25615] – res_pjsip: Setting transport async_operations > 1 causes segfault on tls transports
- [ASTERISK-25616] – Warning with a Codec Module which supports PLC with FEC
- [ASTERISK-25619] – res_chan_stats not sending the correct information to StatsD
- [ASTERISK-25625] – res_sorcery_memory_cache: Add full backend caching
- [ASTERISK-25640] – pbx: Deadlock on features reload and state change hint.
- [ASTERISK-25664] – ast_format_cap_append_by_type leaks a reference
- [ASTERISK-25689] – pjsip show contacts not working in Asterisk 13.7rc2
- [ASTERISK-24718] – Add inital support of “sanitize” to configure
- [ASTERISK-25477] – pjsip show “command” like [criteria]
- [ASTERISK-25518] – taskprocessor: Add high water mark
- [ASTERISK-25571] – PJSIP: Add StatsD stats for some common PJSIP objects
- [ASTERISK-25572] – Endpoints: Add StatsD stats for Asterisk endpoints
- [ASTERISK-25618] – res_pjsip: Check for readability of TLS files at startup
- [ASTERISK-24922] – ARI: Add the ability to intercept hold and raise an event
- [ASTERISK-25419] – Dialplan Application for Integration of StatsD
- [ASTERISK-25549] – Confbridge: Add participant timeout option
For a full list of changes in this release, please see the ChangeLog:
Thank you for your continued support of Asterisk!