The Asterisk Development Team has announced the release of Asterisk 13.7.0-rc1.
This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.7.0-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-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-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-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-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 candidate, please see the ChangeLog:
Thank you for your continued support of Asterisk!