Asterisk 13.7.0-rc1 Now Available

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.

Thank you!

The following are the issues resolved in this release candidate:

Bug

  • [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

Improvement

New Feature

  • [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:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.7.0-rc1

Thank you for your continued support of Asterisk!

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