Asterisk 11.23.0-rc1 Now Available

The Asterisk Development Team has announced the first release candidate of Asterisk 11.23.0.

This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 11.23.0-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: Release Notes – Asterisk – Version 13.10.0

Bug

  • [ASTERISK-16115] – problem with ringinuse=no, queue members receive sometimes two calls
  • [ASTERISK-24436] – Missing header in res/res_srtp.c when compiling against libsrtp-1.5.0
  • [ASTERISK-24463] – Voicemail email address corrupt or not sent when message is in the process of being recorded during reload
  • [ASTERISK-24986] – keepalive INFO packages ignored by asterisk
  • [ASTERISK-25262] – Memory leak when a caller channel does multiple dials and CEL is enabled
  • [ASTERISK-25352] – res_hep_rtcp correlation_id is different then res_hep
  • [ASTERISK-25777] – data race in threadpool
  • [ASTERISK-25826] – PJSIP / Sorcery slow load from realtime
  • [ASTERISK-25917] – app_voicemail: passwordlocation=spooldir only works if you manually add secret.conf yourself
  • [ASTERISK-25938] – res_odbc: MySQL/MariaDB statement LAST_INSERT_ID() always returns zero.
  • [ASTERISK-25941] – chan_pjsip: Crash on an immediate SIP final response
  • [ASTERISK-25950] – SIP channel does not send PeerStatus events for autocreated peers
  • [ASTERISK-25954] – Manager QueueSummary and QueueStatus Actions are case sensitive to QueueName
  • [ASTERISK-25956] – Compilation error in conditionally compiled code in config_options.c
  • [ASTERISK-25961] – tests/channels/SIP/sip_tls_call: Sporadic crash when running test
  • [ASTERISK-25963] – func_odbc requires reconnect checks for stale connections
  • [ASTERISK-25964] – Outbound registrations created via ARI/push configuration do not clean up outbound registrations currently in flight
  • [ASTERISK-25968] – pjproject_bundled: Configure and make need to be re-tested
  • [ASTERISK-25970] – Segfault in pjsip_url_compare
  • [ASTERISK-25978] – res_pjsip_authenticator_digest: Should not use source port in nonce verification
  • [ASTERISK-25990] – PJSIP TLS registration should respect client_uri scheme when generating Contact URI
  • [ASTERISK-25993] – pjproject: Allow bundling to not require everything it does
  • [ASTERISK-25998] – file: Crash when using nativeformats
  • [ASTERISK-26005] – res_pjsip: Multiple SIP messages are combined into 1 TCP packet
  • [ASTERISK-26007] – res_pjsip: Endpoints deleting early after upgrade from 13.8.2 to 13.9
  • [ASTERISK-26008] – app_followme does not delete recorded name prompt
  • [ASTERISK-26014] – res_sorcery_astdb: Make tolerant of unknown fields
  • [ASTERISK-26029] – parking: ast_parking_park_call should return parking_space instead of parking_exten
  • [ASTERISK-26030] – call cut because of double Session-Expires header in re-invite after proxy authentication is required
  • [ASTERISK-26034] – T.38 passthrough problem behind firewall due to early nosignal packet
  • [ASTERISK-26038] – ‘make install’ doesn’t seem to install OS/X init files
  • [ASTERISK-26054] – Asterisk crashes (core dump)
  • [ASTERISK-26063] – ${PJSIP_HEADER(read,Call-ID)} does not work – documentation needs clarification for when read/write is possible
  • [ASTERISK-26065] – chan_pjsip: MWI NOTIFY contents not ordered properly
  • [ASTERISK-26069] – Asterisk truncates To: header, dropping the closing ‘>’
  • [ASTERISK-26070] – ari/channels: Creating a local channel without an originator adds all audio formats to it’s capabilities
  • [ASTERISK-26074] – res_odbc: Deadlock within UnixODBC
  • [ASTERISK-26078] – core: Memory leak in logging
  • [ASTERISK-26083] – ARI: Announcer channels staying around after playback to a bridge is finished
  • [ASTERISK-26089] – Invalid security events during boot using PJSIP Realtime
  • [ASTERISK-26091] – ar cru creates warning, instead use ar cr
  • [ASTERISK-26092] – [Segfault] in res_rtp_asterisk.c:4268 after Remotely bridged channels
  • [ASTERISK-26096] – res_hep: Crash when configuration file is missing
  • [ASTERISK-26097] – CLI: show maximum file descriptors
  • [ASTERISK-26099] – res_pjsip_pubsub: Crash when sending request due to server timeout
  • [ASTERISK-26126] – leverage ‘bindaddr’ for TLS in http.conf
  • [ASTERISK-26127] – res_pjsip_session: Crash due to race condition between res_pjsip_session unload and timer
  • [ASTERISK-26128] – Alembic scripts are failing
  • [ASTERISK-26130] – WebRTC: Should use latest DTLS version.
  • [ASTERISK-26138] – chan_unistim: Under FreeBSD, chan_unistim generates a compile error
  • [ASTERISK-26139] – test_res_pjsip_scheduler: Compile failure if pjproject isn’t installed in a system location
  • [ASTERISK-26140] – res_rtp_asterisk: gcc 6 caught a self-comparison
  • [ASTERISK-26141] – res_fax: fax_v21_session_new leaks reference to v21_details

Improvement

  • [ASTERISK-25835] – Authentication using ‘Username’ field from Digest
  • [ASTERISK-25930] – PJSIP: disable multi domain to improve realtime performace
  • [ASTERISK-25994] – res_pjsip: module load priority
  • [ASTERISK-26088] – Investigate heavy memory utilization by res_pjsip_pubsub

For a full list of changes in this release candidate, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0-rc1

Thank you for your continued support of Asterisk!

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