Asterisk 12.2.0 Now Available

The Asterisk Development Team has announced the release of Asterisk 12.2.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 12.2.0 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:

Bug

  • [ASTERISK-17523] – Qualify for static realtime peers does not work
  • [ASTERISK-19465] – P-Asserted-Identity Privacy
  • [ASTERISK-19499] – ConfBridge MOH is not working for transferee after attended transfer
  • [ASTERISK-20149] – Crash when faxing SIP to SIP with strictrtp set to yes
  • [ASTERISK-20841] – fromdomain not honored on outbound INVITE request
  • [ASTERISK-21406] – chan_sip deadlock on monlock between unload_module and do_monitor
  • [ASTERISK-21930] – WebRTC over WSS is not working.
  • [ASTERISK-22079] – Segfault: INTERNAL_OBJ (user_data=0x6374652f) at astobj2.c:120
  • [ASTERISK-22738] – “Security denial” error in calls from H323 trunk (ooh323.c)
  • [ASTERISK-22911] – Asterisk fails to resume WebRTC call from hold
  • [ASTERISK-23020] – PJSip – Multihomed machine returning wrong IP address
  • [ASTERISK-23069] – Custom CDR variable not recorded when set in macro called from app_queue
  • [ASTERISK-23073] – Asterisk crashes randomly when using chan_unistim
  • [ASTERISK-23092] – cli: pjsip show endpoint <endpoint> shows allow/disallow codecs the same
  • [ASTERISK-23098] – possible null pointer dereference in format.c
  • [ASTERISK-23103] – Crash in ast_format_cmp, in ao2_find
  • [ASTERISK-23104] – Specifying the SetVar AMI without a Channel cause Asterisk to crash
  • [ASTERISK-23125] – ARI: URI is case sensitive
  • [ASTERISK-23135] – Crash – segfault in ast_channel_hangupcause_set – probably introduced in 11.7.0
  • [ASTERISK-23141] – Asterisk crashes on Dial(), in pbx_find_extension at pbx.c
  • [ASTERISK-23204] – Device state cache requires improvement
  • [ASTERISK-23210] – Security: Remote crash in res_pjsip.
  • [ASTERISK-23231] – Since 405693 If we have res_fax.conf file set to minrate=2400, then res_fax refuse to load
  • [ASTERISK-23235] – pjsip transport/tos interpreted differently than endpoint/tos_audio
  • [ASTERISK-23254] – Bad ao2_find() usage in pjsip_options.c
  • [ASTERISK-23258] – Target_uri for LiveRecording model in ARI
  • [ASTERISK-23261] – Output mixup in ${CHANNEL(rtpqos,audio,all)}
  • [ASTERISK-23265] – Preloading Certain Modules in Asterisk 12 causes a core dump
  • [ASTERISK-23266] – pjsip_cli: Memory leak in ast_sip_cli_print_sorcery_objectset
  • [ASTERISK-23279] – Asterisk doesn’t support the dynamic payload change in rtp mapping in the 200 OK response
  • [ASTERISK-23287] – res_pjsip_refer: Crash during attended transfer when attended->transferer_second channel is NULL
  • [ASTERISK-23290] – chan_sip: ast_bridge_transfer_blind causes channel to be hung up immediately, leading to BYE request being sent before NOTIFY
  • [ASTERISK-23295] – ARI: ChannelEnteredBridge event not delivered to client during bridge move operation
  • [ASTERISK-23297] – Asterisk 12, pbx_config.so segfaults if res_parking.so is not loaded, or if res_parking.conf has no configuration
  • [ASTERISK-23311] – Manager – MoH Stop Event fails to show up when leaving Conference
  • [ASTERISK-23320] – Preloading pbx_config.so with a CustomPresence hint defined results in crash
  • [ASTERISK-23323] – chan_sip: missing p->owner checks in handle_response_invite
  • [ASTERISK-23336] – Asterisk warning “Don’t know how to indicate condition 33 on ooh323c” on outgoing calls from H323 to SIP peer
  • [ASTERISK-23340] – Security Vulnerability: stack allocation of cookie headers in loop allows for unauthenticated remote denial of service attack
  • [ASTERISK-23373] – Security: Open FD exhaustion with chan_sip Session-Timers
  • [ASTERISK-23383] – Wrong sense test on stat return code causes unchanged config check to break with include files.
  • [ASTERISK-23391] – Audit dialplan function usage of channel variable
  • [ASTERISK-23406] – Fix typo in “sip show peer”
  • [ASTERISK-23420] – Memory leak in manager_add_filter function in manager.c
  • [ASTERISK-23444] – Playback and Record events not subscribed to when calling Play/Record on bridge
  • [ASTERISK-23460] – ooh323 channel stuck if call is placed directly and gatekeeper is not available
  • [ASTERISK-23461] – Only first user is muted when joining confbridge with ‘startmuted=yes’
  • [ASTERISK-23487] – features.conf cant load from realtime because features_config.c starts before loader.c
  • [ASTERISK-23488] – Logic error in callerid checksum processing
  • [ASTERISK-23509] – SayNumber for Polish language tries to play empty files for numbers divisible by 100
  • [ASTERISK-23548] – POST to ARI sometimes returns no body on success

Improvement

  • [ASTERISK-22008] – Config framework docs should display the see-also information in CLI output.
  • [ASTERISK-22499] – ARI documentation – point to HTTP server configuration sample and wiki docs where appropriate
  • [ASTERISK-22537] – Create Sorcery equivalent to the AST_CONFIG function
  • [ASTERISK-22661] – Unable to exit ChanSpy if spied channel does not have a call in progress
  • [ASTERISK-23099] – WSS: enable ast_websocket_read() function to read the whole available data at first and then wait for any fragmented packets
  • [ASTERISK-23120] – ARI/AMI: allow objects created via interfaces to have their unique ID specified by the external application
  • [ASTERISK-23233] – alembic missing scripts for certain realtime tables
  • [ASTERISK-23275] – CLI command ‘pjsip show registrations’ missing
  • [ASTERISK-23435] – PJSIP: Fix the DNS resolution (whoops)
  • [ASTERISK-23437] – ARI: Add the ability to add properties to a bridge on creation

New Feature

  • [ASTERISK-23276] – Look at adding the ‘pjsip show channel’ command
  • [ASTERISK-23557] – HEP/PJSIP: Add modules to support integrating Homer with PJSIP

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-12.2.0

Thank you for your continued support of Asterisk!

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