Asterisk 12.2.0-rc1 Now Available

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

The release of Asterisk 12.2.0-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-17523] – Qualify for static realtime peers does not work
• [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-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 candidate, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-12.2.0-rc1

Thank you for your continued support of Asterisk!