Asterisk 11.9.0-rc1 Now Available

The Asterisk Development Team has announced the first release candidate of Asterisk 11.9.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 11.9.0-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-17727] – TLS doesn’t get all certificate chain
  • [ASTERISK-17837] – extconfig.conf – Maximum Include level (1) exceeded
  • [ASTERISK-19499] – ConfBridge MOH is not working for transferee after attended transfer
  • [ASTERISK-19773] – Asterisk crash on issuing Asterisk-CLI ‘reload’ command multiple times on cli_aliases
  • [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-22662] – Documentation fix? – queues.conf says persistentmembers defaults to yes, it appears to lie
  • [ASTERISK-22757] – segfault in res_clialiases.so on reload when mapping “module reload” command
  • [ASTERISK-22790] – check_modem_rate() may return incorrect rate for V.27
  • [ASTERISK-22861] – Specifying a null time as parameter to GotoIfTime or ExecIfTime causes segmentation fault
  • [ASTERISK-22911] – Asterisk fails to resume WebRTC call from hold
  • [ASTERISK-22988] – T38 , SIP 488 after Rejecting image media offer due to invalid or unsupported syntax
  • [ASTERISK-23008] – Local channels loose CALLERID name when DAHDI channel connects
  • [ASTERISK-23027] – Spelling typo “transfered” instead of “transferred”
  • [ASTERISK-23028] – Asterisk man pages contains unquoted minus signs
  • [ASTERISK-23034] – manager Originate doesn’t abort on failed format_cap allocation
  • [ASTERISK-23046] – Custom CDR fields set during a GoSUB called from app_queue are not inserted
  • [ASTERISK-23061] – [Patch] ‘textsupport’ setting not mentioned in sip.conf.sample
  • [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-23098] – possible null pointer dereference in format.c
  • [ASTERISK-23100] – In chan_mgcp the ident in transmitted request and request queue may differ – fix for locking
  • [ASTERISK-23103] – Crash in ast_format_cmp, in ao2_find
  • [ASTERISK-23104] – Specifying the SetVar AMI without a Channel cause Asterisk to crash
  • [ASTERISK-23134] – res_rtp_asterisk port selection cannot handle selinux port restrictions
  • [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-23178] – devicestate.h: device state setting functions are documented with the wrong return values
  • [ASTERISK-23220] – STACK_PEEK function with no arguments causes crash/core dump
  • [ASTERISK-23231] – Since 405693 If we have res_fax.conf file set to minrate=2400, then res_fax refuse to load
  • [ASTERISK-23232] – LocalBridge AMI Event LocalOptimization value is opposite to what’s expected
  • [ASTERISK-23255] – UUID included for Redhat, but missing for Debian distros in install_prereq script
  • [ASTERISK-23260] – ForkCDR v option does not keep CDR variables for subsequent records
  • [ASTERISK-23261] – Output mixup in ${CHANNEL(rtpqos,audio,all)}
  • [ASTERISK-23279] – Asterisk doesn’t support the dynamic payload change in rtp mapping in the 200 OK response
  • [ASTERISK-23297] – Asterisk 12, pbx_config.so segfaults if res_parking.so is not loaded, or if res_parking.conf has no configuration
  • [ASTERISK-23310] – bridged channel crashes in bridge_p2p_rtp_write
  • [ASTERISK-23311] – Manager – MoH Stop Event fails to show up when leaving Conference
  • [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-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-22661] – Unable to exit ChanSpy if spied channel does not have a call in progress
  • [ASTERISK-22980] – Allow building cdr_radius and cel_radius against libfreeradius-client
  • [ASTERISK-23099] – WSS: enable ast_websocket_read() function to read the whole available data at first and then wait for any fragmented packets

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.9.0-rc1

Thank you for your continued support of Asterisk!

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