Asterisk 11.9.0 Now Available

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

The release of Asterisk 11.9.0 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-17727] – TLS doesn’t get all certificate chain
  • [ASTERISK-17837] – extconfig.conf – Maximum Include level (1) exceeded
  • [ASTERISK-19465] – P-Asserted-Identity Privacy
  • [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
  • [ASTERISK-23559] – app_voicemail fails to load after fix to dialplan functions
  • [ASTERISK-23616] – Big memory leak in logger.c

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, please see the ChangeLog:

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

Thank you for your continued support of Asterisk!

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