Asterisk News

Asterisk 1.8.27.0-rc2 Now Available

Apr 22, 2014

The Asterisk Development Team has announced the second release candidate of Asterisk 1.8.27.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 1.8.27.0-rc2 resolves an issue reported by the community and would have not been possible without your participation.

Thank you!

The following is the issue resolved in this release candidate:

  • chan_sip: Add sendrpid trust options
    (Closes issue ASTERISK-19465. Reported by Krzysztof Chmielewski)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.27.0-rc2

Thank you for your continued support of Asterisk!


Asterisk 12.2.0-rc2 Now Available

Apr 14, 2014

The Asterisk Development Team has announced the second 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-rc2 resolves an issue reported by the community and would have not been possible without your participation.

Thank you!

The following is the issue resolved in this release candidate:

  • autoservice: fix reference leak of logger callid.
    (Closes issue ASTERISK-23616. Reported by ibercom)

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-rc2

Thank you for your continued support of Asterisk!


Asterisk 11.9.0-rc2 Now Available

Apr 14, 2014

The Asterisk Development Team has announced the second 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-rc2 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:

  • app_voicemail: fix missing symbol
    (Closes issue ASTERISK-23559. Reported by Corey Farrell)
  • autoservice: fix reference leak of logger callid.
    (Closes issue ASTERISK-23616. Reported by ibercom)

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-rc2

Thank you for your continued support of Asterisk!


Asterisk 12.2.0-rc1 Now Available

Mar 28, 2014

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] - [patch] chan_sip deadlock on monlock between unload_module and do_monitor
  • [ASTERISK-21930] - [patch]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] - [patch]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] - [patch]possible null pointer dereference in format.c
  • [ASTERISK-23103] - [patch]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] - [patch]Output mixup in ${CHANNEL(rtpqos,audio,all)}
  • [ASTERISK-23265] - Preloading Certain Modules in Asterisk 12 causes a core dump
  • [ASTERISK-23266] - [patch]pjsip_cli: Memory leak in ast_sip_cli_print_sorcery_objectset
  • [ASTERISK-23279] - [patch]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] - [patch]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] - [patch]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] - [patch]Fix typo in "sip show peer"
  • [ASTERISK-23420] - [patch]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] - [patch]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] - [patch] 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!


Asterisk 11.9.0-rc1 Now Available

Mar 28, 2014

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] - [patch] 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] - [patch] chan_sip deadlock on monlock between unload_module and do_monitor
  • [ASTERISK-21930] - [patch]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] - [patch]Specifying a null time as parameter to GotoIfTime or ExecIfTime causes segmentation fault
  • [ASTERISK-22911] - [patch]Asterisk fails to resume WebRTC call from hold
  • [ASTERISK-22988] - [patch]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] - [patch] Spelling typo "transfered" instead of "transferred"
  • [ASTERISK-23028] - [patch] Asterisk man pages contains unquoted minus signs
  • [ASTERISK-23034] - [patch] 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] - [patch]possible null pointer dereference in format.c
  • [ASTERISK-23100] - [patch] In chan_mgcp the ident in transmitted request and request queue may differ - fix for locking
  • [ASTERISK-23103] - [patch]Crash in ast_format_cmp, in ao2_find
  • [ASTERISK-23104] - Specifying the SetVar AMI without a Channel cause Asterisk to crash
  • [ASTERISK-23134] - [patch] 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] - [patch]ForkCDR v option does not keep CDR variables for subsequent records
  • [ASTERISK-23261] - [patch]Output mixup in ${CHANNEL(rtpqos,audio,all)}
  • [ASTERISK-23279] - [patch]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] - [patch]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] - [patch]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] - [patch]Fix typo in "sip show peer"
  • [ASTERISK-23420] - [patch]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] - [patch]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] - [patch]Allow building cdr_radius and cel_radius against libfreeradius-client
  • [ASTERISK-23099] - [patch] 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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