Asterisk News

Asterisk Releases

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!


Asterisk 1.8.27.0-rc1 Now Available

Mar 28, 2014

The Asterisk Development Team has announced the first 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-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-20841] - fromdomain not honored on outbound INVITE request
  • [ASTERISK-21406] - [patch] chan_sip deadlock on monlock between unload_module and do_monitor
  • [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-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-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-23100] - [patch] In chan_mgcp the ident in transmitted request and request queue may differ - fix for locking
  • [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-23260] - [patch]ForkCDR v option does not keep CDR variables for subsequent records
  • [ASTERISK-23261] - [patch]Output mixup in ${CHANNEL(rtpqos,audio,all)}
  • [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-23323] - [patch]chan_sip: missing p->owner checks in handle_response_invite
  • [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-23382] - [patch]Build System: make -qp can corrupt menuselect-tree and related files
  • [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-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

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-rc1

Thank you for your continued support of Asterisk!


Security Release: Asterisk 1.8.15-cert5, 1.8.26.1, 11.6-cert2, 11.8.1, 12.1.1 Now Available

Mar 10, 2014

The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.15, 11.6, and Asterisk 1.8, 11, and 12. The available security releases are released as versions 1.8.15-cert5, 11.6-cert2, 1.8.26.1, 11.8.1, and 12.1.1.

These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases

The release of these versions resolve the following issues:

  • AST-2014-001: Stack overflow in HTTP processing of Cookie headers.
    Sending a HTTP request that is handled by Asterisk with a large number of Cookie headers could overflow the stack. Another vulnerability along similar lines is any HTTP request with a ridiculous number of headers in the request could exhaust system memory.
  • AST-2014-002: chan_sip: Exit early on bad session timers request.
    This change allows chan_sip to avoid creation of the channel and consumption of associated file descriptors altogether if the inbound request is going to be rejected anyway.

Additionally, the release of 12.1.1 resolves the following issue:

  • AST-2014-003: res_pjsip: When handling 401/407 responses don't assume a request will have an endpoint.
    This change removes the assumption that an outgoing request will always have an endpoint and makes the authenticate_qualify option work once again.

Finally, a security advisory, AST-2014-004, was released for a vulnerability fixed in Asterisk 12.1.0. Users of Asterisk 12.0.0 are encouraged to upgrade to 12.1.1 to resolve both vulnerabilities. These issues and their resolutions are described in the security advisories.

For more information about the details of these vulnerabilities, please read security advisories AST-2014-001, AST-2014-002, AST-2014-003, and AST-2014-004, which were released at the same time as this announcement.

For a full list of changes in the current releases, please see the ChangeLogs:

The security advisories are available at:

Thank you for your continued support of Asterisk!


Asterisk 12.1.0 Now Available

Mar 3, 2014

The Asterisk Development Team has announced the release of Asterisk 12.1.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 12.1.0 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:

Bugs

  • [ASTERISK-17138] - [patch] Asterisk not re-registering after it receives "Forbidden - wrong password on authentication"
  • [ASTERISK-17727] - [patch] TLS doesn't get all certificate chain
  • [ASTERISK-17837] - extconfig.conf - Maximum Include level (1) exceeded
  • [ASTERISK-19773] - Asterisk crash on issuing Asterisk-CLI 'reload' command multiple times on cli_aliases
  • [ASTERISK-22486] - ARI: TCP Reset after 204 response
  • [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-22854] - [patch] - Deadlock between cel_pgsql unload and core_event_dispatcher taskprocessor thread
  • [ASTERISK-22861] - [patch]Specifying a null time as parameter to GotoIfTime or ExecIfTime causes segmentation fault
  • [ASTERISK-22871] - cel_pgsql module not loading after "reload" or "reload cel_pgsql.so" command
  • [ASTERISK-22884] - hangup_handler end with h extension: tests currently fail in Asterisk 12 +
  • [ASTERISK-22910] - [patch] - REPLACE() calls strcpy on overlapping memory when <replace-char> is empty
  • [ASTERISK-22924] - PJSIP MESSAGE support does not present the contact information on outbound messages
  • [ASTERISK-22946] - Local From tag regression with sipgate.de
  • [ASTERISK-22952] - res_pjsip_pubsub: crash when subscription_destructor is terminated from a non-PJSIP thread
  • [ASTERISK-22962] - performance spike on Local channels originated using ARI
  • [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-23011] - [patch]configure.ac and pbx_lua don't support lua 5.2
  • [ASTERISK-23018] - PJSip 'allow=all' results in failed SDP negotiation
  • [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-23051] - ARI: channel variables in JSON breaks passing parameters in JSON
  • [ASTERISK-23053] - The users of ao2_iterator_cleanup() are violating the ao2_iterator opacity.
  • [ASTERISK-23056] - [patch]INFINITY and NAN undefined
  • [ASTERISK-23061] - [Patch] 'textsupport' setting not mentioned in sip.conf.sample
  • [ASTERISK-23062] - res_pjsip AOR config option qualify_frequency is inconsistently respected
  • [ASTERISK-23065] - On Asterisk start, device state is INVALID for previously registered PJSIP endpoints, despite re-registrations
  • [ASTERISK-23071] - pjsip: mailboxes documentation is lacking
  • [ASTERISK-23072] - MWI subscription from Cisco SPA fails with PJSIP
  • [ASTERISK-23074] - Crash in ChanIsAvail app
  • [ASTERISK-23081] - PJSip Tab Expansion erroring
  • [ASTERISK-23082] - Including g722 in pjsip codec configuration results in unexpected SDP offers
  • [ASTERISK-23084] - [patch]rasterisk needlessly prints the AST-2013-007 warning
  • [ASTERISK-23100] - [patch] In chan_mgcp the ident in transmitted request and request queue may differ - fix for locking
  • [ASTERISK-23101] - pjsip: crash when parsing scheme from SIP URI
  • [ASTERISK-23106] - pjsip: ACK to 200 OK sent to private IP address on outbound channel's INVITE request
  • [ASTERISK-23128] - res_ari: Memory leak on response headers and some JSON response messages
  • [ASTERISK-23129] - segfault in res_pjsip_pubsub.so
  • [ASTERISK-23134] - [patch] res_rtp_asterisk port selection cannot handle selinux port restrictions
  • [ASTERISK-23143] - ARI: subscribing to an already subscribed resource returns a 500 error
  • [ASTERISK-23164] - CDRs: mid-call/pre-dial handlers perturb context/exten/app/data fields during Dial
  • [ASTERISK-23168] - Overriding outbound_auth in a pjsip registration causes ERROR, assert failure.
  • [ASTERISK-23177] - [patch] RealTime cant update sipbuddies table when registering or updating friend
  • [ASTERISK-23178] - devicestate.h: device state setting functions are documented with the wrong return values
  • [ASTERISK-23213] - SIP over WS: Audio problems when upgrading to 11.8 from 11.7 with endpoints behind NAT
  • [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-23249] - Skinny subchannel locking issues
  • [ASTERISK-23250] - CDR(start) function is broken due to sizeof dereference

Improvements

New Features

  • [ASTERISK-23038] - Need config option to enable PJSIP logger at load time

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-12.1.0

Thank you for your continued support of Asterisk!


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