Asterisk News

Asterisk Releases

Asterisk 13.3.1 Now Available

Apr 6, 2015

The Asterisk Development Team has announced the release of Asterisk 13.3.1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.3.1 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:

  • pjsip: resolve ABI compatibility problem with external modules
    (Closes issue ASTERISK-24941. Reported by Matt Jordan)

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

 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.1

Thank you for your continued support of Asterisk!

 


Asterisk 13.2.1 Now Available

Apr 6, 2015

The Asterisk Development Team has announced the release of Asterisk 13.2.1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.2.1 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:

  • pjsip: resolve ABI compatibility problem with external modules
    (Closes issue ASTERISK-24941. Reported by Matt Jordan)

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

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.2.1

Thank you for your continued support of Asterisk!


Asterisk 13.3.0 Now Available

Apr 1, 2015

The Asterisk Development Team has announced the release of Asterisk 13.3.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.3.0 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-15434] - [patch] When ast_pbx_start failed, both an error response and BYE are sent to the caller
  • [ASTERISK-16779] - Cannot disallow unknown format ''
  • [ASTERISK-17721] - Incoming SRTP calls that specify a key lifetime fail
  • [ASTERISK-18105] - most of asterisk modules are unbuildable in cygwin environment
  • [ASTERISK-18708] - func_curl hangs channel under load
  • [ASTERISK-19470] - Documentation on app_amd is incorrect
  • [ASTERISK-20850] - [patch]Nested functions aren't portable. Adapting RAII_VAR to use clang/llvm blocks to get the same/similar functionality.
  • [ASTERISK-21038] - Bad command completion of "core set debug channel"
  • [ASTERISK-22670] - Asterisk crashes when processing ISDN AoC Events
  • [ASTERISK-23214] - chan_sip WARNING message 'We are requesting SRTP for audio, but they responded without it' is ambiguous and wrong in some cases
  • [ASTERISK-23390] - NewExten Event with application AGI shows up before and after AGI runs
  • [ASTERISK-24015] - app_transfer fails with PJSIP channels
  • [ASTERISK-24085] - Documentation - We should remove or further document the 'contact' section in pjsip.conf
  • [ASTERISK-24451] - chan_iax2: reference leak in sched_delay_remove
  • [ASTERISK-24479] - Enable REF_DEBUG for module references
  • [ASTERISK-24499] - Need more explicit debug when PJSIP dialstring is invalid
  • [ASTERISK-24612] - res_pjsip: No information if a required sorcery wizard is not loaded
  • [ASTERISK-24616] - Crash in res_format_attr_h264 due to invalid string copy
  • [ASTERISK-24632] - install_prereq script installs pjproject without IPv6 support
  • [ASTERISK-24677] - ARI GET variable on channel provides unhelpful response on non-existent variable
  • [ASTERISK-24685] - "pjsip show version" CLI command
  • [ASTERISK-24689] - Segfault on hangup after outgoing PRI-Euroisdn call
  • [ASTERISK-24700] - CRASH: NULL channel is being passed to ast_bridge_transfer_attended()
  • [ASTERISK-24701] - Stasis: Write timeout on WebSocket fails to fully disconnect underlying socket, leading to events being dropped with no additional information
  • [ASTERISK-24716] - Improve pjsip log messages for presence subscription failure
  • [ASTERISK-24724] - 'httpstatus' Web Page Produces Incomplete HTML
  • [ASTERISK-24727] - PJSIP: Crash experienced during multi-Asterisk transfer scenario.
  • [ASTERISK-24739] - [patch] - Out of files -- call fails -- numerous files with inodes from under /usr/share/zoneinfo, mostly posixrules
  • [ASTERISK-24740] - [patch]Segmentation fault on aoc-e event
  • [ASTERISK-24741] - dtls_handler causes Asterisk to crash
  • [ASTERISK-24742] - [patch] Fix ast_odbc_find_table function in res_odbc
  • [ASTERISK-24748] - res_pjsip: If wizards explicitly configured in sorcery.conf false ERROR messages may occur
  • [ASTERISK-24751] - Integer values in json payload to ARI cause asterisk to crash
  • [ASTERISK-24752] - Crash in bridge_manager_service_req when bridge is destroyed by ARI during shutdown
  • [ASTERISK-24755] - Asterisk sends unexpected early BYE to transferrer during attended transfer when using a Stasis bridge
  • [ASTERISK-24768] - res_timing_pthread: file descriptor leak
  • [ASTERISK-24769] - res_pjsip_sdp_rtp: Local ICE candidates leaked
  • [ASTERISK-24771] - ${CHANNEL(pjsip)} - segfault
  • [ASTERISK-24772] - ODBC error in realtime sippeers when device unregisters under MariaDB
  • [ASTERISK-24785] - 'Expires' header missing from 200 OK on REGISTER
  • [ASTERISK-24786] - [patch] - Asterisk terminates when playing a voicemail stored in LDAP
  • [ASTERISK-24787] - [patch] - Microsoft exchange incompatibility for playing back messages stored in IMAP - play_message: No origtime
  • [ASTERISK-24791] - Crash in ast_rtcp_write_report
  • [ASTERISK-24796] - Codecs and bucket schema's prevent module unload
  • [ASTERISK-24797] - bridge_softmix: G.729 codec license held
  • [ASTERISK-24799] - [patch] make fails with undefined reference to SSLv3_client_method
  • [ASTERISK-24800] - Crash in __sip_reliable_xmit due to invalid thread ID being passed to pthread_kill
  • [ASTERISK-24807] - Missing mandatory field Max-Forwards
  • [ASTERISK-24808] - res_config_odbc: Improper escaping of backslashes occurs with MySQL
  • [ASTERISK-24812] - ARI: Creating channels through /channels resource always uses SLIN, which results in unneeded transcoding
  • [ASTERISK-24814] - asterisk/lock.h: Fix syntax errors for non-gcc OSX with 64 bit integers
  • [ASTERISK-24817] - init_logger_chain: unreachable code block
  • [ASTERISK-24825] - Caller ID not recognized using Centrex/Distinctive dialing
  • [ASTERISK-24828] - Fix Frame Leaks
  • [ASTERISK-24830] - res_rtp_asterisk.c checks USE_PJPROJECT not HAVE_PJPROJECT
  • [ASTERISK-24838] - chan_sip: Locking inversion occurs when building a peer causes a peer poke during request handling
  • [ASTERISK-24840] - res_pjsip: conflicting endpoint identifiers
  • [ASTERISK-24872] - [patch] AMI PJSIPShowEndpoint closes AMI connection on error
  • [ASTERISK-24876] - Investigate reference leaks from tests/channels/local/local_optimize_away
  • [ASTERISK-24879] - [patch]Compilation fails due to 64bit time under OpenBSD
  • [ASTERISK-24880] - [patch]Compilation under OpenBSD
  • [ASTERISK-24882] - chan_sip: Improve usage of REF_DEBUG

Improvement

  • [ASTERISK-24745] - [patch]Add no_answer to ARI hangup causes
  • [ASTERISK-24790] - Reduce spurious noise in logs from voicemail - Couldn't find mailbox %s in context
  • [ASTERISK-24811] - asterisk-publication sorcery object does not use realtime

New Feature

  • [ASTERISK-17899] - Handle crypto lifetime in SDES-SRTP negotiation
  • [ASTERISK-24703] - ARI: Add the ability to "transfer" (redirect) a channel


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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.3.0

Thank you for your continued support of Asterisk!

 

 

 

 


Asterisk 11.17.0 Now Available

Apr 1, 2015

The Asterisk Development Team has announced the release of Asterisk 11.17.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 11.17.0 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-15434] - [patch] When ast_pbx_start failed, both an error response and BYE are sent to the caller
  • [ASTERISK-16779] - Cannot disallow unknown format ''
  • [ASTERISK-17721] - Incoming SRTP calls that specify a key lifetime fail
  • [ASTERISK-18105] - most of asterisk modules are unbuildable in cygwin environment
  • [ASTERISK-18708] - func_curl hangs channel under load
  • [ASTERISK-19470] - Documentation on app_amd is incorrect
  • [ASTERISK-20850] - [patch]Nested functions aren't portable. Adapting RAII_VAR to use clang/llvm blocks to get the same/similar functionality.
  • [ASTERISK-21038] - Bad command completion of "core set debug channel"
  • [ASTERISK-22436] - [patch] No BYE to masqueraded channel on INVITE with replaces
  • [ASTERISK-23214] - chan_sip WARNING message 'We are requesting SRTP for audio, but they responded without it' is ambiguous and wrong in some cases
  • [ASTERISK-23390] - NewExten Event with application AGI shows up before and after AGI runs
  • [ASTERISK-24451] - chan_iax2: reference leak in sched_delay_remove
  • [ASTERISK-24479] - Enable REF_DEBUG for module references
  • [ASTERISK-24701] - Stasis: Write timeout on WebSocket fails to fully disconnect underlying socket, leading to events being dropped with no additional information
  • [ASTERISK-24724] - 'httpstatus' Web Page Produces Incomplete HTML
  • [ASTERISK-24739] - [patch] - Out of files -- call fails -- numerous files with inodes from under /usr/share/zoneinfo, mostly posixrules
  • [ASTERISK-24742] - [patch] Fix ast_odbc_find_table function in res_odbc
  • [ASTERISK-24772] - ODBC error in realtime sippeers when device unregisters under MariaDB
  • [ASTERISK-24786] - [patch] - Asterisk terminates when playing a voicemail stored in LDAP
  • [ASTERISK-24787] - [patch] - Microsoft exchange incompatibility for playing back messages stored in IMAP - play_message: No origtime
  • [ASTERISK-24796] - Codecs and bucket schema's prevent module unload
  • [ASTERISK-24797] - bridge_softmix: G.729 codec license held
  • [ASTERISK-24799] - [patch] make fails with undefined reference to SSLv3_client_method
  • [ASTERISK-24800] - Crash in __sip_reliable_xmit due to invalid thread ID being passed to pthread_kill
  • [ASTERISK-24808] - res_config_odbc: Improper escaping of backslashes occurs with MySQL
  • [ASTERISK-24814] - asterisk/lock.h: Fix syntax errors for non-gcc OSX with 64 bit integers
  • [ASTERISK-24817] - init_logger_chain: unreachable code block
  • [ASTERISK-24825] - Caller ID not recognized using Centrex/Distinctive dialing
  • [ASTERISK-24828] - Fix Frame Leaks
  • [ASTERISK-24838] - chan_sip: Locking inversion occurs when building a peer causes a peer poke during request handling
  • [ASTERISK-24876] - Investigate reference leaks from tests/channels/local/local_optimize_away
  • [ASTERISK-24879] - [patch]Compilation fails due to 64bit time under OpenBSD
  • [ASTERISK-24880] - [patch]Compilation under OpenBSD

Improvement

  • [ASTERISK-24790] - Reduce spurious noise in logs from voicemail - Couldn't find mailbox %s in context

New Feature

  • [ASTERISK-17899] - [patch] Handle crypto lifetime in SDES-SRTP negotiation

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0

Thank you for your continued support of Asterisk!


Asterisk 13.3.0-rc1 Now Available

Mar 23, 2015

The Asterisk Development Team has announced the first release candidate of Asterisk 13.3.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.3.0-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-15434] - [patch] When ast_pbx_start failed, both an error response and BYE are sent to the caller
  • [ASTERISK-16779] - Cannot disallow unknown format ''
  • [ASTERISK-17721] - Incoming SRTP calls that specify a key lifetime fail
  • [ASTERISK-18105] - most of asterisk modules are unbuildable in cygwin environment
  • [ASTERISK-18708] - func_curl hangs channel under load
  • [ASTERISK-19470] - Documentation on app_amd is incorrect
  • [ASTERISK-20850] - [patch]Nested functions aren't portable. Adapting RAII_VAR to use clang/llvm blocks to get the same/similar functionality.
  • [ASTERISK-21038] - Bad command completion of "core set debug channel"
  • [ASTERISK-22670] - Asterisk crashes when processing ISDN AoC Events
  • [ASTERISK-23214] - chan_sip WARNING message 'We are requesting SRTP for audio, but they responded without it' is ambiguous and wrong in some cases
  • [ASTERISK-23390] - NewExten Event with application AGI shows up before and after AGI runs
  • [ASTERISK-24015] - app_transfer fails with PJSIP channels
  • [ASTERISK-24085] - Documentation - We should remove or further document the 'contact' section in pjsip.conf
  • [ASTERISK-24451] - chan_iax2: reference leak in sched_delay_remove
  • [ASTERISK-24479] - Enable REF_DEBUG for module references
  • [ASTERISK-24499] - Need more explicit debug when PJSIP dialstring is invalid
  • [ASTERISK-24612] - res_pjsip: No information if a required sorcery wizard is not loaded
  • [ASTERISK-24616] - Crash in res_format_attr_h264 due to invalid string copy
  • [ASTERISK-24632] - install_prereq script installs pjproject without IPv6 support
  • [ASTERISK-24677] - ARI GET variable on channel provides unhelpful response on non-existent variable
  • [ASTERISK-24685] - "pjsip show version" CLI command
  • [ASTERISK-24689] - Segfault on hangup after outgoing PRI-Euroisdn call
  • [ASTERISK-24700] - CRASH: NULL channel is being passed to ast_bridge_transfer_attended()
  • [ASTERISK-24701] - Stasis: Write timeout on WebSocket fails to fully disconnect underlying socket, leading to events being dropped with no additional information
  • [ASTERISK-24716] - Improve pjsip log messages for presence subscription failure
  • [ASTERISK-24724] - 'httpstatus' Web Page Produces Incomplete HTML
  • [ASTERISK-24727] - PJSIP: Crash experienced during multi-Asterisk transfer scenario.
  • [ASTERISK-24739] - [patch] - Out of files -- call fails -- numerous files with inodes from under /usr/share/zoneinfo, mostly posixrules
  • [ASTERISK-24740] - [patch]Segmentation fault on aoc-e event
  • [ASTERISK-24741] - dtls_handler causes Asterisk to crash
  • [ASTERISK-24742] - [patch] Fix ast_odbc_find_table function in res_odbc
  • [ASTERISK-24748] - res_pjsip: If wizards explicitly configured in sorcery.conf false ERROR messages may occur
  • [ASTERISK-24751] - Integer values in json payload to ARI cause asterisk to crash
  • [ASTERISK-24752] - Crash in bridge_manager_service_req when bridge is destroyed by ARI during shutdown
  • [ASTERISK-24755] - Asterisk sends unexpected early BYE to transferrer during attended transfer when using a Stasis bridge
  • [ASTERISK-24768] - res_timing_pthread: file descriptor leak
  • [ASTERISK-24769] - res_pjsip_sdp_rtp: Local ICE candidates leaked
  • [ASTERISK-24771] - ${CHANNEL(pjsip)} - segfault
  • [ASTERISK-24772] - ODBC error in realtime sippeers when device unregisters under MariaDB
  • [ASTERISK-24785] - 'Expires' header missing from 200 OK on REGISTER
  • [ASTERISK-24786] - [patch] - Asterisk terminates when playing a voicemail stored in LDAP
  • [ASTERISK-24787] - [patch] - Microsoft exchange incompatibility for playing back messages stored in IMAP - play_message: No origtime
  • [ASTERISK-24791] - Crash in ast_rtcp_write_report
  • [ASTERISK-24796] - Codecs and bucket schema's prevent module unload
  • [ASTERISK-24797] - bridge_softmix: G.729 codec license held
  • [ASTERISK-24799] - [patch] make fails with undefined reference to SSLv3_client_method
  • [ASTERISK-24800] - Crash in __sip_reliable_xmit due to invalid thread ID being passed to pthread_kill
  • [ASTERISK-24807] - Missing mandatory field Max-Forwards
  • [ASTERISK-24808] - res_config_odbc: Improper escaping of backslashes occurs with MySQL
  • [ASTERISK-24812] - ARI: Creating channels through /channels resource always uses SLIN, which results in unneeded transcoding
  • [ASTERISK-24814] - asterisk/lock.h: Fix syntax errors for non-gcc OSX with 64 bit integers
  • [ASTERISK-24817] - init_logger_chain: unreachable code block
  • [ASTERISK-24825] - Caller ID not recognized using Centrex/Distinctive dialing
  • [ASTERISK-24828] - Fix Frame Leaks
  • [ASTERISK-24830] - res_rtp_asterisk.c checks USE_PJPROJECT not HAVE_PJPROJECT
  • [ASTERISK-24838] - chan_sip: Locking inversion occurs when building a peer causes a peer poke during request handling
  • [ASTERISK-24840] - res_pjsip: conflicting endpoint identifiers
  • [ASTERISK-24872] - [patch] AMI PJSIPShowEndpoint closes AMI connection on error
  • [ASTERISK-24876] - Investigate reference leaks from tests/channels/local/local_optimize_away
  • [ASTERISK-24879] - [patch]Compilation fails due to 64bit time under OpenBSD
  • [ASTERISK-24880] - [patch]Compilation under OpenBSD
  • [ASTERISK-24882] - chan_sip: Improve usage of REF_DEBUG

Improvement

  • [ASTERISK-24745] - [patch]Add no_answer to ARI hangup causes
  • [ASTERISK-24790] - Reduce spurious noise in logs from voicemail - Couldn't find mailbox %s in context
  • [ASTERISK-24811] - asterisk-publication sorcery object does not use realtime

New Feature

  • [ASTERISK-17899] - [patch] Adds a 'ignorecryptolifetime' (Ignore Crypto Lifetime) option to sip.conf for SRTP keys specifying optional 'lifetime'
  • [ASTERISK-24703] - ARI: Add the ability to "transfer" (redirect) a channel


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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.3.0-rc1


Thank you for your continued support of Asterisk!


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