Asterisk News

Asterisk Releases

Asterisk 13.10.0 Now Available

Jul 21, 2016

The Asterisk Development Team has announced the release of Asterisk 13.10.0.

This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.10.0 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-16115] - [patch] problem with ringinuse=no, queue members receive sometimes two calls
  • [ASTERISK-24436] - Missing header in res/res_srtp.c when compiling against libsrtp-1.5.0
  • [ASTERISK-24463] - Voicemail email address corrupt or not sent when message is in the process of being recorded during reload
  • [ASTERISK-24986] - keepalive INFO packages ignored by asterisk
  • [ASTERISK-25262] - Memory leak when a caller channel does multiple dials and CEL is enabled
  • [ASTERISK-25352] - res_hep_rtcp correlation_id is different then res_hep
  • [ASTERISK-25777] - data race in threadpool
  • [ASTERISK-25826] - PJSIP / Sorcery slow load from realtime
  • [ASTERISK-25917] - [patch]app_voicemail: passwordlocation=spooldir only works if you manually add secret.conf yourself
  • [ASTERISK-25938] - res_odbc: MySQL/MariaDB statement LAST_INSERT_ID() always returns zero.
  • [ASTERISK-25941] - chan_pjsip: Crash on an immediate SIP final response
  • [ASTERISK-25950] - [patch]SIP channel does not send PeerStatus events for autocreated peers
  • [ASTERISK-25954] - Manager QueueSummary and QueueStatus Actions are case sensitive to QueueName
  • [ASTERISK-25956] - Compilation error in conditionally compiled code in config_options.c
  • [ASTERISK-25961] - tests/channels/SIP/sip_tls_call: Sporadic crash when running test
  • [ASTERISK-25963] - func_odbc requires reconnect checks for stale connections
  • [ASTERISK-25964] - Outbound registrations created via ARI/push configuration do not clean up outbound registrations currently in flight
  • [ASTERISK-25968] - pjproject_bundled: Configure and make need to be re-tested
  • [ASTERISK-25970] - Segfault in pjsip_url_compare
  • [ASTERISK-25978] - res_pjsip_authenticator_digest: Should not use source port in nonce verification
  • [ASTERISK-25990] - PJSIP TLS registration should respect client_uri scheme when generating Contact URI
  • [ASTERISK-25993] - pjproject: Allow bundling to not require everything it does
  • [ASTERISK-25998] - file: Crash when using nativeformats
  • [ASTERISK-26005] - res_pjsip: Multiple SIP messages are combined into 1 TCP packet
  • [ASTERISK-26007] - res_pjsip: Endpoints deleting early after upgrade from 13.8.2 to 13.9
  • [ASTERISK-26008] - app_followme does not delete recorded name prompt
  • [ASTERISK-26014] - res_sorcery_astdb: Make tolerant of unknown fields
  • [ASTERISK-26029] - parking: ast_parking_park_call should return parking_space instead of parking_exten
  • [ASTERISK-26030] - call cut because of double Session-Expires header in re-invite after proxy authentication is required
  • [ASTERISK-26034] - T.38 passthrough problem behind firewall due to early nosignal packet
  • [ASTERISK-26038] - 'make install' doesn't seem to install OS/X init files
  • [ASTERISK-26054] - Asterisk crashes (core dump)
  • [ASTERISK-26063] - ${PJSIP_HEADER(read,Call-ID)} does not work - documentation needs clarification for when read/write is possible
  • [ASTERISK-26065] - chan_pjsip: MWI NOTIFY contents not ordered properly
  • [ASTERISK-26069] - Asterisk truncates To: header, dropping the closing '>'
  • [ASTERISK-26070] - ari/channels: Creating a local channel without an originator adds all audio formats to it's capabilities
  • [ASTERISK-26074] - res_odbc: Deadlock within UnixODBC
  • [ASTERISK-26078] - core: Memory leak in logging
  • [ASTERISK-26083] - ARI: Announcer channels staying around after playback to a bridge is finished
  • [ASTERISK-26089] - Invalid security events during boot using PJSIP Realtime
  • [ASTERISK-26091] - [patch] ar cru creates warning, instead use ar cr
  • [ASTERISK-26092] - [Segfault] in res_rtp_asterisk.c:4268 after Remotely bridged channels
  • [ASTERISK-26096] - res_hep: Crash when configuration file is missing
  • [ASTERISK-26097] - [patch] CLI: show maximum file descriptors
  • [ASTERISK-26099] - res_pjsip_pubsub: Crash when sending request due to server timeout
  • [ASTERISK-26126] - [patch] leverage 'bindaddr' for TLS in http.conf
  • [ASTERISK-26127] - res_pjsip_session: Crash due to race condition between res_pjsip_session unload and timer
  • [ASTERISK-26128] - Alembic scripts are failing
  • [ASTERISK-26130] - [patch] WebRTC: Should use latest DTLS version.
  • [ASTERISK-26138] - chan_unistim: Under FreeBSD, chan_unistim generates a compile error
  • [ASTERISK-26139] - test_res_pjsip_scheduler: Compile failure if pjproject isn't installed in a system location
  • [ASTERISK-26140] - res_rtp_asterisk: gcc 6 caught a self-comparison
  • [ASTERISK-26141] - res_fax: fax_v21_session_new leaks reference to v21_details
  • [ASTERISK-26160] - pjsip: Updated->Reachable during qualify
  • [ASTERISK-26177] - func_odbc: Database handle is kept when it should be released

Improvement

New Feature

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.10.0

Thank you for your continued support of Asterisk!


Asterisk 11.23.0 Now Available

Jul 21, 2016

The Asterisk Development Team has announced the release of Asterisk 11.23.0.

This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 11.23.0 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-16115] - [patch] problem with ringinuse=no, queue members receive sometimes two calls
  • [ASTERISK-24436] - Missing header in res/res_srtp.c when compiling against libsrtp-1.5.0
  • [ASTERISK-24463] - Voicemail email address corrupt or not sent when message is in the process of being recorded during reload
  • [ASTERISK-25407] - Asterisk fails to log to multiple syslog destinations
  • [ASTERISK-25510] - [patch]Log to syslog failing
  • [ASTERISK-25874] - app_voicemail: Stack buffer overflow in test_voicemail_notify_endl
  • [ASTERISK-25888] - Frequent segfaults in function can_ring_entry() of app_queue.c
  • [ASTERISK-25912] - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding them to the local hangupcauses via ast_channel_hangupcause_hash_set
  • [ASTERISK-25917] - [patch]app_voicemail: passwordlocation=spooldir only works if you manually add secret.conf yourself
  • [ASTERISK-25934] - chan_sip should not require sipregs or updateable sippeers table unless rt
  • [ASTERISK-25954] - Manager QueueSummary and QueueStatus Actions are case sensitive to QueueName
  • [ASTERISK-26008] - app_followme does not delete recorded name prompt
  • [ASTERISK-26030] - call cut because of double Session-Expires header in re-invite after proxy authentication is required
  • [ASTERISK-26034] - T.38 passthrough problem behind firewall due to early nosignal packet
  • [ASTERISK-26038] - 'make install' doesn't seem to install OS/X init files
  • [ASTERISK-26069] - Asterisk truncates To: header, dropping the closing '>'
  • [ASTERISK-26091] - [patch] ar cru creates warning, instead use ar cr
  • [ASTERISK-26097] - [patch] CLI: show maximum file descriptors
  • [ASTERISK-26126] - [patch] leverage 'bindaddr' for TLS in http.conf
  • [ASTERISK-26130] - [patch] WebRTC: Should use latest DTLS version.
  • [ASTERISK-26138] - chan_unistim: Under FreeBSD, chan_unistim generates a compile error
  • [ASTERISK-26140] - res_rtp_asterisk: gcc 6 caught a self-comparison
  • [ASTERISK-26141] - res_fax: fax_v21_session_new leaks reference to v21_details

Improvement

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0

Thank you for your continued support of Asterisk!


Asterisk 13.10.0-rc3 Now Available

Jul 14, 2016

The Asterisk Development Team has announced the third release candidate of Asterisk 13.10.0.

This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.10.0-rc3 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: Release Notes - Asterisk - Version 13.10.0

Bug

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.10.0-rc3

Thank you for your continued support of Asterisk!


Asterisk 13.10.0-rc2 Now Available

Jul 5, 2016

The Asterisk Development Team has announced the second release candidate of Asterisk 13.10.0.

This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.10.0-rc2 resolves an issue 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-26099] - ASTERISK-26099 - res_pjsip_pubsub: Crash when sending request due to server timeout
  • [ASTERISK-26144] - ASTERISK-26144 - Crash on loading codecs g729/g723

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.10.0-rc2

Thank you for your continued support of Asterisk!

 


Asterisk 13.10.0-rc1 Now Available

Jun 23, 2016

The Asterisk Development Team has announced the first release candidate of Asterisk 13.10.0.

This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.10.0-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-16115] - [patch] problem with ringinuse=no, queue members receive sometimes two calls
  • [ASTERISK-24436] - Missing header in res/res_srtp.c when compiling against libsrtp-1.5.0
  • [ASTERISK-24463] - Voicemail email address corrupt or not sent when message is in the process of being recorded during reload
  • [ASTERISK-24986] - keepalive INFO packages ignored by asterisk
  • [ASTERISK-25262] - Memory leak when a caller channel does multiple dials and CEL is enabled
  • [ASTERISK-25352] - res_hep_rtcp correlation_id is different then res_hep
  • [ASTERISK-25777] - data race in threadpool
  • [ASTERISK-25826] - PJSIP / Sorcery slow load from realtime
  • [ASTERISK-25917] - [patch]app_voicemail: passwordlocation=spooldir only works if you manually add secret.conf yourself
  • [ASTERISK-25938] - res_odbc: MySQL/MariaDB statement LAST_INSERT_ID() always returns zero.
  • [ASTERISK-25941] - chan_pjsip: Crash on an immediate SIP final response
  • [ASTERISK-25950] - [patch]SIP channel does not send PeerStatus events for autocreated peers
  • [ASTERISK-25954] - Manager QueueSummary and QueueStatus Actions are case sensitive to QueueName
  • [ASTERISK-25956] - Compilation error in conditionally compiled code in config_options.c
  • [ASTERISK-25961] - tests/channels/SIP/sip_tls_call: Sporadic crash when running test
  • [ASTERISK-25963] - func_odbc requires reconnect checks for stale connections
  • [ASTERISK-25964] - Outbound registrations created via ARI/push configuration do not clean up outbound registrations currently in flight
  • [ASTERISK-25968] - pjproject_bundled: Configure and make need to be re-tested
  • [ASTERISK-25970] - Segfault in pjsip_url_compare
  • [ASTERISK-25978] - res_pjsip_authenticator_digest: Should not use source port in nonce verification
  • [ASTERISK-25990] - PJSIP TLS registration should respect client_uri scheme when generating Contact URI
  • [ASTERISK-25993] - pjproject: Allow bundling to not require everything it does
  • [ASTERISK-25998] - file: Crash when using nativeformats
  • [ASTERISK-26005] - res_pjsip: Multiple SIP messages are combined into 1 TCP packet
  • [ASTERISK-26007] - res_pjsip: Endpoints deleting early after upgrade from 13.8.2 to 13.9
  • [ASTERISK-26008] - app_followme does not delete recorded name prompt
  • [ASTERISK-26014] - res_sorcery_astdb: Make tolerant of unknown fields
  • [ASTERISK-26029] - parking: ast_parking_park_call should return parking_space instead of parking_exten
  • [ASTERISK-26030] - call cut because of double Session-Expires header in re-invite after proxy authentication is required
  • [ASTERISK-26034] - T.38 passthrough problem behind firewall due to early nosignal packet
  • [ASTERISK-26038] - 'make install' doesn't seem to install OS/X init files
  • [ASTERISK-26054] - Asterisk crashes (core dump)
  • [ASTERISK-26063] - ${PJSIP_HEADER(read,Call-ID)} does not work - documentation needs clarification for when read/write is possible
  • [ASTERISK-26065] - chan_pjsip: MWI NOTIFY contents not ordered properly
  • [ASTERISK-26069] - Asterisk truncates To: header, dropping the closing '>'
  • [ASTERISK-26070] - ari/channels: Creating a local channel without an originator adds all audio formats to it's capabilities
  • [ASTERISK-26074] - res_odbc: Deadlock within UnixODBC
  • [ASTERISK-26078] - core: Memory leak in logging
  • [ASTERISK-26083] - ARI: Announcer channels staying around after playback to a bridge is finished
  • [ASTERISK-26089] - Invalid security events during boot using PJSIP Realtime
  • [ASTERISK-26091] - [patch] ar cru creates warning, instead use ar cr
  • [ASTERISK-26092] - [Segfault] in res_rtp_asterisk.c:4268 after Remotely bridged channels
  • [ASTERISK-26096] - res_hep: Crash when configuration file is missing
  • [ASTERISK-26097] - [patch] CLI: show maximum file descriptors
  • [ASTERISK-26099] - res_pjsip_pubsub: Crash when sending request due to server timeout
  • [ASTERISK-26126] - [patch] leverage 'bindaddr' for TLS in http.conf
  • [ASTERISK-26127] - res_pjsip_session: Crash due to race condition between res_pjsip_session unload and timer
  • [ASTERISK-26128] - Alembic scripts are failing
  • [ASTERISK-26130] - [patch] WebRTC: Should use latest DTLS version.
  • [ASTERISK-26138] - chan_unistim: Under FreeBSD, chan_unistim generates a compile error
  • [ASTERISK-26139] - test_res_pjsip_scheduler: Compile failure if pjproject isn't installed in a system location
  • [ASTERISK-26140] - res_rtp_asterisk: gcc 6 caught a self-comparison
  • [ASTERISK-26141] - res_fax: fax_v21_session_new leaks reference to v21_details

Improvement

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.10.0-rc1

Thank you for your continued support of Asterisk!


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