Asterisk News

Asterisk Releases

Asterisk 11.14.0-rc2 Now Available

Nov 7, 2014

The Asterisk Development Team has announced the second release candidate of Asterisk 11.14.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 11.14.0-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:

  • Fix unintential memory retention in stringfields.
    (Closes issue ASTERISK-24307. Reported by Etienne Lessard)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.14.0-rc2

Thank you for your continued support of Asterisk!


Asterisk 1.8.32.0-rc2 Now Available

Nov 7, 2014

The Asterisk Development Team has announced the second release candidate of Asterisk 1.8.32.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 1.8.32.0-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:

  • Fix unintential memory retention in stringfields.
    (Closes issue ASTERISK-24307. Reported by Etienne Lessard)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.32.0-rc2

Thank you for your continued support of Asterisk!


Asterisk 12.7.0-rc1 Now Available

Nov 3, 2014

The Asterisk Development Team has announced the first release candidate of Asterisk 12.7.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 12.7.0-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-13797] - [patch] relax badshell tilde test
  • [ASTERISK-15879] - [patch] Failure to receive an ACK to a SIP Re-INVITE results in a SIP channel leak
  • [ASTERISK-18923] - res_fax_spandsp usage counter is wrong
  • [ASTERISK-20567] - bashism in autosupport
  • [ASTERISK-20784] - Failure to receive an ACK to a SIP Re-INVITE results in a SIP channel leak
  • [ASTERISK-21721] - SIP Failed to parse multiple Supported: headers
  • [ASTERISK-22791] - asterisk sends Re-INVITE after receiving a BYE
  • [ASTERISK-22945] - [patch] Memory leaks in chan_sip.c with realtime peers
  • [ASTERISK-23768] - [patch] Asterisk man page contains a (new) unquoted minus sign
  • [ASTERISK-23781] - outgoing missing as enum from contrib/ast-db-manage/config
  • [ASTERISK-23846] - Unistim multilines. Loss of voice after second call drops (on a second line).
  • [ASTERISK-24011] - [patch]safe_asterisk tries to set ulimit -n too high on linux systems with lots of RAM
  • [ASTERISK-24063] - [patch]Asterisk does not respect outbound proxy when sending qualify requests
  • [ASTERISK-24122] - Documentaton for res_pjsip option use_avpf needs to be fixed
  • [ASTERISK-24190] - IMAP voicemail causes segfault
  • [ASTERISK-24195] - bridge_native_rtp: Removing mixmonitor from a native RTP capable smart bridge doesn't cause the bridge to resume being a native rtp bridge
  • [ASTERISK-24199] - 'ALL' is specified in pjsip.conf.sample for TLS cipher but it is not valid
  • [ASTERISK-24224] - When using Bridge() dialplan application, surrogate channel appears in list and call count is inflated.
  • [ASTERISK-24262] - AMI CoreShowChannel missing several output fields and event documentation
  • [ASTERISK-24295] - crash: creating out of dialog OPTIONS request crashes
  • [ASTERISK-24304] - asterisk crashing randomly because of unistim channel
  • [ASTERISK-24312] - SIGABRT when improperly configured realtime pjsip
  • [ASTERISK-24321] - SIP deadlock when running automated queues tests
  • [ASTERISK-24325] - res_calendar_ews: cannot be used with neon 0.30
  • [ASTERISK-24326] - res_rtp_asterisk: ICE-TCP candidates are incorrectly attempted
  • [ASTERISK-24327] - bridge_native_rtp: Smart bridge operation to softmix sometimes fails to properly re-INVITE remotely bridged participants
  • [ASTERISK-24335] - [PATCH] Asterisk incorrectly responds 503 to INVITE retransmissions of rejected calls
  • [ASTERISK-24339] - Swagger API Docs have incorrect basePath
  • [ASTERISK-24348] - Built-in editline tab complete segfault with MALLOC_DEBUG
  • [ASTERISK-24350] - PJSIP shows commands prints unneeded headers
  • [ASTERISK-24354] - AMI sendMessage closes AMI connection on error
  • [ASTERISK-24356] - PJSIP: Directed pickup causes deadlock
  • [ASTERISK-24357] - [fax] Out of bounds error in update_modem_bits
  • [ASTERISK-24362] - res_hep leaks reference to configuration
  • [ASTERISK-24369] - res_pjsip: Large message on reliable transport can cause empty messages to be passed from the PJSIP stack up, causing crashes in multiple locations
  • [ASTERISK-24370] - res_pjsip/pjsip_options: OPTIONS request sent to Asterisk with no user in request is always 404'd
  • [ASTERISK-24378] - Release AMI connections on shutdown
  • [ASTERISK-24381] - res_pjsip_sdp_rtp: Declined media streams are interpreted, leading to erroneous 488 rejections
  • [ASTERISK-24382] - chan_pjsip: Calling PJSIP_MEDIA_OFFER on a non-PJSIP channel results in an invalid reference of a channel pvt and a FRACK
  • [ASTERISK-24383] - res_rtp_asterisk: Crash if no candidates received for component
  • [ASTERISK-24384] - chan_motif: format capabilities leak on module load error
  • [ASTERISK-24385] - chan_sip: process_sdp leaks on an error path
  • [ASTERISK-24387] - res_pjsip: rport sent from UAS MUST include the port that the UAC sent the request on
  • [ASTERISK-24392] - res_fax: fax gateway sessions leak
  • [ASTERISK-24393] - rtptimeout=0 doesn't disable rtptimeout
  • [ASTERISK-24394] - CDR: FRACK with PJSIP directed pickup.
  • [ASTERISK-24398] - Initialize auth_rejection_permanent on client state to the configuration parameter value
  • [ASTERISK-24406] - Some caller ID strings are parsed differently since 11.13.0
  • [ASTERISK-24411] - [patch] Status of outbound registration is not changed upon unregistering.
  • [ASTERISK-24415] - Missing AMI VarSet events when channels inherit variables.
  • [ASTERISK-24425] - [patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
  • [ASTERISK-24426] - CDR Batch mode: size used as time value after first expire
  • [ASTERISK-24430] - missing letter "p" in word response in OriginateResponse event documentation
  • [ASTERISK-24432] - Install refcounter.py when REF_DEBUG is enabled
  • [ASTERISK-24436] - Missing header in res/res_srtp.c when compiling against libsrtp-1.5.0
  • [ASTERISK-24437] - Review implementation of ast_bridge_impart for leaks and document proper usage
  • [ASTERISK-24453] - manager: acl_change_sub leaks
  • [ASTERISK-24454] - app_queue: ao2_iterator not destroyed, causing leak
  • [ASTERISK-24457] - res_fax: fax gateway frames leak
  • [ASTERISK-24462] - res_pjsip: Stale qualify statistics after disablementation
  • [ASTERISK-24466] - app_queue: fix a couple leaks to struct call_queue
  • [ASTERISK-24476] - main/app.c / app_voicemail: ast_writestream leaks

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-12.7.0-rc1

Thank you for your continued support of Asterisk!


Asterisk 11.14.0-rc1 Now Available

Nov 3, 2014

The Asterisk Development Team has announced the first release candidate of Asterisk 11.14.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 11.14.0-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-13797] - [patch] relax badshell tilde test
  • [ASTERISK-15879] - [patch] Failure to receive an ACK to a SIP Re-INVITE results in a SIP channel leak
  • [ASTERISK-18923] - res_fax_spandsp usage counter is wrong
  • [ASTERISK-20567] - bashism in autosupport
  • [ASTERISK-20784] - Failure to receive an ACK to a SIP Re-INVITE results in a SIP channel leak
  • [ASTERISK-21721] - SIP Failed to parse multiple Supported: headers
  • [ASTERISK-22791] - asterisk sends Re-INVITE after receiving a BYE
  • [ASTERISK-22945] - [patch] Memory leaks in chan_sip.c with realtime peers
  • [ASTERISK-23768] - [patch] Asterisk man page contains a (new) unquoted minus sign
  • [ASTERISK-23846] - Unistim multilines. Loss of voice after second call drops (on a second line).
  • [ASTERISK-24011] - [patch]safe_asterisk tries to set ulimit -n too high on linux systems with lots of RAM
  • [ASTERISK-24063] - [patch]Asterisk does not respect outbound proxy when sending qualify requests
  • [ASTERISK-24190] - IMAP voicemail causes segfault
  • [ASTERISK-24304] - asterisk crashing randomly because of unistim channel
  • [ASTERISK-24325] - res_calendar_ews: cannot be used with neon 0.30
  • [ASTERISK-24326] - res_rtp_asterisk: ICE-TCP candidates are incorrectly attempted
  • [ASTERISK-24335] - [PATCH] Asterisk incorrectly responds 503 to INVITE retransmissions of rejected calls
  • [ASTERISK-24348] - Built-in editline tab complete segfault with MALLOC_DEBUG
  • [ASTERISK-24354] - AMI sendMessage closes AMI connection on error
  • [ASTERISK-24357] - [fax] Out of bounds error in update_modem_bits
  • [ASTERISK-24378] - Release AMI connections on shutdown
  • [ASTERISK-24383] - res_rtp_asterisk: Crash if no candidates received for component
  • [ASTERISK-24384] - chan_motif: format capabilities leak on module load error
  • [ASTERISK-24385] - chan_sip: process_sdp leaks on an error path
  • [ASTERISK-24390] - astobj2: REF_DEBUG reports false leaks with ao2_callback with OBJ_MULTIPLE
  • [ASTERISK-24392] - res_fax: fax gateway sessions leak
  • [ASTERISK-24393] - rtptimeout=0 doesn't disable rtptimeout
  • [ASTERISK-24406] - Some caller ID strings are parsed differently since 11.13.0
  • [ASTERISK-24425] - [patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
  • [ASTERISK-24430] - missing letter "p" in word response in OriginateResponse event documentation
  • [ASTERISK-24432] - Install refcounter.py when REF_DEBUG is enabled
  • [ASTERISK-24436] - Missing header in res/res_srtp.c when compiling against libsrtp-1.5.0
  • [ASTERISK-24454] - app_queue: ao2_iterator not destroyed, causing leak
  • [ASTERISK-24457] - res_fax: fax gateway frames leak
  • [ASTERISK-24466] - app_queue: fix a couple leaks to struct call_queue
  • [ASTERISK-24476] - main/app.c / app_voicemail: ast_writestream leaks

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.14.0-rc1

Thank you for your continued support of Asterisk!


Asterisk 1.8.32.0-rc1 Now Available

Nov 3, 2014

The Asterisk Development Team has announced the first release candidate of Asterisk 1.8.32.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 1.8.32.0-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-13797] - [patch] relax badshell tilde test
  • [ASTERISK-15879] - [patch] Failure to receive an ACK to a SIP Re-INVITE results in a SIP channel leak
  • [ASTERISK-18923] - res_fax_spandsp usage counter is wrong
  • [ASTERISK-20784] - Failure to receive an ACK to a SIP Re-INVITE results in a SIP channel leak
  • [ASTERISK-21721] - SIP Failed to parse multiple Supported: headers
  • [ASTERISK-22791] - asterisk sends Re-INVITE after receiving a BYE
  • [ASTERISK-22945] - [patch] Memory leaks in chan_sip.c with realtime peers
  • [ASTERISK-23768] - [patch] Asterisk man page contains a (new) unquoted minus sign
  • [ASTERISK-23846] - Unistim multilines. Loss of voice after second call drops (on a second line).
  • [ASTERISK-24011] - [patch]safe_asterisk tries to set ulimit -n too high on linux systems with lots of RAM
  • [ASTERISK-24063] - [patch]Asterisk does not respect outbound proxy when sending qualify requests
  • [ASTERISK-24190] - IMAP voicemail causes segfault
  • [ASTERISK-24325] - res_calendar_ews: cannot be used with neon 0.30
  • [ASTERISK-24335] - [PATCH] Asterisk incorrectly responds 503 to INVITE retransmissions of rejected calls
  • [ASTERISK-24348] - Built-in editline tab complete segfault with MALLOC_DEBUG
  • [ASTERISK-24357] - [fax] Out of bounds error in update_modem_bits
  • [ASTERISK-24390] - astobj2: REF_DEBUG reports false leaks with ao2_callback with OBJ_MULTIPLE
  • [ASTERISK-24393] - rtptimeout=0 doesn't disable rtptimeout
  • [ASTERISK-24406] - Some caller ID strings are parsed differently since 11.13.0
  • [ASTERISK-24425] - [patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
  • [ASTERISK-24432] - Install refcounter.py when REF_DEBUG is enabled
  • [ASTERISK-24436] - Missing header in res/res_srtp.c when compiling against libsrtp-1.5.0
  • [ASTERISK-24476] - main/app.c / app_voicemail: ast_writestream leaks

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.32.0-rc1

Thank you for your continued support of Asterisk!


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