Asterisk 12.7.0 Now Available

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

The release of Asterisk 12.7.0 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-13797] – relax badshell tilde test
  • [ASTERISK-15879] – 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] – Memory leaks in chan_sip.c with realtime peers
  • [ASTERISK-23768] – 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] – safe_asterisk tries to set ulimit -n too high on linux systems with lots of RAM
  • [ASTERISK-24063] – 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-24307] – Unintentional memory retention in stringfields
  • [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] – Status of outbound registration is not changed upon unregistering.
  • [ASTERISK-24415] – Missing AMI VarSet events when channels inherit variables.
  • [ASTERISK-24425] – 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
  • [ASTERISK-24487] – configuration: sections should be loadable as template even when not marked

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

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

Thank you for your continued support of Asterisk!

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