Asterisk 12.7.0-rc1 Now Available

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] – 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-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

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!

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