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!