Asterisk News

Asterisk Releases

Asterisk 12.1.0-rc3 Now Available

Mar 1, 2014

The Asterisk Development Team has announced the third release candidate of Asterisk 12.1.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 12.1.0-rc3 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:

  • chan_sip: Fix crash in ast_channel_hangupcause_set().
    (Closes issue ASTERISK-23135. Reported by OK)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-12.1.0-rc3

Thank you for your continued support of Asterisk!


Asterisk 11.8.0-rc3 Now Available

Mar 1, 2014

The Asterisk Development Team has announced the third release candidate of Asterisk 11.8.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 11.8.0-rc3 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:

  • chan_sip: Fix crash in ast_channel_hangupcause_set().
    (Closes issue ASTERISK-23135. Reported by OK)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.8.0-rc3

Thank you for your continued support of Asterisk!


Asterisk 12.1.0-rc2 Now Available

Feb 27, 2014

The Asterisk Development Team has announced the second release candidate of Asterisk 12.1.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 12.1.0-rc2 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:

  • main: Initialize dialplan providing core components prior to module pre-load
    (Closes issue ASTERISK-23320. Reported by xrobau)
  • res_fax: Warn that minrate=2400 is not valid for V.27 instead of failing load.
    (Closes issue ASTERISK-23231. Reported by David Brillert)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-12.1.0-rc2

Thank you for your continued support of Asterisk!


Asterisk 11.8.0-rc2 Now Available

Feb 27, 2014

The Asterisk Development Team has announced the second release candidate of Asterisk 11.8.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 11.8.0-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:

  • res_rtp_asterisk: Fix checklist creating problems in ICE sessions
    (Issue ASTERISK-23213. Reported by Andrea Suisani)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.8.0-rc2

Thank you for your continued support of Asterisk!


Asterisk 12.1.0-rc1 Now Available

Feb 6, 2014

The Asterisk Development Team has announced the first release candidate of Asterisk 12.1.0. This release candidate is available for immediate download at  http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 12.1.0-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:

Bugs

  • [ASTERISK-17138] - [patch] Asterisk not re-registering after it receives "Forbidden - wrong password on authentication"
  • [ASTERISK-17727] - [patch] TLS doesn't get all certificate chain
  • [ASTERISK-17837] - extconfig.conf - Maximum Include level (1) exceeded
  • [ASTERISK-19773] - Asterisk crash on issuing Asterisk-CLI 'reload' command multiple times on cli_aliases
  • [ASTERISK-22486] - ARI: TCP Reset after 204 response
  • [ASTERISK-22662] - Documentation fix? - queues.conf says persistentmembers defaults to yes, it appears to lie
  • [ASTERISK-22757] - segfault in res_clialiases.so on reload when mapping "module reload" command
  • [ASTERISK-22790] - check_modem_rate() may return incorrect rate for V.27
  • [ASTERISK-22854] - [patch] - Deadlock between cel_pgsql unload and core_event_dispatcher taskprocessor thread
  • [ASTERISK-22861] - [patch]Specifying a null time as parameter to GotoIfTime or ExecIfTime causes segmentation fault
  • [ASTERISK-22871] - cel_pgsql module not loading after "reload" or "reload cel_pgsql.so" command
  • [ASTERISK-22884] - hangup_handler end with h extension: tests currently fail in Asterisk 12 +
  • [ASTERISK-22910] - [patch] - REPLACE() calls strcpy on overlapping memory when <replace-char> is empty
  • [ASTERISK-22911] - [patch]Asterisk fails to resume WebRTC call from hold
  • [ASTERISK-22924] - PJSIP MESSAGE support does not present the contact information on outbound messages
  • [ASTERISK-22946] - Local From tag regression with sipgate.de
  • [ASTERISK-22952] - res_pjsip_pubsub: crash when subscription_destructor is terminated from a non-PJSIP thread
  • [ASTERISK-22962] - performance spike on Local channels originated using ARI
  • [ASTERISK-22988] - [patch]T38 , SIP 488 after Rejecting image media offer due to invalid or unsupported syntax
  • [ASTERISK-23008] - Local channels loose CALLERID name when DAHDI channel connects
  • [ASTERISK-23011] - [patch]configure.ac and pbx_lua don't support lua 5.2
  • [ASTERISK-23018] - PJSip 'allow=all' results in failed SDP negotiation
  • [ASTERISK-23027] - [patch] Spelling typo "transfered" instead of "transferred"
  • [ASTERISK-23028] - [patch] Asterisk man pages contains unquoted minus signs
  • [ASTERISK-23034] - [patch] manager Originate doesn't abort on failed format_cap allocation
  • [ASTERISK-23046] - Custom CDR fields set during a GoSUB called from app_queue are not inserted
  • [ASTERISK-23051] - ARI: channel variables in JSON breaks passing parameters in JSON
  • [ASTERISK-23053] - The users of ao2_iterator_cleanup() are violating the ao2_iterator opacity.
  • [ASTERISK-23056] - [patch]INFINITY and NAN undefined
  • [ASTERISK-23061] - [Patch] 'textsupport' setting not mentioned in sip.conf.sample
  • [ASTERISK-23062] - res_pjsip AOR config option qualify_frequency is inconsistently respected
  • [ASTERISK-23065] - On Asterisk start, device state is INVALID for previously registered PJSIP endpoints, despite re-registrations
  • [ASTERISK-23071] - pjsip: mailboxes documentation is lacking
  • [ASTERISK-23072] - MWI subscription from Cisco SPA fails with PJSIP
  • [ASTERISK-23074] - Crash in ChanIsAvail app
  • [ASTERISK-23081] - PJSip Tab Expansion erroring
  • [ASTERISK-23082] - Including g722 in pjsip codec configuration results in unexpected SDP offers
  • [ASTERISK-23084] - [patch]rasterisk needlessly prints the AST-2013-007 warning
  • [ASTERISK-23100] - [patch] In chan_mgcp the ident in transmitted request and request queue may differ - fix for locking
  • [ASTERISK-23101] - pjsip: crash when parsing scheme from SIP URI
  • [ASTERISK-23106] - pjsip: ACK to 200 OK sent to private IP address on outbound channel's INVITE request
  • [ASTERISK-23128] - res_ari: Memory leak on response headers and some JSON response messages
  • [ASTERISK-23129] - segfault in res_pjsip_pubsub.so
  • [ASTERISK-23134] - [patch] res_rtp_asterisk port selection cannot handle selinux port restrictions
  • [ASTERISK-23143] - ARI: subscribing to an already subscribed resource returns a 500 error
  • [ASTERISK-23164] - CDRs: mid-call/pre-dial handlers perturb context/exten/app/data fields during Dial
  • [ASTERISK-23168] - Overriding outbound_auth in a pjsip registration causes ERROR, assert failure.
  • [ASTERISK-23177] - [patch] RealTime cant update sipbuddies table when registering or updating friend
  • [ASTERISK-23178] - devicestate.h: device state setting functions are documented with the wrong return values
  • [ASTERISK-23220] - STACK_PEEK function with no arguments causes crash/core dump
  • [ASTERISK-23249] - Skinny subchannel locking issues
  • [ASTERISK-23250] - CDR(start) function is broken due to sizeof dereference

Improvement

New Feature

  • [ASTERISK-23038] - Need config option to enable PJSIP logger at load time

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

http://downloads.asterisk.org/pub/telephony/Asterisk/ChangeLog-12.1.0-rc1

Thank you for your continued support of Asterisk!


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