[asterisk-dev] Asterisk 12.5.0-rc1 Now Available

[Asterisk Development Team]

The Asterisk Development Team has announced the first release candidate of
Asterisk 12.5.0. This release candidate is available for immediate
download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 12.5.0-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:

Improvements made in this release:
-----------------------------------
 * ASTERISK-24036 - ARI: Recording resource should allow copying a
      recording (Reported by Samuel Galarneau)
 * ASTERISK-24037 - ARI: RecordingFinished event should return
      duration of recording (Reported by Samuel Galarneau)
 * ASTERISK-21178 - Improve documentation for manager command
      Getvar, Setvar (Reported by Rusty Newton)
 * ASTERISK-23692 - ARI: Add a Messaging Capability (Reported by
      Matt Jordan)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-23852 - ARI mixing bridges should propagate linkedids.
      (Reported by Richard Mudgett)
 * ASTERISK-23911 - URIENCODE/URIDECODE: WARNING about passing an
      empty string is a bit over zealous (Reported by Matt Jordan)
 * ASTERISK-23985 - PresenceState Action response does not contain
      ActionID; duplicates Message Header (Reported by Matt Jordan)
 * ASTERISK-23814 - No call started after peer dialed (Reported by
      Igor Goncharovsky)
 * ASTERISK-24087 - [patch]chan_sip: sip_subscribe_mwi_destroy
      should not call sip_destroy (Reported by Corey Farrell)
 * ASTERISK-23987 - BridgeWait: channel entering into holding
      bridge that is being destroyed fails to successfully join the
      newly created holding bridge (Reported by Matt Jordan)
 * ASTERISK-23969 - SendMessage AMI action Cant Send Text Message
      Over PJSIP (Reported by Andrew Nagy)
 * ASTERISK-23818 - PBX_Lua: after asterisk startup module is
      loaded, but dialplan not available (Reported by Dennis Guse)
 * ASTERISK-23847 - Alembic voicemail script - 'recording' column
      should be longblob on MySQL (Reported by Stephen More)
 * ASTERISK-23825 - Alembic scripts - table queue_members missing
      unique index on column uniqueid (Reported by Stephen More)
 * ASTERISK-23909 - Alembic scripts - table sippeers could use a
      longer useragent column (Reported by Stephen More)
 * ASTERISK-23941 - ARI: Attended transfers of channels into Stasis
      application lose information (Reported by Matt Jordan)
 * ASTERISK-18345 - [patch] sips connection dropped by asterisk
      with a large INVITE (Reported by Stephane Chazelas)
 * ASTERISK-23508 - Memory Corruption in
      __ast_string_field_ptr_build_va (Reported by Arnd Schmitter)

New Features made in this release:
-----------------------------------
 * ASTERISK-24000 - chan_pjsip: Add accountcode setting (Reported
      by Matt Jordan)
 * ASTERISK-24119 - HEP: Add module that exports RTCP information
      to a Homer Capture Server (Reported by Matt Jordan)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-12.5.0-rc1

Thank you for your continued support of Asterisk!