[AsteriskBrasil] Fwd: [asterisk-dev] Asterisk 12.5.0 Now Available

[Sylvio Jollenbeck]

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

The release of Asterisk 12.5.0 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:

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, please see the ChangeLog:

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

Thank you for your continued support of Asterisk!

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-- 
Sylvio Jollenbeck
skype: sylvio.jollenbeck
www.hosannatecnologia.com.br
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