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asteriskteam at digium... Guest
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Posted: Tue Aug 19, 2014 2:39 pm Post subject: [asterisk-users] Asterisk 12.5.0 Now Available |
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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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noloader at gmail.com Guest
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Posted: Tue Aug 19, 2014 3:29 pm Post subject: [asterisk-users] Asterisk 12.5.0 Now Available |
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On Tue, Aug 19, 2014 at 3:38 PM, Asterisk Development Team
<asteriskteam@digium.com> wrote:
Quote: | 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!
| Congratulations, team.
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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