<div dir="ltr"><div class="gmail_quote"><br>The Asterisk Development Team has announced the release of Asterisk 12.5.0.<br>
This release is available for immediate download at<br>
<a href="http://downloads.asterisk.org/pub/telephony/asterisk" target="_blank">http://downloads.asterisk.org/pub/telephony/asterisk</a><br>
<br>
The release of Asterisk 12.5.0 resolves several issues reported by the<br>
community and would have not been possible without your participation.<br>
Thank you!<br>
<br>
The following are the issues resolved in this release:<br>
<br>
Improvements made in this release:<br>
-----------------------------------<br>
* ASTERISK-24036 - ARI: Recording resource should allow copying a<br>
recording (Reported by Samuel Galarneau)<br>
* ASTERISK-24037 - ARI: RecordingFinished event should return<br>
duration of recording (Reported by Samuel Galarneau)<br>
* ASTERISK-21178 - Improve documentation for manager command<br>
Getvar, Setvar (Reported by Rusty Newton)<br>
* ASTERISK-23692 - ARI: Add a Messaging Capability (Reported by<br>
Matt Jordan)<br>
<br>
Bugs fixed in this release:<br>
-----------------------------------<br>
* ASTERISK-23852 - ARI mixing bridges should propagate linkedids.<br>
(Reported by Richard Mudgett)<br>
* ASTERISK-23911 - URIENCODE/URIDECODE: WARNING about passing an<br>
empty string is a bit over zealous (Reported by Matt Jordan)<br>
* ASTERISK-23985 - PresenceState Action response does not contain<br>
ActionID; duplicates Message Header (Reported by Matt Jordan)<br>
* ASTERISK-23814 - No call started after peer dialed (Reported by<br>
Igor Goncharovsky)<br>
* ASTERISK-24087 - [patch]chan_sip: sip_subscribe_mwi_destroy<br>
should not call sip_destroy (Reported by Corey Farrell)<br>
* ASTERISK-23987 - BridgeWait: channel entering into holding<br>
bridge that is being destroyed fails to successfully join the<br>
newly created holding bridge (Reported by Matt Jordan)<br>
* ASTERISK-23969 - SendMessage AMI action Cant Send Text Message<br>
Over PJSIP (Reported by Andrew Nagy)<br>
* ASTERISK-23818 - PBX_Lua: after asterisk startup module is<br>
loaded, but dialplan not available (Reported by Dennis Guse)<br>
* ASTERISK-23847 - Alembic voicemail script - 'recording' column<br>
should be longblob on MySQL (Reported by Stephen More)<br>
* ASTERISK-23825 - Alembic scripts - table queue_members missing<br>
unique index on column uniqueid (Reported by Stephen More)<br>
* ASTERISK-23909 - Alembic scripts - table sippeers could use a<br>
longer useragent column (Reported by Stephen More)<br>
* ASTERISK-23941 - ARI: Attended transfers of channels into Stasis<br>
application lose information (Reported by Matt Jordan)<br>
* ASTERISK-18345 - [patch] sips connection dropped by asterisk<br>
with a large INVITE (Reported by Stephane Chazelas)<br>
* ASTERISK-23508 - Memory Corruption in<br>
__ast_string_field_ptr_build_va (Reported by Arnd Schmitter)<br>
<br>
New Features made in this release:<br>
-----------------------------------<br>
* ASTERISK-24000 - chan_pjsip: Add accountcode setting (Reported<br>
by Matt Jordan)<br>
* ASTERISK-24119 - HEP: Add module that exports RTCP information<br>
to a Homer Capture Server (Reported by Matt Jordan)<br>
<br>
For a full list of changes in this release, please see the ChangeLog:<br>
<br>
<a href="http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-12.5.0" target="_blank">http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-12.5.0</a><br>
<br>
Thank you for your continued support of Asterisk!<br>
<span class="HOEnZb"><font color="#888888"><br>
--<br>
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</font></span></div><br><br clear="all"><div><br></div>-- <br><div dir="ltr"><span style="font-family:trebuchet ms,sans-serif">Sylvio Jollenbeck</span><div><span style="font-family:trebuchet ms,sans-serif">skype: sylvio.jollenbeck<br>
<font size="1"><a href="http://www.hosannatecnologia.com.br/" target="_blank">www.hosannatecnologia.com.br</a></font></span><br><img src="http://www.hosannatecnologia.com.br/pixel.fw.png"><br></div></div>
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