Changelog for
asterisk-odbc-13.20.0-7.1.x86_64.rpm :
Tue Apr 3 14:00:00 2018 jamespAATTvicidial.com
- Update to Asterisk v.13.20.0
o See ChangeLog for more details
Sat Feb 3 13:00:00 2018 jamespAATTvicidial.com
- Added amd.patch to correct AMD issues
Mon Jan 22 13:00:00 2018 jamespAATTvicidial.com
- Update to Asterisk v.13.190.0
o See ChangeLog for more details
Sun Jan 7 13:00:00 2018 jamespAATTvicidial.com
- Update to Asterisk v.13.18.5
Sat Nov 11 13:00:00 2017 jamespAATTvicidial.com
- Remove old wait for silence patch
Sat Nov 11 13:00:00 2017 jamespAATTvicidial.com
- Update to Asterisk v.13.18.2
Wed Oct 11 14:00:00 2017 jamespAATTvicidial.com
- Updates to Asterisk v.13.17.2
- Added scripts for debugging
Fri Oct 6 14:00:00 2017 jamespAATTvicidial.com
- Enabled app_meetme conferencing engine
- Disabled native architecture building
Mon Sep 25 14:00:00 2017 jamespAATTvicidial.com
- Add short enter.h and leave.h meetme sounds for vicidial
- Remove language prefix and runuser/rungroup settings
- Append \'-vici\' to version string for console output
Sat Sep 10 14:00:00 2016 michaelAATTstroeder.com
- Update to new upstream maintenance release 13.11.2
Thu Jul 21 14:00:00 2016 michaelAATTstroeder.com
- Update to new upstream maintenance release 13.10.0
Fri May 13 14:00:00 2016 michaelAATTstroeder.com
- Update to new upstream maintenance release 13.9.1
Wed Mar 30 14:00:00 2016 michaelAATTstroeder.com
- Update to new upstream maintenance release 13.9.0
Sat Feb 6 13:00:00 2016 jengelhAATTinai.de
- Update to new upstream maintenance release 13.7.2
Fri Feb 5 13:00:00 2016 michaelAATTstroeder.com
- Update to new upstream maintenance release 13.7.1
with security fixes:
* AST-2016-001: BEAST vulnerability in HTTP server
* AST-2016-002: File descriptor exhaustion in chan_sip
* AST-2016-003: Remote crash vulnerability when receiving UDPTL FAX data.
- Added build dependencies:
* libv4l-devel
* libSDL2-devel
Thu Dec 10 13:00:00 2015 zawel1AATTgmail.com
- Update to new upstream maintenance release 13.6.0
Thu Aug 13 14:00:00 2015 jengelhAATTinai.de
- Update to new upstream maintenance release 13.5.0
Mon Jun 8 14:00:00 2015 jengelhAATTinai.de
- Update to new upstream maintenenace release 13.4.0
Thu Apr 9 14:00:00 2015 jengelhAATTinai.de
- Update to new upstream maintenance release 13.3.2
* fix for CVE-2015-3008 asterisk: TLS Certificate Common name NULL
byte exploit
Mon Mar 16 13:00:00 2015 jengelhAATTinai.de
- Update to new upstream maintenance release 13.2
Sat Jan 3 13:00:00 2015 jengelhAATTinai.de
- Update to new upstream maintenance release 13.1
Thu Nov 20 13:00:00 2014 joop.boonenAATTopensuse.org
- Build version 13.0.1
Thu Nov 20 13:00:00 2014 joop.boonenAATTopensuse.org
- Corrected the file paths
- Added missing files
- Added excludes
Mon Nov 17 13:00:00 2014 jengelhAATTinai.de
- Update to new upstream release 13
* Asterisk security events are now provided via AMI, allowing end
users to monitor their Asterisk system in real time for security
related issues.
* Both AMI and ARI now allow external systems to control the state
of a mailbox. Using AMI actions or ARI resources, external
systems can programmatically trigger Message Waiting Indicators
(MWI) on subscribed phones. This is of particular use to those
who want to build their own VoiceMail application using ARI.
* ARI now supports the reception/transmission of out of call text
messages using any supported channel driver/protocol stack
through ARI. Users receive out of call text messages as JSON
events over the ARI websocket connection, and can send out of
call text messages using HTTP requests.
* The PJSIP stack now supports RFC 4662 Resource Lists, allowing
Asterisk to act as a Resource List Server. This includes defining lists of presence state, mailbox state, or lists of presence state/mailbox state; managing subscriptions to lists; and batched delivery of NOTIFY requests to subscribers.
* The PJSIP stack can now be used as a means of distributing device
state or mailbox state via PUBLISH requests to other Asterisk
instances. This is analogous to Asterisk\'s clustering support
using XMPP or Corosync; unlike existing clustering mechanisms,
using the PJSIP stack to perform the distribution of state does
not rely on another daemon or server to perform the work.
Fri Aug 22 14:00:00 2014 jengelhAATTinai.de
- Update to new upstream maintenance release 12.5.0
* http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-12.5.0-summary.txt
Sun Jul 13 14:00:00 2014 jengelhAATTinai.de
- Update to new upstream release 12.4.0
* http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-12.4.0-summary.txt
- Reenable SS7 support in chan_dahdi (for libss7-2.0)
Fri Jun 27 14:00:00 2014 jengelhAATTinai.de
- Update to new upstream release 12.3.2
* http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-12.3.2-summary.txt
Wed Apr 23 14:00:00 2014 marcelloceschiaAATTusers.sourceforge.net
- Update to new upstream release 12.2.0
* http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-12.2.0-summary.txt
Sat Mar 22 13:00:00 2014 marcelloceschiaAATTusers.sourceforge.net
- Update to new upstream release 12.1.1 (security release)
* http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-12.1.1-summary.txt
Thu Mar 6 13:00:00 2014 jengelhAATTinai.de
- Update to new upstream release 12.1.0 (bugfix release)
* http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-12.1.0-summary.txt
Tue Dec 24 13:00:00 2013 jengelhAATTinai.de
- Update to new upstream release 12.0.0
* A more flexible bridging core based on the Bridging API
* A new internal message bus, Stasis
* Major standardization and consistency improvements to AMI
* Addition of the Asterisk REST Interface (ARI)
* A new SIP channel driver, chan_pjsip
* https://wiki.asterisk.org/wiki/display/AST/New+in+12
Tue Dec 24 13:00:00 2013 jengelhAATTinai.de
- Update to new upstream release 11.7.0 (bugfix release)
* See http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-11.7.0-summary.html
for details
Sun Nov 24 13:00:00 2013 jengelhAATTinai.de
- Update to new upstream release 11.6.0 (bugfix release)
* See http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-11.6.0-summary.html
for details
Thu Aug 15 14:00:00 2013 jengelhAATTinai.de
- Use libuuid to reenable res_rtp_asterisk
Thu Aug 8 14:00:00 2013 jengelhAATTinai.de
- Update to new upstream release 11.5.0 (bugfix release)
* See http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-11.5.0-summary.html
for details
Sun Jun 2 14:00:00 2013 jengelhAATTinai.de
- Update to new upstream release 11.4.0 (bugfix release)
* See http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-11.4.0-summary.html
for details
Sun Feb 17 13:00:00 2013 jengelhAATTinai.de
- Enable building res_corosync (replaces res_ais from asterisk-10)
- Order asterisk after network (bnc#796148)
Sat Feb 16 13:00:00 2013 jengelhAATTinai.de
- Enable building chan_ooh323
- Put config sample files into their respective subpackages
- Split off asterisk-freetds
- Make libasteriskssl.so symlink point to actual file
- Call ldconfig for libasteriskssl1
Thu Jan 24 13:00:00 2013 jengelhAATTinai.de
- Update to new upstream release 11.2.1 (bugfix release)
* Fixed stuck DTMF when using ChannelRedirect to split a two
channel bridge
* Asterisk deadlocked during startup with mutex errors
* Resolved segfault in chan_sip while performing connected line
update
Fri Dec 21 13:00:00 2012 joop.boonenAATTopensuse.org
- Update to new upstream release 11.1.0
* chan_local: Fix local_pvt ref leak in local_devicestate().
* Fix a SIP request memory leak with TLS connections.
* Fix a bug where our Motif ICE candidates were not quite proper,
and make us more forgiving.
Wed Dec 5 13:00:00 2012 joop.boonenAATTopensuse.org
- Update to new upstream release 11.0.1
* Fix a bug which made ConfBridge not record conferences when the
record command was initiated from AMI/CLI commands
* Fix a bug causing SIP reloads to remove all entries from the registry
* Fix an issue with res_http_websocket where the chan_sip WebSocket
handler could not be registered.
Sat Nov 3 13:00:00 2012 jengelhAATTinai.de
- Update to new upstream release 11.0.0
* WebRTC Support with WebSocket transport over SIP.
* DTLS-SRTP - A secure transport for RTP media streams used by
WebRTC and SIP endpoints.
* ICE, STUN and TURN – A set of related technologies for
establishing live media streams between software agents running
behind network address translators (NATs) and firewalls. ICE,
STUN and TURN have been incorporated into the Asterisk RTP engine.
Sun Apr 8 14:00:00 2012 jengelhAATTmedozas.de
- Update to new upstream release 10.3.0
* http://downloads.asterisk.org/pub/telephony/asterisk/old-releases/asterisk-10.3.0-summary.html
- Make /var/lib/asterisk writable, so that the sqlite db can
be automatically created
- Replace init script by something less convoluted;
also add a systemd service file (bnc#750762, bnc#750763)
Fri Mar 16 13:00:00 2012 jengelhAATTmedozas.de
- Update to new upstream release 10.2.1
* Fix AST-2012-002, AST-2012-003 security vulnerabilities
Sun Mar 11 13:00:00 2012 jengelhAATTmedozas.de
- Update to new upstream release 10.2.0
* http://downloads.asterisk.org/pub/telephony/asterisk/old-releases/asterisk-10.2.0-summary.html
- Restore spandsp support (bnc#731943)
- Set permissions on files (bnc#750761)
Wed Feb 1 13:00:00 2012 jengelhAATTmedozas.de
- Update to new upstream release 10.1.0
* http://downloads.asterisk.org/pub/telephony/asterisk/old-releases/asterisk-10.1.0-summary.html
- Add autotools BuildRequires for factory/12.2
Fri Dec 16 13:00:00 2011 jengelhAATTmedozas.de
- Update to final 10.0.0
Sat Oct 8 14:00:00 2011 jengelhAATTmedozas.de
- New package, for a change list see
https://wiki.asterisk.org/wiki/display/AST/New+in+10