Changelog for asterisk-odbc-11.5.0-2.11.i586.rpm :
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