[asterisk-commits] branch oej/test-this-branch r15150 - /team/oej/test-this-branch/

[asterisk-commits at lists.digium.com]

Author: oej
Date: Sun Mar 26 21:44:48 2006
New Revision: 15150

URL: http://svn.digium.com/view/asterisk?rev=15150&view=rev
Log:
update README

Modified:
    team/oej/test-this-branch/README.test-this-branch
    team/oej/test-this-branch/README.test-this-branch.html

Modified: team/oej/test-this-branch/README.test-this-branch
URL: http://svn.digium.com/view/asterisk/team/oej/test-this-branch/README.test-this-branch?rev=15150&r1=15149&r2=15150&view=diff
==============================================================================
--- team/oej/test-this-branch/README.test-this-branch (original)
+++ team/oej/test-this-branch/README.test-this-branch Sun Mar 26 21:44:48 2006
@@ -23,7 +23,6 @@
 This branch includes the following branches
 - sipdiversion: Additional support for the Diversion: header
 - jitterbuffer: Jitterbuffer for RTP in chan_sip (#3854, Securax/ZOA)
-- videosupport: Improved support for video (#5427, John Martin)
 - peermatch: New peer matching algorithm (#6612, oej)
 - rtcp: Improved support for RTCP (#2863, folsson/John Martin)
 - dialplan-ami-events: Report dialplan reload in manager (#5741, oej)
@@ -64,6 +63,7 @@
 - Fix race condition in voicemail (corydon76, #6714)
 - disable-ol-and-sub: Settings for disabling sip subscriptions and overlap 
   dialing (#6705, oej) - See configs/sip.conf.sample
+- videosupport: Improved support for video (#5427, John Martin)
 
 Coming here soon:
 - siptransfer: Improved SIP transfer support (branch)

Modified: team/oej/test-this-branch/README.test-this-branch.html
URL: http://svn.digium.com/view/asterisk/team/oej/test-this-branch/README.test-this-branch.html?rev=15150&r1=15149&r2=15150&view=diff
==============================================================================
--- team/oej/test-this-branch/README.test-this-branch.html (original)
+++ team/oej/test-this-branch/README.test-this-branch.html Sun Mar 26 21:44:48 2006
@@ -22,7 +22,6 @@
 <ul>
 <li> sipdiversion: Additional support for the Diversion: header<br />
 <li> jitterbuffer: Jitterbuffer for RTP in chan_sip (<a href="http://bugs.digium.com/view.php?id=3854">#3854</a>, Securax/ZOA)<br />
-<li> videosupport: Improved support for video (<a href="http://bugs.digium.com/view.php?id=5427">#5427</a>, John Martin)<br />
 <li> peermatch: New peer matching algorithm (<a href="http://bugs.digium.com/view.php?id=6612">#6612</a>, oej)<br />
 <li> rtcp: Improved support for RTCP (<a href="http://bugs.digium.com/view.php?id=2863">#2863</a>, folsson/John Martin)<br />
 <li> dialplan-ami-events: Report dialplan reload in manager (<a href="http://bugs.digium.com/view.php?id=5741">#5741</a>, oej)<br />
@@ -69,6 +68,7 @@
 <li> End CDR before 'h' extension (russellb, <a href="http://bugs.digium.com/view.php?id=6193">#6193</a>)<br />
 <li> disable-ol-and-sub: Settings for disabling sip subscriptions and overlap 
   dialing (<a href="http://bugs.digium.com/view.php?id=6705">#6705</a>, oej) - See <a href="http://svn.digium.com/view/asterisk/team/oej/test-this-branch/configs/sip.conf.sample?view=markup">configs/sip.conf.sample</a><br />
+<li> videosupport: Improved support for video (<a href="http://bugs.digium.com/view.php?id=5427">#5427</a>, John Martin)<br />
 </ul>
 
 <h3>Coming here soon:</h3>