[asterisk-commits] branch oej/test-this-branch r14471 - /team/oej/test-this-branch/

[asterisk-commits at lists.digium.com]

Author: oej
Date: Thu Mar 23 14:55:48 2006
New Revision: 14471

URL: http://svn.digium.com/view/asterisk?rev=14471&view=rev
Log:
Update docs for rtptiming patch

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=14471&r1=14470&r2=14471&view=diff
==============================================================================
--- team/oej/test-this-branch/README.test-this-branch (original)
+++ team/oej/test-this-branch/README.test-this-branch Thu Mar 23 14:55:48 2006
@@ -57,13 +57,13 @@
 - Manager playDTMF command (squinky, #6682) 
   (Note: I changed the name in this version...)
 - G.722 support in Asterisk (passthrough, formats) (andrew, #5084)
-- Fix race condition in voicemail (corydon76, #6714)
 - TOUPPER and TOLOWER ASCII functions (powerkill, #6668)
   (With some changes)
 
 Things that has been commited to svn trunk:
 - Abandon Queue manager event (tim_ringenbach, #6459)
 - End CDR before 'h' extension (russellb, #6193)
+- Fix race condition in voicemail (corydon76, #6714)
 
 Coming here soon:
 - siptransfer: Improved SIP transfer support (branch)
@@ -78,7 +78,9 @@
 
 * Open BUGS (fixes welcome!)
 ----------------------------
-- ???
+- sipregister: We still do not match peer directly on incoming
+  calls from registered peer.
+  Work is going on in the sipregister branch to fix this.
 
 * Metermaids
 ------------
@@ -173,6 +175,19 @@
 be ended, but still present in the 'h' extension so that all values can
 still be read from it.
 
+* RTPtiming: Clocked RTP stream
+-------------------------------
+The work in the rtptiming branch tries to fix the problem with RTP being
+timed on incoming packets, thus not supporting phones that implement
+silence suppression. With this patch, phones with silence suppression
+will still hear audio, or muted phones may still listen in to meetme
+conferences without sending any audio.
+ 
+You enable this functionality in asterisk.conf
+
+It's a patch that needs more testing and is widely discussed in the
+developer community. (#6374)
+
 * 6251: Support for HDLC mode in ZAP channels
 ----------------------------------------------
 The patch adds 2 things to chan_zap:

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=14471&r1=14470&r2=14471&view=diff
==============================================================================
--- team/oej/test-this-branch/README.test-this-branch.html (original)
+++ team/oej/test-this-branch/README.test-this-branch.html Thu Mar 23 14:55:48 2006
@@ -174,6 +174,18 @@
 be ended, but still present in the 'h' extension so that all values can
 still be read from it.</p>
 
+<h4>RTPtiming: Clocked RTP stream</h4>
+<p>The work in the rtptiming branch tries to fix the problem with RTP being
+timed on incoming packets, thus not supporting phones that implement
+silence suppression. With this patch, phones with silence suppression
+will still hear audio, or muted phones may still listen in to meetme
+conferences without sending any audio.</p>
+ 
+<p>You enable this functionality in asterisk.conf</p>
+
+<p><i>It's a patch that needs more testing and is widely discussed in the
+developer community. (#6374)</i></p>
+
 <h4>6251: Support for HDLC mode in ZAP channels</h4><p>
 The patch adds 2 things to chan_zap:</p>
 <p>