[svn-commits] oej: trunk r229607 - /trunk/configs/sip.conf.sample

[SVN commits to the Digium repositories]

Author: oej
Date: Thu Nov 12 04:24:20 2009
New Revision: 229607

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=229607
Log:
Clarification

Modified:
    trunk/configs/sip.conf.sample

Modified: trunk/configs/sip.conf.sample
URL: http://svnview.digium.com/svn/asterisk/trunk/configs/sip.conf.sample?view=diff&rev=229607&r1=229606&r2=229607
==============================================================================
--- trunk/configs/sip.conf.sample (original)
+++ trunk/configs/sip.conf.sample Thu Nov 12 04:24:20 2009
@@ -87,12 +87,12 @@
 ;       combination with the "defaultip" setting.
 ;-----------------------------------------------------------------------------
 
-; ** Deprecated configuration options **
-; The "call-limit" configuation option is deprecated. It still works in
-; this version of Asterisk, but will disappear in the next version.
+; ** Old configuration options **
+; The "call-limit" configuation option is considered old is replaced
+; by new functionality. To enable callcounters, you use the new 
+; "callcounter" setting (for extension states in queue and subscriptions)
 ; You are encouraged to use the dialplan groupcount functionality
 ; to enforce call limits instead of using this channel-specific method.
-;
 ; You can still set limits per device in sip.conf or in a database by using
 ; "setvar" to set variables that can be used in the dialplan for various limits.