[asterisk-commits] oej: branch oej/earl-grey-sip2cause-configurable-1.8 r376899 - /team/oej/earl...

[SVN commits to the Asterisk project]

Author: oej
Date: Fri Nov 30 02:56:44 2012
New Revision: 376899

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=376899
Log:
Clarify documentation. Scare people away. 

Modified:
    team/oej/earl-grey-sip2cause-configurable-1.8/configs/sip2cause.conf.sample

Modified: team/oej/earl-grey-sip2cause-configurable-1.8/configs/sip2cause.conf.sample
URL: http://svnview.digium.com/svn/asterisk/team/oej/earl-grey-sip2cause-configurable-1.8/configs/sip2cause.conf.sample?view=diff&rev=376899&r1=376898&r2=376899
==============================================================================
--- team/oej/earl-grey-sip2cause-configurable-1.8/configs/sip2cause.conf.sample (original)
+++ team/oej/earl-grey-sip2cause-configurable-1.8/configs/sip2cause.conf.sample Fri Nov 30 02:56:44 2012
@@ -1,7 +1,14 @@
 ;
 ; Conversion table additions for the translation between SIP hangup codes and ISDN cause codes
+; ============================================================================================
+; This table is used to convert a negative SIP response to an ISDN cause code, that is sent
+; over the PBX call bridge to the calling party, or to convert a received ISDN hangup cause
+; from the called party to a SIP response. Changes will not affect how Asterisk treats a SIP
+; response, but how the other party reacts to it.
+;
 ; For reference, check Asterisk source code include/asterisk/causes.h
 ; and channels/sip/sip2cause.c
+;
 ; Note also that this translation is not used in all cases in the SIP implementation in Asterisk.
 ; Fixes are always welcome.
 ;
@@ -11,7 +18,7 @@
 ;       In 99.99% of Asterisk installations, you will not need to add ANYTHING to this file.
 ;
 [sip2cause]
-; 200 => AST_CAUSE_NORMAL_CLEARING
+; 404 => USER_BUSY
 
 [cause2sip]
-; AST_CAUSE_NORMAL_CLEARING => 200 All right, dude
+; USER_BUSY => 403 Forbidden