[asterisk-commits] oej: branch oej/earl-grey-sip2cause-configurable-1.8 r402116 - in /team/oej/e...

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Mon Oct 28 10:07:15 CDT 2013


Author: oej
Date: Mon Oct 28 10:07:08 2013
New Revision: 402116

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=402116
Log:
Adding some documentation

Modified:
    team/oej/earl-grey-sip2cause-configurable-1.8/configs/sip2cause.conf.sample
    team/oej/earl-grey-sip2cause-configurable-1.8/include/asterisk/causes.h

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=402116&r1=402115&r2=402116
==============================================================================
--- 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 Mon Oct 28 10:07:08 2013
@@ -20,5 +20,10 @@
 [sip2cause]
 ; 404 => USER_BUSY
 
+; To convert a SIP2SIP call with a private error code, use CUSTOM1 - CUSTOM5 cause codes
+;497 => CUSTOM1
+
 [cause2sip]
 ; USER_BUSY => 403 Forbidden
+
+;CUSTOM1 => 497 IAX2 not supported.

Modified: team/oej/earl-grey-sip2cause-configurable-1.8/include/asterisk/causes.h
URL: http://svnview.digium.com/svn/asterisk/team/oej/earl-grey-sip2cause-configurable-1.8/include/asterisk/causes.h?view=diff&rev=402116&r1=402115&r2=402116
==============================================================================
--- team/oej/earl-grey-sip2cause-configurable-1.8/include/asterisk/causes.h (original)
+++ team/oej/earl-grey-sip2cause-configurable-1.8/include/asterisk/causes.h Mon Oct 28 10:07:08 2013
@@ -90,6 +90,10 @@
 	- AST_CAUSE_CUSTOM5                        119
 	- AST_CAUSE_INTERWORKING                   127
 
+The custom codes (115-119) are for sip2cause translations outside of the normal translation path.
+Only use these in sip2sip situations as ISDN equipment or other channel drivers will not understand
+them at all.
+
 For more information:
 - \ref app_dial.c
 */




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