[asterisk-commits] file: branch file/chan_jingle2 r365685 - /team/file/chan_jingle2/configs/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Tue May 8 14:01:00 CDT 2012


Author: file
Date: Tue May  8 14:00:56 2012
New Revision: 365685

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=365685
Log:
Add comment about caller id.

Modified:
    team/file/chan_jingle2/configs/jingle2.conf.sample

Modified: team/file/chan_jingle2/configs/jingle2.conf.sample
URL: http://svnview.digium.com/svn/asterisk/team/file/chan_jingle2/configs/jingle2.conf.sample?view=diff&rev=365685&r1=365684&r2=365685
==============================================================================
--- team/file/chan_jingle2/configs/jingle2.conf.sample (original)
+++ team/file/chan_jingle2/configs/jingle2.conf.sample Tue May  8 14:00:56 2012
@@ -49,6 +49,14 @@
 ;
 ; The only supported method for DTMF is RFC2833. This is always enabled on audio streams and negotiated if possible.
 
+; CallerID
+;
+; The incoming caller id number is populated with the username of the caller and the name is populated with the full
+; identity of the caller. If you would like to perform authentication or filtering of incoming calls it is recommended
+; that you use these fields to do so.
+;
+; Outgoing caller id is *not* settable.
+
 ; Default template for endpoints, to be included in their definition
 [default](!)
 disallow=all ;




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