Merged revisions 171838 via svnmerge from

https://origsvn.digium.com/svn/asterisk/trunk

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r171838 | oej | 2009-01-28 14:11:44 +0100 (Ons, 28 Jan 2009) | 10 lines

Merged revisions 171837 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r171837 | oej | 2009-01-28 14:07:27 +0100 (Ons, 28 Jan 2009) | 2 lines

Add a better explanation of the difference between the device namespace and the dialplan for newbies.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@171857 65c4cc65-6c06-0410-ace0-fbb531ad65f3

configs/sip.conf.sample

@@ -51,6 +51,25 @@
 ;   module reload chan_sip.so    Reload configuration file
 ;                                Active SIP peers will not be reconfigured
 ;
+;------- Naming devices ------------------------------------------------------
+;
+; When naming devices, make sure you understand how Asterisk matches calls
+; that come in.
+;	1. Asterisk checks the SIP From: address username and matches against
+;	   names of devices with type=user 
+;	   The name is the text between square brackets [name]
+;	2. Asterisk checks the IP address (and port number) that the INVITE
+;	   was sent from and matches against any devices with type=peer
+;
+; Don't mix extensions with the names of the devices. Devices need a unique
+; name. The device name is *not* used as phone numbers. Phone numbers are
+; anything you declare as an extension in the dialplan (extensions.conf).
+;
+; Note: The parameter "username" is not the username and in most cases is
+;       not needed at all. Check below. In later releases, it's renamed
+;       to "defaultuser" which is a better name, since it is used in 
+;       combination with the "defaultip" setting.
+;-----------------------------------------------------------------------------
 
 ; ** Deprecated configuration options **
 ; The "call-limit" configuation option is deprecated. It still works in