Index: cgi-bin/set_voiplines.pl
===================================================================
--- cgi-bin/set_voiplines.pl	(revision 159)
+++ cgi-bin/set_voiplines.pl	(working copy)
@@ -52,11 +52,7 @@
 		print ";";
 	    }
 
-            # note final /$user seems to be necessary so that incoming VOIP
-	    # calls work.  I really don't get the syntax but messed around for
-	    # hours until it worked!
-
-	    print "register => $user\@$user/$user; $3\n";
+	    print "register => $user\@$user/s; $3\n";
 	}
 	else {
 	    # OK so this is a regular sip.conf line, just echo to stdout
Index: etc/asterisk/sip.conf
===================================================================
--- etc/asterisk/sip.conf	(revision 159)
+++ etc/asterisk/sip.conf	(working copy)
@@ -153,7 +153,7 @@
 
 ; register mini-asterisk voip line providers here
 
-;register => user@user/user ; mini-asterisk - do not change this comment
+;register => user@user/s ; mini-asterisk - do not change this comment
 
 ;----------------------------------------- NAT SUPPORT ------------------------
 ; The externip, externhost and localnet settings are used if you use Asterisk
@@ -592,6 +592,7 @@
 ;allow=ulaw,alaw
 ;qualify=yes
 ;nat=yes
+;context=incoming
 
 ;[user] ; "SIP" mini-asterisk do not remove this comment
 ;; No NAT router between your Phone system and your ITSP
@@ -604,6 +605,7 @@
 ;disallow=all
 ;allow=ulaw,alaw
 ;qualify=yes
+;context=incoming
 
 ;[user] ; "Jazmin" mini-asterisk do not remove this comment
 ;; <a href="http://www.jazmin.com.au/">Jazmin</a> are a South Australian ITSP
@@ -620,5 +622,40 @@
 ;dtmfmod=rfc2833                                                    
 ;qualify=yes                                                   
 ;canreinvite=no                                                     
+;context=incoming
 ;nat=yes
 
+;[user] ; "Nodephone" mini-asterisk do not remove this comment
+;; <a href="http://www.internode.on.net/residential/home_phone/nodephone/">Nodephone</a> (<a href="http://www.internode.on.net/">Internode</a>'s VoIP service)
+;type=peer
+;username=<username>
+;secret=<password>
+;host=sip.internode.on.net
+;dtmfmode=rfc2833
+;fromdomain=sip.internode.on.net
+;fromuser=<username>
+;insecure=port,invite
+;canreinvite=no
+;disallow=all
+;allow=ulaw,alaw
+;qualify=yes
+;incominglimit=2
+;context=incoming
+
+;[user] ; "Nodephone-NAT" mini-asterisk do not remove this comment
+;; <a href="http://www.internode.on.net/residential/home_phone/nodephone/">Nodephone</a> (<a href="http://www.internode.on.net/">Internode</a>'s VoIP service) via a NAT router
+;type=peer
+;username=<username>
+;secret=<password>
+;host=sip.internode.on.net
+;dtmfmode=rfc2833
+;fromdomain=sip.internode.on.net
+;fromuser=<username>
+;insecure=port,invite
+;canreinvite=no
+;disallow=all
+;allow=ulaw,alaw
+;qualify=yes
+;incominglimit=2
+;context=incoming
+;nat=yes
Index: etc/asterisk/extensions.conf
===================================================================
--- etc/asterisk/extensions.conf	(revision 159)
+++ etc/asterisk/extensions.conf	(working copy)
@@ -52,6 +52,7 @@
 
 exten => _1.,1,Dial(SIP/voip/${EXTEN:1}) 
 
+[incoming]
 ; Pre-configured incoming calls
 
 exten => s,1,Dial(SIP/6011) ;; mini-asterisk - don't remove this comment