Index: configs/sip.conf.sample
===================================================================
RCS file: /usr/cvsroot/asterisk/configs/sip.conf.sample,v
retrieving revision 1.36
diff -u -r1.36 sip.conf.sample
--- configs/sip.conf.sample	29 Jun 2004 14:44:29 -0000	1.36
+++ configs/sip.conf.sample	27 Jul 2004 17:54:12 -0000
@@ -135,7 +135,6 @@
 ; accountcode
 ; amaflags
 ; incominglimit
-; outgoinglimit
 ; restrictcid
 ;                             mailbox
 ;                             username
@@ -172,8 +171,8 @@
 ;nat=no                        ; there is not NAT between phone and Asterisk
 ;canreinvite=yes               ; allow RTP voice traffic to bypass Asterisk
 ;dtmfmode=info                 ; either RFC2833 or INFO for the BudgeTone
-;outgoinglimit=1               ; disable callwaiting signal (2nd call to phone)
 ;incominglimit=1               ; permit only 1 outgoing call at a time
+			       ; from the phone to asterisk
 ;mailbox=1234@default  ; mailbox 1234 in voicemail context "default"
 ;disallow=all                  ; need to disallow=all before we can use allow=
 ;allow=ulaw                    ; Note: In user sections the order of codecs