[Asterisk-cvs] asterisk/configs sip.conf.sample,1.66,1.67

[kpfleming]

Update of /usr/cvsroot/asterisk/configs
In directory mongoose.digium.com:/tmp/cvs-serv22548/configs

Modified Files:
	sip.conf.sample 
Log Message:
remove unused 'outgoinglimit' code, rename 'incominglimit' to 'call-limit' (old syntax is still supported) (issue #5068)


Index: sip.conf.sample
===================================================================
RCS file: /usr/cvsroot/asterisk/configs/sip.conf.sample,v
retrieving revision 1.66
retrieving revision 1.67
diff -u -d -r1.66 -r1.67
--- sip.conf.sample	25 Aug 2005 02:25:30 -0000	1.66
+++ sip.conf.sample	30 Aug 2005 21:26:33 -0000	1.67
@@ -234,7 +234,7 @@
 ; setvar                      setvar
 ; callerid		      callerid
 ; amaflags		      amaflags
-; incominglimit		      incominglimit
+; call-limit		      call-limit
 ; restrictcid		      restrictcid
 ;                             mailbox
 ;                             username
@@ -266,6 +266,7 @@
 ;fromdomain=provider.sip.domain	
 ;host=box.provider.com
 ;usereqphone=yes		; This provider requires ";user=phone" on URI
+;call-limit=5			; permit only 5 simultaneous outgoing calls to this peer
 
 ;------------------------------------------------------------------------------
 ; Definitions of locally connected SIP phones
@@ -290,8 +291,11 @@
 ;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
-;incominglimit=1		; permit only 1 outgoing call at a time
+;call-limit=1			; permit only 1 outgoing call and 1 incoming call at a time
 				; from the phone to asterisk
+				; (1 for the explicit peer, 1 for the explicit user,
+				; remember that a friend equals 1 peer and 1 user in
+				; memory)
 ;mailbox=1234 at 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