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)
;[EMAIL PROTECTED] ; 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
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