From 5f07eec58a4b2be269262bda8510dd0552d02dba Mon Sep 17 00:00:00 2001 From: "Kevin P. Fleming" Date: Tue, 30 Aug 2005 21:26:33 +0000 Subject: remove unused 'outgoinglimit' code, rename 'incominglimit' to 'call-limit' (old syntax is still supported) (issue #5068) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6458 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- configs/sip.conf.sample | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) (limited to 'configs/sip.conf.sample') diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index 2eaba6c3f..23cc922f2 100755 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -234,7 +234,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; setvar setvar ; callerid callerid ; amaflags amaflags -; incominglimit incominglimit +; call-limit call-limit ; restrictcid restrictcid ; mailbox ; username @@ -266,6 +266,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;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 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;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@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 -- cgit v1.2.3