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remove unused 'outgoinglimit' code, rename 'incominglimit' to 'call-limit' (old syntax is still supported) (issue #5068)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6458 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Kevin P. Fleming
2005-08-30 21:26:33 +00:00
parent 6d19c704b7
commit 5f07eec58a
2 changed files with 68 additions and 126 deletions
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8
configs/sip.conf.sample
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@@ -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