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Bug # 1013: More explanation in the sip.conf.sample thanks to oej 

git-svn-id: http://svn.digium.com/svn/asterisk/trunk@2476 f38db490-d61c-443f-a65b-d21fe96a405b
This commit is contained in:
malcolmd 2004-03-19 20:30:03 +00:00
parent 66adba44e8
commit e6b720f4f4
1 changed files with 56 additions and 14 deletions
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70
configs/sip.conf.sample
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@ -1,28 +1,57 @@
;
; SIP Configuration for Asterisk
;
; Syntax for specifying a SIP device in extensions.conf is
; SIP/devicename where devicename is defined in a section below.
;
; You may also use
; SIP/username@domain to call any SIP user on the Internet
; (Don't forget to enable DNS SRV records if you want to use this)
;
; If you define a SIP proxy as a peer below, you may call
; SIP/proxyhostname/user or SIP/user@proxyhostname
; where the proxyhostname is defined in a section below
;
; Useful CLI commands to check peers/users:
; sip show peers Show all SIP peers (including friends)
; sip show users Show all SIP users (including friends)
; sip show registry Show status of hosts we register with
;
; sip debug Show all SIP messages
;
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
;externip = 200.201.202.203 ; Address that we're going to put in SIP messages if we're behind a NAT
;localnet = 192.168.1.0 ; Internal NETWORK address
;localmask = 255.255.255.0 ; Internal netmask
context = default ; Default for incoming calls
;srvlookup = yes ; Enable SRV lookups on outbound calls
bindaddr = 0.0.0.0 ; Address to bind SIP channel to
context = default ; Default context for incoming calls
;srvlookup = yes ; Enable DNS SRV lookups on outbound calls
; Asterisk only uses the first host in SRV records
;pedantic = yes ; Enable slow, pedantic checking for Pingtel
;tos=lowdelay
;tos=184
;tos=lowdelay ; IP QoS parameter, either keyword or value
; like tos=184
;maxexpirey=3600 ; Max length of incoming registration we allow
;defaultexpirey=120 ; Default length of incoming/outoing registration
;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY
;videosupport=yes ; Turn on support for SIP video
;disallow=all ; Disallow all codecs
;allow=ulaw ; Allow codecs in order of preference
;allow=ilbc
;
;register => 1234@mysipprovider.com ; Register with a SIP provider
;register => 2345@mysipprovider.com/1234 ; Register 2345 at sip provider as 1234 here.
;
;register => 1234:password@mysipprovider.com
;Register with a SIP provider
;register => 2345@mysipprovider.com/1234
;Register 2345 at sip provider. Calls from this provider connect to local extension 1234 in extensions.conf.
;externip = 200.201.202.203 ; Address that we're going to put in outbound SIP messages
; if we're behind a NAT
;localnet = 192.168.1.0 ; Internal NETWORK address
;localmask = 255.255.255.0 ; Internal netmask
; The externip, localnet and localmask is used
; when registering and communication with other proxies
; that we're registred with
;[snomsip]
;type=friend
;secret=blah
@ -38,6 +67,9 @@ context = default ; Default for incoming calls
;secret=blah
;host=dynamic
;qualify=1000 ; Consider it down if it's 1 second to reply
; Helps with NAT session
; qualify=yes uses default value
;callgroup=1,3-4
;pickupgroup=1,3-4
;defaultip=192.168.0.60
@ -47,8 +79,14 @@ context = default ; Default for incoming calls
;username=cisco
;secret=blah
;nat=yes ; This phone may be natted
; Use IP address that packet is received from
; instead of trusting SIP headers
;host=dynamic
;canreinvite=no ; Cisco poops on reinvite sometimes
;canreinvite=no ; Asterisk by default tries to redirect the
; RTP media stream (audio) to go directly from
; the caller to the callee. Some devices do not
; support this (especially if one of them is
; behinda a NAT).
;qualify=200 ; Qualify peer is no more than 200ms away
;defaultip=192.168.0.4
@ -56,8 +94,12 @@ context = default ; Default for incoming calls
;type=friend
;username=cisco1
;fromuser=markster ; Specify user to put in "from" instead of callerid
;fromdomain=yourdomain.com ; Specify domain to put in "from" instead of callerid
; fromuser and fromdomain are used when Asterisk
; places calls to this account. It is not used for
; calls from this account.
;secret=blah
;host=dynamic
;defaultip=192.168.0.4
;amaflags=default ; Choices are default, omit, billing, documentation
;accountcode=markster ; Users may be associated with an accountcode tp ease billing
;accountcode=markster ; Users may be associated with an accountcode to ease billing