git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@2905 65c4cc65-6c06-0410-ace0-fbb531ad65f3
130 lines
4.4 KiB
Text
Executable file
130 lines
4.4 KiB
Text
Executable file
;
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; SIP Configuration for Asterisk
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;
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; Syntax for specifying a SIP device in extensions.conf is
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; SIP/devicename where devicename is defined in a section below.
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;
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; You may also use
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; SIP/username@domain to call any SIP user on the Internet
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; (Don't forget to enable DNS SRV records if you want to use this)
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;
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; If you define a SIP proxy as a peer below, you may call
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; SIP/proxyhostname/user or SIP/user@proxyhostname
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; where the proxyhostname is defined in a section below
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;
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; Useful CLI commands to check peers/users:
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; sip show peers Show all SIP peers (including friends)
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; sip show users Show all SIP users (including friends)
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; sip show registry Show status of hosts we register with
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;
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; sip debug Show all SIP messages
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;
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[general]
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port = 5060 ; Port to bind to
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bindaddr = 0.0.0.0 ; Address to bind SIP channel to
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context = default ; Default context for incoming calls
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;srvlookup = yes ; Enable DNS SRV lookups on outbound calls
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; Asterisk only uses the first host in SRV records
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;pedantic = yes ; Enable slow, pedantic checking for Pingtel
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; and multiline formatted headers for strict
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; SIP compatibility
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;tos=lowdelay ; IP QoS parameter, either keyword or value
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; like tos=184
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;maxexpirey=3600 ; Max length of incoming registration we allow
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;realm=asterisk ; Our global authentication realm
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;defaultexpirey=120 ; Default length of incoming/outoing registration
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;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY
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;videosupport=yes ; Turn on support for SIP video
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;disallow=all ; Disallow all codecs
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;allow=ulaw ; Allow codecs in order of preference
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;allow=ilbc
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; Asterisk can register as a SIP user agent to a SIP proxy (provider)
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; Format for the register statement is:
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; register => user[:secret[:authuser]]@host[:port][/extension]
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;
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; If no extension is given, the 's' extension is used. The extension
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; needs to be defined in extensions.conf to be able to accept calls
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; from this SIP proxy (provider)
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;
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; host is either a host name defined in DNS or the name of a
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; section defined below.
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;
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; Examples:
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;
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;register => 1234:password@mysipprovider.com
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;
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; Will call to the 's' extension
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;
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;
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;register => 2345@mysipprovider.com/1234
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;
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; Register 2345 at sip provider. Calls from this provider connect to local
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; extension 1234 in extensions.conf default context, unless you define
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; [mysipprovider.com] in a section below, and configure a context
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;externip = 200.201.202.203 ; Address that we're going to put in outbound SIP messages
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; if we're behind a NAT
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;localnet = 192.168.1.0 ; Internal NETWORK address
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;localmask = 255.255.255.0 ; Internal netmask
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; The externip, localnet and localmask is used
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; when registering and communication with other proxies
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; that we're registered with
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;[snomsip]
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;type=friend
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;secret=blah
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;host=dynamic
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;dtmfmode=inband ; Choices are inband, rfc2833, or info
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;defaultip=192.168.0.59
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;mailbox=1234,2345 ; Mailbox for message waiting indicator
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;restrictcid=yes ; To have the callerid restriced -> sent as ANI
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;insecure=yes ; To match a peer based by IP address only and not peer
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;insecure=very ; To allow registered hosts to call without re-authenticating
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;[pingtel]
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;type=friend
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;username=pingtel
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;secret=blah
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;host=dynamic
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;qualify=1000 ; Consider it down if it's 1 second to reply
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; Helps with NAT session
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; qualify=yes uses default value
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;callgroup=1,3-4
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;pickupgroup=1,3-4
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;defaultip=192.168.0.60
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;[cisco]
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;type=friend
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;username=cisco
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;secret=blah
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;nat=yes ; This phone may be natted
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; Use IP address that packet is received from
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; instead of trusting SIP headers
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;host=dynamic
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;canreinvite=no ; Asterisk by default tries to redirect the
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; RTP media stream (audio) to go directly from
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; the caller to the callee. Some devices do not
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; support this (especially if one of them is
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; behind a NAT).
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;qualify=200 ; Qualify peer is no more than 200ms away
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;defaultip=192.168.0.4
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;[cisco1]
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;type=friend
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;username=cisco1
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;fromuser=markster ; Specify user to put in "from" instead of callerid
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;fromdomain=yourdomain.com ; Specify domain to put in "from" instead of callerid
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; fromuser and fromdomain are used when Asterisk
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; places calls to this account. It is not used for
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; calls from this account.
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;secret=blah
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;host=dynamic
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;defaultip=192.168.0.4
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;amaflags=default ; Choices are default, omit, billing, documentation
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;accountcode=markster ; Users may be associated with an accountcode to ease billing
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