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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@4372 65c4cc65-6c06-0410-ace0-fbb531ad65f3
291 lines
12 KiB
Plaintext
Executable File
291 lines
12 KiB
Plaintext
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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context=default ; Default context for incoming calls
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;recordhistory=yes ; Record SIP history by default
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; (see sip history / sip no history)
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;realm=mydomain.tld ; Realm for digest authentication
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; defaults to "asterisk"
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; Realms MUST be globally unique according to RFC 3261
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; Set this to your host name or domain name
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port=5060 ; UDP Port to bind to (SIP standard port is 5060)
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bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
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srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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; Note: Asterisk only uses the first host
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; in SRV records
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; Disabling DNS SRV lookups disables the
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; ability to place SIP calls based on domain
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; names to some other SIP users on the Internet
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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 (defaults to "no")
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;tos=184 ; Set IP QoS to either a keyword or numeric val
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;tos=lowdelay ; lowdelay,throughput,reliability,mincost,none
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;maxexpirey=3600 ; Max length of incoming registration we allow
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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 MWI NOTIFY
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;checkmwi=10 ; Default time between mailbox checks for peers
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;videosupport=yes ; Turn on support for SIP video
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;disallow=all ; First disallow all codecs
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;allow=ulaw ; Allow codecs in order of preference
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;allow=ilbc ; Note: codec order is respected only in [general]
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;musicclass=default ; Sets the default music on hold class for all SIP calls
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; This may also be set for individual users/peers
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;language=en ; Default language setting for all users/peers
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; This may also be set for individual users/peers
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;relaxdtmf=yes ; Relax dtmf handling
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;rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity
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; when we're not on hold
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;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP activity
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; when we're on hold (must be > rtptimeout)
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;trustrpid = no ; If Remote-Party-ID should be trusted
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;progressinband=no ; If we should generate in-band ringing always
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; use 'never' to never use in-band signalling, even in cases
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; where some buggy devices might not render it
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;useragent=Asterisk PBX ; Allows you to change the user agent string
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;nat=no ; NAT settings
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; yes = Always ignore info and assume NAT
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; no = Use NAT mode only according to RFC3581
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; never = Never attempt NAT mode or RFC3581 support
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; route = Assume NAT, don't send rport
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; (work around more UNIDEN bugs)
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;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
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; Note that promiscredir when redirects are made to the
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; local system will cause loops since SIP is incapable
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;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
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; a valid phone number
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; of performing a "hairpin" call.
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;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
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; Other options:
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; info : SIP INFO messages
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; inband : Inband audio
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;compactheaders = yes ; send compact sip headers.
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;
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; If regcontext is specified, Asterisk will dynamically
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; create and destroy a NoOp priority 1 extension for a given
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; peer who registers or unregisters with us. The actual extension
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; is the 'regexten' parameter of the registering peer or its
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; name if 'regexten' is not provided. More than one regexten may be supplied
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; if they are separated by '&'. Patterns may be used in regexten.
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;
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;regcontext=sipregistrations
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;
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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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; This will pass incoming calls to the 's' extension
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;
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;
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;register => 2345:password@sip_proxy/1234
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;
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; Register 2345 at sip provider 'sip_proxy'. 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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; unless you configure a [sip_proxy] section below, and configure a context.
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; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
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; Tip 2: Use separate type=peer and type=user sections for SIP providers
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; (instead of type=friend) if you have calls in both directions
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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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; The externip and localnet is used
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; when registering and communicating with other proxies
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; that we're registered with
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; You may add multiple local networks. A reasonable set of defaults
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; are:
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;localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
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;localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
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;localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
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;localnet=169.254.0.0/255.255.0.0 ;Zero conf local network
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;-----------------------------------------------------------------------------------
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; Users and peers have different settings available. Friends have all settings,
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; since a friend is both a peer and a user
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;
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; User config options: Peer configuration:
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; -------------------- -------------------
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; context context
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; permit permit
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; deny deny
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; auth auth
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; secret secret
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; md5secret md5secret
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; dtmfmode dtmfmode
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; canreinvite canreinvite
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; nat nat
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; callgroup callgroup
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; pickupgroup pickupgroup
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; language language
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; allow allow
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; disallow disallow
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; insecure insecure
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; trustrpid trustrpid
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; progressinband progressinband
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; promiscredir promiscredir
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; useclientcode useclientcode
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; setvar
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; callerid
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; accountcode
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; amaflags
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; incominglimit
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; restrictcid
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; mailbox
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; username
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; template
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; fromdomain
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; regexten
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; fromuser
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; host
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; mask
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; port
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; qualify
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; defaultip
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; rtptimeout
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; rtpholdtimeout
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;[sip_proxy]
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; For incoming calls only. Example: FWD (Free World Dialup)
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;type=user
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;context=from-fwd
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;[sip_proxy-out]
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;type=peer ; we only want to call out, not be called
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;secret=guessit
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;username=yourusername ; Authentication user for outbound proxies
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;fromuser=yourusername ; Many SIP providers require this!
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;host=box.provider.com
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;usereqphone=yes ; This provider requires ";user=phone" on URI
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;[grandstream1]
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;type=friend ; either "friend" (peer+user), "peer" or "user"
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;context=from-sip
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;fromuser=grandstream1 ; overrides the callerid, e.g. required by FWD
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;callerid=John Doe <1234>
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;host=192.168.0.23 ; we have a static but private IP address
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;nat=no ; there is not NAT between phone and Asterisk
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;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk
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;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
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;incominglimit=1 ; permit only 1 outgoing call at a time
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; from the phone to asterisk
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;mailbox=1234@default ; mailbox 1234 in voicemail context "default"
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;disallow=all ; need to disallow=all before we can use allow=
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;allow=ulaw ; Note: In user sections the order of codecs
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; listed with allow= does NOT matter!
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;allow=alaw
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;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
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;allow=g729 ; Pass-thru only unless g729 license obtained
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;[xlite1]
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;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
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;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
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;type=friend
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;regexten=1234 ; When they register, create extension 1234
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;username=xlite1
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;callerid="Jane Smith" <5678>
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;host=dynamic
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;nat=yes ; X-Lite is behind a NAT router
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;canreinvite=no ; Typically set to NO if behind NAT
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;disallow=all
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;allow=gsm ; GSM consumes far less bandwidth than ulaw
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;allow=ulaw
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;allow=alaw
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;[snom]
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;type=friend ; Friends place calls and receive calls
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;context=from-sip ; Context for incoming calls from this user
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;secret=blah
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;language=de ; Use German prompts for this user
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;host=dynamic ; This peer register with us
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;dtmfmode=inband ; Choices are inband, rfc2833, or info
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;defaultip=192.168.0.59 ; IP used until peer registers
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;username=snom ; Username to use in INVITE until peer registers
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;mailbox=1234,2345 ; Mailboxes for message waiting indicator
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;restrictcid=yes ; To have the callerid restriced -> sent as ANI
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;disallow=all
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;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
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;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator
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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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;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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;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 ; We are in caller groups 1,3,4
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;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5
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;defaultip=192.168.0.60 ; IP address to use if peer has not registred
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;[cisco1]
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;type=friend
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;username=cisco1
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;secret=blah
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;qualify=200 ; Qualify peer is no more than 200ms away
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;nat=yes ; This phone may be natted
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; Send SIP and RTP to IP address that packet is
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; received from instead of trusting SIP headers
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;host=dynamic ; This device registers with us
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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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;defaultip=192.168.0.4
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;[cisco2]
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;type=friend
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;username=cisco2
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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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