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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
376 lines
16 KiB
Plaintext
Executable File
376 lines
16 KiB
Plaintext
Executable File
;
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; SIP Configuration example 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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; reload chan_sip.so Reload configuration file
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; Active SIP peers will not be reconfigured
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;
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[general]
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context=default ; Default context for incoming calls
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;allowguest=no ; Allow or reject guest calls (default is yes, this can also be set to 'osp'
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; if asterisk was compiled with OSP support.
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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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bindport=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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;vmexten=voicemail ; dialplan extension to reach mailbox sets the
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; Message-Account in the MWI notify message
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; defaults to "asterisk"
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;videosupport=yes ; Turn on support for SIP video
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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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;disallow=all ; First 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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;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=never ; 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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;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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; of performing a "hairpin" call.
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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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;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 (requires 64 kbit codec -alaw, ulaw)
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;compactheaders = yes ; send compact sip headers.
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;sipdebug = yes ; Turn on SIP debugging by default, from
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; the moment the channel loads this configuration
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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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;registertimeout=20 ; retry registration calls every 20 seconds (default)
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;registerattempts=10 ; Number of registration attempts before we give up
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; 0 = continue forever, hammering the other server until it
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; accepts the registration
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; Default is 10 tries
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;callevents=no ; generate manager events when sip ua performs events (e.g. hold)
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;---------------------------------------------- NAT SUPPORT ------------------------
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; The externip, externhost and localnet settings are used if you use Asterisk behind
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; a NAT device to communicate with services on the outside.
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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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;externhost=foo.dyndns.net ; Alternatively you can specify an
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; external host, and Asterisk will
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; perform DNS queries periodically. Not
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; recommended for production
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; environments! Use externip instead
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;externrefresh=10 ; How often to refresh externhost if
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; used
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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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; The nat= setting is used when Asterisk is on a public IP, communicating with
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; devices hidden behind a NAT device (broadband router).
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; If you have one-way audio problems, you usually have problems with your NAT
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; configuration or your firewalls support of SIP+RTP ports.
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; You configure Asterisk choice of RTP ports for incoming audio in rtp.conf
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;
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;nat=no ; Global NAT settings (Affects all peers and users)
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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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;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
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; just like friends added from the config file only on a
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; as-needed basis.
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;rtnoupdate=yes ; do not send the update request over realtime.
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;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
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; as if it had just registered when the registration expires
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; the friend will vanish from the configuration until requested
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; again. If set to an integer, friends expire
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; within this number of seconds instead of the
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; same as the registration interval
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;rtignoreexpire=yes ; when reading a peer from Realtime, if the peer's registration
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; has expired based on its registration interval, used the stored
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; address information regardless
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[authentication]
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; Global credentials for outbound calls, i.e. when a proxy challenges your
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; Asterisk server for authentication. These credentials override
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; any credentials in peer/register definition if realm is matched.
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;
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; This way, Asterisk can authenticate for outbound calls to other
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; realms. We match realm on the proxy challenge and pick an set of
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; credentials from this list
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; Syntax:
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; auth = <user>:<secret>@<realm>
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; auth = <user>#<md5secret>@<realm>
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; Example:
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;auth=mark:topsecret@digium.com
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;
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; You may also add auth= statements to [peer] definitions
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; Peer auth= override all other authentication settings if we match on realm
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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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; 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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; accountcode accountcode
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; setvar setvar
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; callerid callerid
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; amaflags amaflags
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; incominglimit incominglimit
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; restrictcid 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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; We match on IP address of the proxy for incoming calls
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; since we can not match on username (caller id)
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;type=peer
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;context=from-fwd
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;host=fwd.pulver.com
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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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;fromdomain=provider.sip.domain
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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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;------------------------------------------------------------------------------
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; Definitions of locally connected SIP phones
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;
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; type = user a device that calls us
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; type = peer a device we place calls to
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; type = friend two configurations (peer+user) in one
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;
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; For local phones, type=friend works most of the time
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;
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; If you have one-way audio, you propably have NAT problems.
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; If Asterisk is on a public IP, and the phone is inside of a NAT device
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; you will need to configure nat option for those phones.
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; Also, turn on qualify=yes to keep the nat session open
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;[grandstream1]
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;type=friend
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;context=from-sip ; Where to start in the dialplan when this phone calls
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;callerid=John Doe <1234> ; Full caller ID, to override the phones config
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;host=192.168.0.23 ; we have a static but private IP address
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; No registration allowed
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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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;callerid="Jane Smith" <5678>
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;host=dynamic ; This device needs to register
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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@context,2345 ; Mailbox(-es) for message waiting indicator
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;vmexten=voicemail ; dialplan extension to reach mailbox
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; sets the Message-Account in the MWI notify message
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; defaults to global vmexten which defaults to "asterisk"
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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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;[polycom]
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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=blahpoly
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;host=dynamic ; This peer register with us
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;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
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;username=polly ; Username to use in INVITE until peer registers
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;disallow=all
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;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
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;progressinband=no ; Polycom phones don't work properly with "never"
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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=port ; Allow matching of peer by IP address without matching port number
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;insecure=invite ; Do not require authentication of incoming INVITEs
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;insecure=port,invite ; (both)
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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 the 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 ; IP address to use until registration
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;username=goran ; Username to use when calling this device before registration
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