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- ;
- ; 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
- ;
- ; reload chan_sip.so Reload configuration file
- ; Active SIP peers will not be reconfigured
- ;
- [general]
- context=default ; Default context for incoming calls
- ;recordhistory=yes ; Record SIP history by default
- ; (see sip history / sip no history)
- ;realm=mydomain.tld ; Realm for digest authentication
- ; defaults to "asterisk"
- ; Realms MUST be globally unique according to RFC 3261
- ; Set this to your host name or domain name
- port=5060 ; UDP Port to bind to (SIP standard port is 5060)
- bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
- srvlookup=yes ; Enable DNS SRV lookups on outbound calls
- ; Note: Asterisk only uses the first host
- ; in SRV records
- ; Disabling DNS SRV lookups disables the
- ; ability to place SIP calls based on domain
- ; names to some other SIP users on the Internet
-
- ;pedantic=yes ; Enable slow, pedantic checking for Pingtel
- ; and multiline formatted headers for strict
- ; SIP compatibility (defaults to "no")
- ;tos=184 ; Set IP QoS to either a keyword or numeric val
- ;tos=lowdelay ; lowdelay,throughput,reliability,mincost,none
- ;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 MWI NOTIFY
- ;videosupport=yes ; Turn on support for SIP video
- ;disallow=all ; First disallow all codecs
- ;allow=ulaw ; Allow codecs in order of preference
- ;allow=ilbc ; Note: codec order is respected only in [general]
- ;musicclass=default ; Sets the default music on hold class for all SIP calls
- ; This may also be set for individual users/peers
- ;language=en ; Default language setting for all users/peers
- ; This may also be set for individual users/peers
- ;relaxdtmf=yes ; Relax dtmf handling
- ;rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity
- ; when we're not on hold
- ;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP activity
- ; when we're on hold (must be > rtptimeout)
- ;trustrpid = no ; If Remote-Party-ID should be trusted
- ;progressinband=no ; If we should generate in-band ringing always
- ;useragent=Asterisk PBX ; Allows you to change the user agent string
- ;nat=no ; NAT settings
- ; yes = Always ignore info and assume NAT
- ; no = Use NAT mode only according to RFC3581
- ; never = Never attempt NAT mode or RFC3581 support
- ; route = Assume NAT, don't send rport (work around more UNIDEN bugs)
- ;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
- ; ; Note that promiscredir when redirects are made to the
- ; ; local system will cause loops since SIP is incapable
- ; ; of performing a "hairpin" call.
- ;
- ; If regcontext is specified, Asterisk will dynamically
- ; create and destroy a NoOp priority 1 extension for a given
- ; peer who registers or unregisters with us. The actual extension
- ; is the 'regexten' parameter of the registering peer or its
- ; name if 'regexten' is not provided. More than one regexten may be supplied
- ; if they are separated by '&'. Patterns may be used in regexten.
- ;
- ;regcontext=iaxregistrations
- ;
- ; Asterisk can register as a SIP user agent to a SIP proxy (provider)
- ; Format for the register statement is:
- ; register => user[:secret[:authuser]]@host[:port][/extension]
- ;
- ; If no extension is given, the 's' extension is used. The extension
- ; needs to be defined in extensions.conf to be able to accept calls
- ; from this SIP proxy (provider)
- ;
- ; host is either a host name defined in DNS or the name of a
- ; section defined below.
- ;
- ; Examples:
- ;
- ;register => 1234:password@mysipprovider.com
- ;
- ; This will pass incoming calls to the 's' extension
- ;
- ;
- ;register => 2345:password@sip_proxy/1234
- ;
- ; Register 2345 at sip provider 'sip_proxy'. Calls from this provider connect to local
- ; extension 1234 in extensions.conf default context, unless you define
- ; unless you configure a [sip_proxy] section below, and configure a context.
- ; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
- ; Tip 2: Use separate type=peer and type=user sections for SIP providers
- ; (instead of type=friend) if you have calls in both directions
-
- ;externip = 200.201.202.203 ; Address that we're going to put in outbound SIP messages
- ; if we're behind a NAT
- ; The externip and localnet is used
- ; when registering and communicating with other proxies
- ; that we're registered with
- ; You may add multiple local networks. A reasonable set of defaults
- ; are:
- ;localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
- ;localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
- ;localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
- ;localnet=169.254.0.0/255.255.0.0 ;Zero conf local network
- ;-----------------------------------------------------------------------------------
- ; Users and peers have different settings available. Friends have all settings,
- ; since a friend is both a peer and a user
- ;
- ; User config options: Peer configuration:
- ; -------------------- -------------------
- ; context context
- ; permit permit
- ; deny deny
- ; secret secret
- ; md5secret md5secret
- ; dtmfmode dtmfmode
- ; canreinvite canreinvite
- ; nat nat
- ; callgroup callgroup
- ; pickupgroup pickupgroup
- ; language language
- ; allow allow
- ; disallow disallow
- ; insecure insecure
- ; trustrpid trustrpid
- ; progressinband progressinband
- ; promiscredir promiscredir
- ; callerid
- ; accountcode
- ; amaflags
- ; incominglimit
- ; restrictcid
- ; mailbox
- ; username
- ; template
- ; fromdomain
- ; regexten
- ; fromuser
- ; host
- ; mask
- ; port
- ; qualify
- ; defaultip
- ; rtptimeout
- ; rtpholdtimeout
- ;[sip_proxy]
- ; For incoming calls only. Example: FWD (Free World Dialup)
- ;type=user
- ;context=from-fwd
- ;[sip_proxy-out]
- ;type=peer ; we only want to call out, not be called
- ;secret=guessit
- ;username=yourusername ; Authentication user for outbound proxies
- ;fromuser=yourusername ; Many SIP providers require this!
- ;host=box.provider.com
- ;[grandstream1]
- ;type=friend ; either "friend" (peer+user), "peer" or "user"
- ;context=from-sip
- ;fromuser=grandstream1 ; overrides the callerid, e.g. required by FWD
- ;callerid=John Doe <1234>
- ;host=192.168.0.23 ; we have a static but private IP address
- ;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
- ; from the phone to asterisk
- ;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
- ; listed with allow= does NOT matter!
- ;allow=alaw
- ;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
- ;allow=g729 ; Pass-thru only unless g729 license obtained
- ;[xlite1]
- ;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
- ;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
- ;type=friend
- ;regexten=1234 ; When they register, create extension 1234
- ;username=xlite1
- ;callerid="Jane Smith" <5678>
- ;host=dynamic
- ;nat=yes ; X-Lite is behind a NAT router
- ;canreinvite=no ; Typically set to NO if behind NAT
- ;disallow=all
- ;allow=gsm ; GSM consumes far less bandwidth than ulaw
- ;allow=ulaw
- ;allow=alaw
- ;[snom]
- ;type=friend ; Friends place calls and receive calls
- ;context=from-sip ; Context for incoming calls from this user
- ;secret=blah
- ;language=de ; Use German prompts for this user
- ;host=dynamic ; This peer register with us
- ;dtmfmode=inband ; Choices are inband, rfc2833, or info
- ;defaultip=192.168.0.59 ; IP used until peer registers
- ;username=snom ; Username to use in INVITE until peer registers
- ;mailbox=1234,2345 ; Mailboxes for message waiting indicator
- ;restrictcid=yes ; To have the callerid restriced -> sent as ANI
- ;disallow=all
- ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
- ;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator
- ;[polycom]
- ;type=friend ; Friends place calls and receive calls
- ;context=from-sip ; Context for incoming calls from this user
- ;secret=blahpoly
- ;host=dynamic ; This peer register with us
- ;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
- ;username=polly ; Username to use in INVITE until peer registers
- ;disallow=all
- ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
- ;progressinband=no ; Polycom phones don't work properly with "never"
- ;[pingtel]
- ;type=friend
- ;username=pingtel
- ;secret=blah
- ;host=dynamic
- ;insecure=port ; Allow matching of peer by IP address without matching port number
- ;insecure=invite ; Do not require authentication of incoming INVITEs
- ;insecure=port,invite ; (both)
- ;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 ; We are in caller groups 1,3,4
- ;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5
- ;defaultip=192.168.0.60 ; IP address to use if peer has not registred
- ;[cisco1]
- ;type=friend
- ;username=cisco1
- ;secret=blah
- ;qualify=200 ; Qualify peer is no more than 200ms away
- ;nat=yes ; This phone may be natted
- ; Send SIP and RTP to IP address that packet is
- ; received from instead of trusting SIP headers
- ;host=dynamic ; This device registers with us
- ;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
- ; behind a NAT).
- ;defaultip=192.168.0.4
- ;[cisco2]
- ;type=friend
- ;username=cisco2
- ;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 to ease billing
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