mirror of
https://github.com/asterisk/asterisk.git
synced 2025-10-25 14:06:27 +00:00
2c65582b66
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7221 65c4cc65-6c06-0410-ace0-fbb531ad65f3
435 lines
19 KiB
Plaintext
435 lines
19 KiB
Plaintext
;
|
|
; SIP Configuration example 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
|
|
;allowguest=no ; Allow or reject guest calls (default is yes, this can also be set to 'osp'
|
|
; if asterisk was compiled with OSP support.
|
|
;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
|
|
bindport=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
|
|
|
|
;domain=mydomain.tld ; Set default domain for this host
|
|
; If configured, Asterisk will only allow
|
|
; INVITE and REFER to non-local domains
|
|
; Use "sip show domains" to list local domains
|
|
;domain=mydomain.tld,mydomain-incoming
|
|
; Add domain and configure incoming context
|
|
; for external calls to this domain
|
|
;domain=1.2.3.4 ; Add IP address as local domain
|
|
; You can have several "domain" settings
|
|
;allowexternalinvites=no ; Disable INVITE and REFER to non-local domains
|
|
; Default is yes
|
|
;autodomain=yes ; Turn this on to have Asterisk add local host
|
|
; name and local IP to domain list.
|
|
;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
|
|
;maxexpiry=3600 ; Max length of incoming registration we allow
|
|
;defaultexpiry=120 ; Default length of incoming/outoing registration
|
|
;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
|
|
;checkmwi=10 ; Default time between mailbox checks for peers
|
|
;vmexten=voicemail ; dialplan extension to reach mailbox sets the
|
|
; Message-Account in the MWI notify message
|
|
; defaults to "asterisk"
|
|
;videosupport=yes ; Turn on support for SIP video
|
|
;recordhistory=yes ; Record SIP history by default
|
|
; (see sip history / sip no history)
|
|
|
|
;disallow=all ; First disallow all codecs
|
|
;allow=ulaw ; Allow codecs in order of preference
|
|
;allow=ilbc ;
|
|
;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
|
|
;sendrpid = yes ; If Remote-Party-ID should be sent
|
|
;progressinband=never ; If we should generate in-band ringing always
|
|
; use 'never' to never use in-band signalling, even in cases
|
|
; where some buggy devices might not render it
|
|
;useragent=Asterisk PBX ; Allows you to change the user agent string
|
|
;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.
|
|
;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
|
|
; a valid phone number
|
|
;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
|
|
; Other options:
|
|
; info : SIP INFO messages
|
|
; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
|
|
; auto : Use rfc2833 if offered, inband otherwise
|
|
|
|
;compactheaders = yes ; send compact sip headers.
|
|
;sipdebug = yes ; Turn on SIP debugging by default, from
|
|
; the moment the channel loads this configuration
|
|
;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
|
|
; Useful to limit subscriptions to local extensions
|
|
; Settable per peer/user also
|
|
;notifyringing = yes ; Notify subscriptions on RINGING state
|
|
|
|
;
|
|
; 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=sipregistrations
|
|
;
|
|
; 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 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
|
|
|
|
;registertimeout=20 ; retry registration calls every 20 seconds (default)
|
|
;registerattempts=10 ; Number of registration attempts before we give up
|
|
; 0 = continue forever, hammering the other server until it
|
|
; accepts the registration
|
|
; Default is 10 tries
|
|
;callevents=no ; generate manager events when sip ua performs events (e.g. hold)
|
|
|
|
;----------------------------------------- NAT SUPPORT ------------------------
|
|
; The externip, externhost and localnet settings are used if you use Asterisk
|
|
; behind a NAT device to communicate with services on the outside.
|
|
|
|
;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
|
|
;externhost=foo.dyndns.net ; Alternatively you can specify an
|
|
; external host, and Asterisk will
|
|
; perform DNS queries periodically. Not
|
|
; recommended for production
|
|
; environments! Use externip instead
|
|
;externrefresh=10 ; How often to refresh externhost if
|
|
; used
|
|
; 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
|
|
|
|
; The nat= setting is used when Asterisk is on a public IP, communicating with
|
|
; devices hidden behind a NAT device (broadband router). If you have one-way
|
|
; audio problems, you usually have problems with your NAT configuration or your
|
|
; firewall's support of SIP+RTP ports. You configure Asterisk choice of RTP
|
|
; ports for incoming audio in rtp.conf
|
|
;
|
|
;nat=no ; Global NAT settings (Affects all peers and users)
|
|
; 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)
|
|
|
|
;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
|
|
; just like friends added from the config file only on a
|
|
; as-needed basis? (yes|no)
|
|
|
|
;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
|
|
; If set to yes, when a SIP UA registers successfully, the ip address,
|
|
; the origination port, the registration period, and the username of
|
|
; the UA will be set to database via realtime. If not present, defaults to 'yes'.
|
|
|
|
;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
|
|
; as if it had just registered? (yes|no|<seconds>)
|
|
; If set to yes, when the registration expires, the friend will vanish from
|
|
; the configuration until requested again. If set to an integer,
|
|
; friends expire within this number of seconds instead of the
|
|
; registration interval.
|
|
|
|
;ignoreregexpire=yes ; Enabling this setting has two functions:
|
|
;
|
|
; For non-realtime peers, when their registration expires, the information
|
|
; will _not_ be removed from memory or the Asterisk database; if you attempt
|
|
; to place a call to the peer, the existing information will be used in spite
|
|
; of it having expired
|
|
;
|
|
; For realtime peers, when the peer is retrieved from realtime storage,
|
|
; the registration information will be used regardless of whether
|
|
; it has expired or not; if it expires while the realtime peer is still in
|
|
; memory (due to caching or other reasons), the information will not be
|
|
; removed from realtime storage
|
|
|
|
; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
|
|
; domains, each of which can direct the call to a specific context if desired.
|
|
; By default, all domains are accepted and sent to the default context or the
|
|
; context associated with the user/peer placing the call.
|
|
; Domains can be specified using:
|
|
; domain=<domain>[,<context>]
|
|
; Examples:
|
|
; domain=myasterisk.dom
|
|
; domain=customer.com,customer-context
|
|
;
|
|
; In addition, all the 'default' domains associated with a server should be
|
|
; added if incoming request filtering is desired.
|
|
; autodomain=yes
|
|
;
|
|
; To disallow requests for domains not serviced by this server:
|
|
; allowexternaldomains=no
|
|
|
|
[authentication]
|
|
; Global credentials for outbound calls, i.e. when a proxy challenges your
|
|
; Asterisk server for authentication. These credentials override
|
|
; any credentials in peer/register definition if realm is matched.
|
|
;
|
|
; This way, Asterisk can authenticate for outbound calls to other
|
|
; realms. We match realm on the proxy challenge and pick an set of
|
|
; credentials from this list
|
|
; Syntax:
|
|
; auth = <user>:<secret>@<realm>
|
|
; auth = <user>#<md5secret>@<realm>
|
|
; Example:
|
|
;auth=mark:topsecret@digium.com
|
|
;
|
|
; You may also add auth= statements to [peer] definitions
|
|
; Peer auth= override all other authentication settings if we match on realm
|
|
|
|
;------------------------------------------------------------------------------
|
|
; 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
|
|
; useclientcode useclientcode
|
|
; accountcode accountcode
|
|
; setvar setvar
|
|
; callerid callerid
|
|
; amaflags amaflags
|
|
; call-limit call-limit
|
|
; restrictcid restrictcid
|
|
; subscribecontext subscribecontext
|
|
; videosupport videosupport
|
|
; mailbox
|
|
; username
|
|
; template
|
|
; fromdomain
|
|
; regexten
|
|
; fromuser
|
|
; host
|
|
; port
|
|
; qualify
|
|
; defaultip
|
|
; rtptimeout
|
|
; rtpholdtimeout
|
|
; sendrpid
|
|
|
|
;[sip_proxy]
|
|
; For incoming calls only. Example: FWD (Free World Dialup)
|
|
; We match on IP address of the proxy for incoming calls
|
|
; since we can not match on username (caller id)
|
|
;type=peer
|
|
;context=from-fwd
|
|
;host=fwd.pulver.com
|
|
|
|
;[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!
|
|
;fromdomain=provider.sip.domain
|
|
;host=box.provider.com
|
|
;usereqphone=yes ; This provider requires ";user=phone" on URI
|
|
;call-limit=5 ; permit only 5 simultaneous outgoing calls to this peer
|
|
|
|
;------------------------------------------------------------------------------
|
|
; Definitions of locally connected SIP phones
|
|
;
|
|
; type = user a device that authenticates to us by "from" field to place calls
|
|
; type = peer a device we place calls to or that calls us and we match by host
|
|
; type = friend two configurations (peer+user) in one
|
|
;
|
|
; For local phones, type=friend works most of the time
|
|
;
|
|
; If you have one-way audio, you propably have NAT problems.
|
|
; If Asterisk is on a public IP, and the phone is inside of a NAT device
|
|
; you will need to configure nat option for those phones.
|
|
; Also, turn on qualify=yes to keep the nat session open
|
|
|
|
;[grandstream1]
|
|
;type=friend
|
|
;context=from-sip ; Where to start in the dialplan when this phone calls
|
|
;callerid=John Doe <1234> ; Full caller ID, to override the phones config
|
|
;host=192.168.0.23 ; we have a static but private IP address
|
|
; No registration allowed
|
|
;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
|
|
;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time
|
|
; from the phone to asterisk
|
|
; (1 for the explicit peer, 1 for the explicit user,
|
|
; remember that a friend equals 1 peer and 1 user in
|
|
; memory)
|
|
;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
|
|
;astdb=chan2ext/SIP/grandstream1=1234 ; ensures an astDB entry exists
|
|
|
|
|
|
;[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
|
|
;callerid="Jane Smith" <5678>
|
|
;host=dynamic ; This device needs to register
|
|
;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
|
|
;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes
|
|
|
|
|
|
;[snom]
|
|
;type=friend ; Friends place calls and receive calls
|
|
;context=from-sip ; Context for incoming calls from this user
|
|
;secret=blah
|
|
;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions
|
|
;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
|
|
;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator
|
|
;vmexten=voicemail ; dialplan extension to reach mailbox
|
|
; sets the Message-Account in the MWI notify message
|
|
; defaults to global vmexten which defaults to "asterisk"
|
|
;restrictcid=yes ; To have the callerid restriced -> sent as ANI
|
|
;disallow=all
|
|
;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
|
|
|
|
|
|
;[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
|
|
; Normally you do NOT need to set this parameter
|
|
;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
|
|
;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
|
|
;secret=blah
|
|
;qualify=200 ; Qualify peer is no more than 200ms away
|
|
;nat=yes ; This phone may be natted
|
|
; Send SIP and RTP to the 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 ; IP address to use until registration
|
|
;username=goran ; Username to use when calling this device before registration
|
|
; Normally you do NOT need to set this parameter
|
|
;setvar=CUSTID=5678 ; Channel variable to be set for all calls from this device
|
|
|