;! Generator: PBX Manager
;! Creation Date: Thu Jul 19 15:25:41 2007
;!
;
; 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 = demo ; Default context for incoming calls
; Allow or reject guest calls (default is yes, this can also be set to 'osp'
; if asterisk was compiled with OSP support.
; 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 = 168.10.1.15 ; 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
; 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
; Add IP address as local domain
; You can have several "domain" settings
; Disable INVITE and REFER to non-local domains
; Default is yes
; Turn this on to have Asterisk add local host
; name and local IP to domain list.
; Enable slow, pedantic checking for Pingtel
; and multiline formatted headers for strict
; SIP compatibility (defaults to "no")
; Set IP QoS to either a keyword or numeric val
; lowdelay,throughput,reliability,mincost,none
; Max length of incoming registration we allow
; Default length of incoming/outgoing registration
; Allow overriding of mime type in MWI NOTIFY
; Default time between mailbox checks for peers
; dialplan extension to reach mailbox sets the
; Message-Account in the MWI notify message
; defaults to "asterisk"
; Turn on support for SIP video
; Record SIP history by default
; (see sip history / sip no history)
; First disallow all codecs
; Allow codecs in order of preference
;
; Sets the default music on hold class for all SIP calls
; This may also be set for individual users/peers
; Default language setting for all users/peers
; This may also be set for individual users/peers
; Relax dtmf handling
; Terminate call if 60 seconds of no RTP activity
; when we're not on hold
; Terminate call if 300 seconds of no RTP activity
; when we're on hold (must be > rtptimeout)
; If Remote-Party-ID should be trusted
; If Remote-Party-ID should be sent
; 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
; Valid values: yes, no, never Default: never
; Allows you to change the user agent string
; 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.
;user=phone" is added to uri that contains
; a valid phone number
; 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
; send compact sip headers.
; Turn on SIP debugging by default, from
; the moment the channel loads this configuration
; Set a specific context for SUBSCRIBE requests
; Useful to limit subscriptions to local extensions
; Settable per peer/user also
; Notify subscriptions on RINGING state
; When an incoming INVITE or REGISTER is to be rejected,
; for any reason, always reject with '401 Unauthorized'
; instead of letting the requester know whether there was
; a matching user or peer for their request
;
; 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
; retry registration calls every 20 seconds (default)
; Number of registration attempts before we give up
; 0 = continue forever, hammering the other server until it
; accepts the registration
; Default is 0 tries, continue forever
; 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.
; 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
; Alternatively you can specify an
; external host, and Asterisk will
; perform DNS queries periodically. Not
; recommended for production
; environments! Use externip instead
; How often to refresh externhost if
; used
; You may add multiple local networks. A reasonable set of defaults
; are:
; All RFC 1918 addresses are local networks
; Also RFC1918
; Another RFC1918 with CIDR notation
;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
;
; 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)
; 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)
; 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'.
; 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.
; Enabling this setting has two functions:
;
; For non-realtime peers, when their registration expires, the information
; 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
; 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
; When making outbound SIP INVITEs to
; non-peers, use your primary domain "identity"
; for From: headers instead of just your IP
; address. This is to be polite and
; it may be a mandatory requirement for some
; destinations which do not have a prior
; account relationship with your server.
[authentication]
[3000]
type = friend
context = default
username = 3000
host = dynamic
mailbox = 3000
dtmfmode = rfc2833
[3001]
type = friend
context = default
username = 3001
host = dynamic
mailbox = 3001
dtmfmode = rfc2833
[3002]
type = friend
username = 3002
context = default
host = dynamic
mailbox = 3002
dtmfmode = rfc2833
[3003]
type = friend
context = default
username = 3003
host = dynamic
mailbox = 3003
dtmfmode = rfc2833
[3004]
type = friend
context = default
username = 3004
host = dynamic
mailbox = 3004
dtmfmode = rfc2833
[3005]
type = friend
username = 3005
context = default
host = dynamic
mailbox = 3005
dtmfmode = rfc2833
[3006]
type = friend
context = default
username = 3006
host = dynamic
mailbox = 3006
dtmfmode = rfc2833
[3007]
type = friend
context = default
username = 3007
host = dynamic
mailbox = 3007
dtmfmode = rfc2833
[3008]
type = friend
username = 3008
context = default
host = dynamic
mailbox = 3008
dtmfmode = rfc2833
; 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
; mailbox
; username
; template
; fromdomain
; regexten
; fromuser
; host
; port
; qualify
; defaultip
; rtptimeout
; rtpholdtimeout
; sendrpid
; outboundproxy
;[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]
; we only want to call out, not be called
;secret=guessit
; Authentication user for outbound proxies
; Many SIP providers require this!
;fromdomain=provider.sip.domain
;host=box.provider.com
;user=phone" on URI
; permit only 5 simultaneous outgoing calls to this peer
; send outbound signaling to this proxy, not directly to the 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 probably 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
; Where to start in the dialplan when this phone calls
; Full caller ID, to override the phones config
; we have a static but private IP address
; No registration allowed
; there is not NAT between phone and Asterisk
; allow RTP voice traffic to bypass Asterisk
; either RFC2833 or INFO for the BudgeTone
; 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 in voicemail context "default"
; need to disallow=all before we can use allow=
; Note: In user sections the order of codecs
; listed with allow= does NOT matter!
;allow=alaw
; Asterisk only supports g723.1 pass-thru!
; 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
; When they register, create extension 1234
;callerid="Jane Smith" <5678>
; This device needs to register
; X-Lite is behind a NAT router
; Typically set to NO if behind NAT
;disallow=all
; GSM consumes far less bandwidth than ulaw
;allow=ulaw
;allow=alaw
; Subscribe to status of multiple mailboxes
;[snom]
; Friends place calls and receive calls
; Context for incoming calls from this user
;secret=blah
; Only allow SUBSCRIBE for local extensions
; Use German prompts for this user
; This peer register with us
; Choices are inband, rfc2833, or info
; IP used until peer registers
; Mailbox(-es) for message waiting indicator
; dialplan extension to reach mailbox
; sets the Message-Account in the MWI notify message
; defaults to global vmexten which defaults to "asterisk"
; To have the callerid restriced -> sent as ANI
;disallow=all
; dtmfmode=inband only works with ulaw or alaw!
;[polycom]
; Friends place calls and receive calls
; Context for incoming calls from this user
;secret=blahpoly
; This peer register with us
; Choices are inband, rfc2833, or info
; Username to use in INVITE until peer registers
; Normally you do NOT need to set this parameter
;disallow=all
; dtmfmode=inband only works with ulaw or alaw!
; Polycom phones don't work properly with "never"
;[pingtel]
;type=friend
;secret=blah
;host=dynamic
; Allow matching of peer by IP address without matching port number
; Do not require authentication of incoming INVITEs
; (both)
; Consider it down if it's 1 second to reply
; Helps with NAT session
; qualify=yes uses default value
;
; Call group and Pickup group should be in the range from 0 to 63
;
; We are in caller groups 1,3,4
; We can do call pick-p for call group 1,3,4,5
; IP address to use if peer has not registered
;[cisco1]
;type=friend
;secret=blah
; Qualify peer is no more than 200ms away
; This phone may be natted
; Send SIP and RTP to the IP address that packet is
; received from instead of trusting SIP headers
; This device registers with us
; 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).
; IP address to use until registration
; Username to use when calling this device before registration
; Normally you do NOT need to set this parameter
; Channel variable to be set for all calls from this device