; Pre-configured analog extensions, depends on IP0X model and what modules
; are installed. Some of these may map to FXO ports
-exten => 6001,n,Dial(Zap/1)
-exten => 6002,n,Dial(Zap/2)
-exten => 6003,n,Dial(Zap/3)
-exten => 6004,n,Dial(Zap/4)
-exten => 6005,n,Dial(Zap/5)
-exten => 6006,n,Dial(Zap/6)
-exten => 6007,n,Dial(Zap/7)
-exten => 6008,n,Dial(Zap/8)
+exten => 6001,1,Dial(Zap/1)
+exten => 6002,1,Dial(Zap/2)
+exten => 6003,1,Dial(Zap/3)
+exten => 6004,1,Dial(Zap/4)
+exten => 6005,1,Dial(Zap/5)
+exten => 6006,1,Dial(Zap/6)
+exten => 6007,1,Dial(Zap/7)
+exten => 6008,1,Dial(Zap/8)
; Pre-configured SIP-phone extensions. Primary use case is multiple SIP
; extensions and FXO analog Ports
-exten => 6011,n,Dial(SIP/6011)
-exten => 6012,n,Dial(SIP/6012)
-exten => 6013,n,Dial(SIP/6013)
-exten => 6014,n,Dial(SIP/6014)
-exten => 6015,n,Dial(SIP/6015)
-exten => 6016,n,Dial(SIP/6016)
-exten => 6017,n,Dial(SIP/6017)
-exten => 6018,n,Dial(SIP/6018)
-exten => 6019,n,Dial(SIP/6019)
-exten => 6020,n,Dial(SIP/6020)
-exten => 6021,n,Dial(SIP/6021)
-exten => 6022,n,Dial(SIP/6022)
-exten => 6023,n,Dial(SIP/6023)
-exten => 6024,n,Dial(SIP/6024)
-exten => 6025,n,Dial(SIP/6025)
-exten => 6026,n,Dial(SIP/6026)
-exten => 6027,n,Dial(SIP/6027)
-exten => 6028,n,Dial(SIP/6028)
-exten => 6029,n,Dial(SIP/6029)
-exten => 6020,n,Dial(SIP/6020)
+exten => 6011,1,Dial(SIP/6011)
+exten => 6012,1,Dial(SIP/6012)
+exten => 6013,1,Dial(SIP/6013)
+exten => 6014,1,Dial(SIP/6014)
+exten => 6015,1,Dial(SIP/6015)
+exten => 6016,1,Dial(SIP/6016)
+exten => 6017,1,Dial(SIP/6017)
+exten => 6018,1,Dial(SIP/6018)
+exten => 6019,1,Dial(SIP/6019)
+exten => 6020,1,Dial(SIP/6020)
+exten => 6021,1,Dial(SIP/6021)
+exten => 6022,1,Dial(SIP/6022)
+exten => 6023,1,Dial(SIP/6023)
+exten => 6024,1,Dial(SIP/6024)
+exten => 6025,1,Dial(SIP/6025)
+exten => 6026,1,Dial(SIP/6026)
+exten => 6027,1,Dial(SIP/6027)
+exten => 6028,1,Dial(SIP/6028)
+exten => 6029,1,Dial(SIP/6029)
+exten => 6030,1,Dial(SIP/6030)
--- /dev/null
+;
+; 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 0 tries, continue forever
+;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
+
+; fromdomain=mydomain.tld ; 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]
+; 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
+
+; Pre-configured SIP extensions
+
+[6011]
+type=friend
+context=default
+host=dynamic
+user=6011
+secret=6011
+canreinvite=no
+callerid=6011
+disallow=all
+allow=ulaw,g729
+qualify=yes
+
+[6012]
+type=friend
+context=default
+host=dynamic
+user=6012
+secret=6012
+canreinvite=no
+callerid=6012
+disallow=all
+allow=ulaw,g729
+
+[6013]
+type=friend
+context=default
+host=dynamic
+user=6013
+secret=6013
+canreinvite=no
+callerid=6013
+disallow=all
+allow=ulaw,g729
+
+[6014]
+type=friend
+context=default
+host=dynamic
+user=6014
+secret=6014
+canreinvite=no
+callerid=6014
+disallow=all
+allow=ulaw,g729
+
+[6015]
+type=friend
+context=default
+host=dynamic
+user=6015
+secret=6015
+canreinvite=no
+callerid=6015
+disallow=all
+allow=ulaw,g729
+
+[6016]
+type=friend
+context=default
+host=dynamic
+user=6016
+secret=6016
+canreinvite=no
+callerid=6016
+disallow=all
+allow=ulaw,g729
+
+[6017]
+type=friend
+context=default
+host=dynamic
+user=6017
+secret=6017
+canreinvite=no
+callerid=6017
+disallow=all
+allow=ulaw,g729
+
+[6018]
+type=friend
+context=default
+host=dynamic
+user=6018
+secret=6018
+canreinvite=no
+callerid=6018
+disallow=all
+allow=ulaw,g729
+
+; Pre-configured SIP trunks
+
+[voip1]
+type=friend
+context=default
+user=user
+secret=password
+host=192.168.1.28
+canreinvite=no
+disallow=all
+allow=ulaw,g729
+qualify=yes
+
projects / freetel-svn-tracking.git / commitdiff
