;sip.conf ; ; 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 ; [general] pedantic=no context=inbound ; 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 ; 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=gsm 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=yes ; If we should generate in-band ringing always ;useragent=Asterisk PBX ; Allows you to change the user agent string ;nat=yes ; 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: ; ; ; ; This will pass incoming calls to the 's' extension ; ; ;register => 2345:password@sip_proxy/1234 ; ;register => :@switch-1.asterlink.com/ ; ; 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=192.168.0.0/255.255.255.0; All RFC 1918 addresses are local networks ;localnet=192.168.1.0/255.255.255.0 ;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 ;{providers} ;[sip.broadvoice.com] ;permit=147.135.0.128/30 ;type=friend ;host=sip.broadvoice.com ;fromdomain=sip.broadvoice.com ;fromuser= ;secret= ;username= ;insecure=very ;context=inbound ;authuser= ;authname= ;dtmfmode=inband ;dtmf=inband ;qualify=yes ;inscure=very ;maxexpirey=15 ;nat=yes ;canreinvite=no ;disallow=all ;allow=ulaw ;allow=gsm ;allow=ilbc ;allow=g726 ;allow=g729 ;[sipphone] ;type=friend ;secret= ;username=1747 ;fromuser=1747 ;fromdomain=proxy01.sipphone.com ;host=proxy01.sipphone.com ;context=default ;disallow=all ;allow=ulaw ;[asterlink] ;type=peer ;host=switch.asterlink.com ;username= ;fromuser= ;secret= ;disallow=all ; We allow pretty much any codec you can throw at us but g726 is good ;allow=g726 ;[asterlink-switch-01] ;type=peer ;host=switch-01.asterlink.com ;context=default ;insecure=very ;[asterlink-switch-02] ;type=peer ;host=switch-02.asterlink.com ;context=default ;insecure=very ;[asterlink-switch-03] ;type=peer ;host=switch-03.asterlink.com ;context=default ;insecure=very ;[asterlink-switch-04] ;type=peer ;host=switch-04.asterlink.com ;context=default ;insecure=very ;[asterlink-switch-05] ;type=peer ;host=switch-05.asterlink.com ;context=default ;insecure=very ;[asterlink-switch-06] ;type=peer ;host=switch-06.asterlink.com ;context=default ;insecure=very ;[asterlink-switch-07] ;type=peer ;host=switch-07.asterlink.com ;context=default ;insecure=very ;[extenions] ;[1001] ;type=friend ; Friends place calls and receive calls ;context=admin ; Context for incoming calls from this user ;secret=1234 ;host=dynamic ; This peer register with us ;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info ;dtmf=rfc2833 ;username=1001 ; Username to use in INVITE until peer registers ;mailbox=1001 ;nat=yes ; X-Lite is behind a NAT router ;canreinvite=no ; Typically set to NO if behind NAT ;callerid= Name ;disallow=all ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw! ;allow=gsm ;allow=ilbc