[general] context=from-sip ; 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=5067 ; 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 ;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=250 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY ;videosupport=yes ; Turn on support for SIP video disallow=all ; First disallow all codecs allow=gsm ; Note: codec order is respected only in [general] allow=ilbc ; Note: codec order is respected only in [general] allow=ulaw ; Allow codecs in order of preference ;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=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 promiscredir = yes ; If yes, allows 302 or REDIR to non-local SIP address ; 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 = 213.10.134.39 ; 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=192.168.2.6/255.255.255.0; All RFC 1918 addresses are local networks register => 9991651:XXXXXX@sipgate.co.uk:5060/9991651 ;CHECKNR [checker] allow=gsm context=checker type=peer secret=XXXXXX username=9991651 host=sipgate.co.uk fromuser=9991651 fromdomain=sipgate.co.uk insecure=very