sip.conf [general] context=default ; Default context for incoming calls ;allowguest=no ; Allow or reject guest calls (default is yes) allowoverlap=no ; Disable overlap dialing support. (Default is yes) ;allowtransfer=no ; Disable all transfers (unless enabled in peers or users) ; Default is enabled realm=somehost.dyndns.org ; Realm for digest authentication ; defaults to "asterisk". If you set a system name in ; asterisk.conf, it defaults to that system name ; 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) ; bindport is the local UDP port that Asterisk will listen on bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ;----------------------------------------- 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 = 11.11.11.11 ; 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=192.168.10.0/255.255.255.0; 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=yes ; Global NAT settings (Affects all peers and users) ; yes = Always ignore info and assume NAT ; no = Use NAT mode only according to RFC3581 (;rport) ; never = Never attempt NAT mode or RFC3581 support ; route = Assume NAT, don't send rport ; (work around more UNIDEN bugs) [4202] type=friend host=dynamic username=4202 secret=somesecret nat=yes context=local-sip canreinvite=no