; ; 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] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind SIP channel to context = default ; Default context for incoming calls ;srvlookup = yes ; Enable DNS SRV lookups on outbound calls ;pedantic = yes ; Enable slow, pedantic checking for Pingtel ;tos=lowdelay ; IP QoS parameter, either keyword or value ;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 NOTIFY ;videosupport=yes ; Turn on support for SIP video ;disallow=all ; Disallow all codecs ;allow=ulaw ; Allow codecs in order of preference ;allow=ilbc ; 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 ; Will call to the 's' extension ; ;register => 2345@mysipprovider.com/1234 ; ; Register 2345 at sip provider. Calls from this provider connect to local ; extension 1234 in extensions.conf default context, unless you define ; [mysipprovider.com] in a section below, and configure a context ;externip = 200.201.202.203 ; Address that we're going to put in outbound SIP messages ; if we're behind a NAT ;localnet = 192.168.1.0 ; Internet NETWORK address ;localmask = 255.255.255.0 ; Internet netmask ; The externip, localnet and localmask is used ; when registering and communication with other proxies ; that we're registered with ;[snomsip] ;type=friend ;secret=blah ;host=dynamic ;dtmfmode=inband ; Choices are inband, rfc2833, or info ;defaultip=192.168.0.59 ;mailbox=1234,2345 ; Mailbox for message waiting indicator ;restrictcid=yes ; To have the callerid restriced -> sent as ANI ;[pingtel] ;type=friend ;username=pingtel ;secret=blah ;host=dynamic ;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 ;pickupgroup=1,3-4 ;defaultip=192.168.0.60 ;[cisco] ;type=friend ;username=cisco ;secret=blah ;nat=yes ; This phone may be natted ; Use IP address that packet is received from ; instead of trusting SIP headers ;host=dynamic ;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). ;qualify=200 ; Qualify peer is no more than 200ms away ;defaultip=192.168.0.4 ;[cisco1] ;type=friend ;username=cisco1 ;fromuser=markster ; Specify user to put in "from" instead of callerid ;fromdomain=yourdomain.com ; Specify domain to put in "from" instead of callerid ; fromuser and fromdomain are used when Asterisk ; places calls to this account. It is not used for ; calls from this account. ;secret=blah ;host=dynamic ;defaultip=192.168.0.4 ;amaflags=default ; Choices are default, omit, billing, documentation ;accountcode=markster ; Users may be associated with an accountcode to ease billing