---sip.conf--- ; ; SIP configuration ; ; ; 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 = xx.xx.xx.xx ; Address to bind SIP channel to context = default ; Default context for incoming calls srvlookup = yes ; Enable DNS SRV lookups on outbound calls ;allowguest = yes pedantic = no ; 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=alaw ;allow=gsm ;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 => 1747xxxxxxx:*******@proxy01.sipphone.com ;gizmo config ; ; This section configures Asterisk to receive incoming calls from the Gizmo VoIP network ; if you have multiple Gizmo accounts, just create one entry for each account. [proxy01.sipphone.com] host=proxy01.sipphone.com username=1747nnnnnnn secret=******* type=friend context=gizmocontext disallow=all allow=ulaw allow=alaw dtmfmode=inband insecure=very canreinvite=no [voxbone] [213.246.216.81] host = 213.246.216.81 type = friend insecure = very context = voxbonecontext [213.246.216.82] host = 213.246.216.82 type = friend insecure = very context = voxbonecontext [213.246.216.90] host = 213.246.216.90 type = friend insecure = very context = voxbonecontext [213.246.216.91] host = 213.246.216.91 type = friend insecure = very context = voxbonecontext [213.246.216.92] host = 213.246.216.92 type = friend insecure = very context = voxbonecontext