--- sip-cvs.conf.sample Sat Apr 10 18:26:12 2004 +++ sip.conf.sample Sat Apr 10 19:28:02 2004 @@ -21,24 +21,35 @@ ; [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 - ; Asterisk only uses the first host in SRV records -;pedantic = yes ; Enable slow, pedantic checking for Pingtel -;tos=lowdelay ; IP QoS parameter, either keyword or value - ; like tos=184 -;maxexpirey=3600 ; Max length of incoming registration we allow -;defaultexpirey=120 ; Default length of incoming/outoing registration +context=default ; Default context for incoming calls +port=5060 ; Port to bind to +bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) + + ; The externip, localnet and localmask is used + ; when registering and communicating with other + ; proxies that we're registered with +;externip=200.201.202.203 ; Address for SIP messages if we are behind NAT +;localnet=192.168.1.0 ; Internal NETWORK address if we are behind NAT +;localmask=255.255.255.0 ; Internal netmask if we are behind NAT + +;srvlookup=yes ; Enable DNS SRV lookups on outbound calls + ; Asterisk only uses the first host in SRV records +;pedantic=yes ; Enable slow, pedantic checking for Pingtel +;tos=184 ; Set IP QoS to either a keyword or numeric val +;tos=lowdelay ; lowdelay,throughput,reliability,mincost,none +;maxexpirey=3600 ; Max validity of incoming registration we allow +;defaultexpirey=120 ; Default validity 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 +;disallow=all ; First disallow all codecs +;allow=ulaw ; Now allow codecs in order of preference +;allow=ilbc ; Note: codec order is respected only in [general] ; Asterisk can register as a SIP user agent to a SIP proxy (provider) +; The ONLY purpose of registration is to inform the remote server of +; our current IP address. +; ; Format for the register statement is: ; register => user[:secret[:authuser]]@host[:port][/extension] ; @@ -51,33 +62,81 @@ ; ; Examples: ; -;register => 1234:password@mysipprovider.com +;register => 2345:password@mysipprovider.com +; +; This will pass incoming calls to the 's' extension ; -; Will call to the 's' extension ; +;register => 2345:password@sip/1234 ; -;register => 2345@mysipprovider.com/1234 +; Passes call to extension 1234 in context "default" ; -; 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 - +; +;register => 2345@other_sipprovider/1234 +; +; Register 2345 at sip provider. Calls coming from this provider +; connect to local extension 1234 in extensions.conf default context, +; unless you configure a context in the section [other_sipprovider] +; +; 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 + +[other_sipprovider] +; For incoming calls only. Example: FWD (Free World Dialup) +type=user +context=from-fwd +host=fwd.pulver.com + +[sipprovider-out] +;type=peer ; we only want to call out, not be called +;secret=guessit +;username=yourusername +;fromuser=yourusername ; Many SIP providers require this! +;host=box.provider.com + +;[grandstream1] +;type=friend ; either "friend" (peer+user), "peer" or "user" +;context=from-sip +;username=grandstream1 ; usually matches the [section] title +;fromuser=grandstream1 ; overrides the callerid, e.g. required by FWD +;callerid=John Doe <1234> +;host=192.168.0.23 ; we have a static but private IP address +;nat=no ; there is not NAT between phone and Asterisk +;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk +;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone +;outgoinglimit=1 ; disable callwaiting signal (2nd call to phone) +;incominglimit=1 ; permit only 1 outgoing call at a time +;mailbox=1234@default ; mailbox 1234 in voicemail context "default" +;disallow=all ; need to disallow=all before we can use allow= +;allow=ulaw ; Note: In user sections the order of codecs + ; listed with allow= does NOT matter! +;allow=alaw +;allow=g723.1 ; Asterisk only supports g723.1 pass-thru! +;allow=g729 ; Pass-thru only unless g729 license obtained -;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 ; Internal NETWORK address -;localmask = 255.255.255.0 ; Internal netmask - ; The externip, localnet and localmask is used - ; when registering and communication with other proxies - ; that we're registered with +;[xlite1] +;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)! +;type=friend +;username=xlite1 +;callerid="Jane Smith" <5678> +;host=dynamic +;nat=yes ; X-Lite is behind a NAT router +;canreinvite=no ; Typically set to NO if behind NAT +;disallow=all +;allow=gsm ; GSM consumes far less bandwidth than ulaw +;allow=ulaw +;allow=alaw -;[snomsip] +;[snom] +;context=from-sip ;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 +;disallow=all +;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw! +;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator ;restrictcid=yes ; To have the callerid restriced -> sent as ANI ;[pingtel] @@ -85,39 +144,37 @@ ;username=pingtel ;secret=blah ;host=dynamic +;defaultip=192.168.0.60 ; try to find us here if phone hasn't registered ;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 + ; Can help with NAT session to keep FW open + ; qualify=yes uses default value +;callgroup=1,3-5 ; we are in callgroup 1,3,4,5 +;pickupgroup=1,3-4 ; we can do call pick-up for callgroup 1,3,4 -;[cisco] +;[cisco1] ;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 +;username=cisco1 +;secret=blahblubb +;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 +;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 ; Allow peer to be no more than 200ms away -;[cisco1] +;[cisco2] ;type=friend -;username=cisco1 +;username=cisco2 ;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 +;secret=tiptop ;host=dynamic ;defaultip=192.168.0.4 ;amaflags=default ; Choices are default, omit, billing, documentation