diff options
author | markster <markster@f38db490-d61c-443f-a65b-d21fe96a405b> | 2004-05-24 15:09:34 +0000 |
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committer | markster <markster@f38db490-d61c-443f-a65b-d21fe96a405b> | 2004-05-24 15:09:34 +0000 |
commit | bb92b0b9b210ee937a498970963d871e9ea6f0dc (patch) | |
tree | 9ac06380786425108e8184efce3e5e106a46d574 /configs/sip.conf.sample | |
parent | c1bd7a4b524fac75e9d6c065f83c986ee84820b0 (diff) |
Improve sample configuration files (bug #1125)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@3057 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'configs/sip.conf.sample')
-rwxr-xr-x | configs/sip.conf.sample | 170 |
1 files changed, 137 insertions, 33 deletions
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index c5e7146cf..c314b7099 100755 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -21,25 +21,34 @@ ; [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 +context=default ; Default context for incoming calls +;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=5060 ; 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 +;pedantic=yes ; Enable slow, pedantic checking for Pingtel ; and multiline formatted headers for strict ; SIP compatibility -;tos=lowdelay ; IP QoS parameter, either keyword or value - ; like tos=184 +;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 -;realm=asterisk ; Our global authentication realm ;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 +;disallow=all ; First disallow all codecs ;allow=ulaw ; Allow codecs in order of preference -;allow=ilbc +;allow=ilbc ; Note: codec order is respected only in [general] +;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 + ; Asterisk can register as a SIP user agent to a SIP proxy (provider) ; Format for the register statement is: @@ -56,14 +65,17 @@ context = default ; Default context for incoming calls ; ;register => 1234:password@mysipprovider.com ; -; Will call to the 's' extension +; This will pass incoming calls to the 's' extension ; ; -;register => 2345@mysipprovider.com/1234 +;register => 2345:password@sip_proxy/1234 ; -; Register 2345 at sip provider. Calls from this provider connect to local +; Register 2345 at sip provider 'sip_proxy'. 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 +; 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 = 200.201.202.203 ; Address that we're going to put in outbound SIP messages @@ -76,51 +88,143 @@ context = default ; Default context for incoming calls ; are: ;localnet=192.168.0.0/255.255.0.0; 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 -;[snomsip] +;----------------------------------------------------------------------------------- +; Users and peers have different settings available. Friends have all settings, +; since a friend is both a peer and a user +; +; User config options: Peer configuration: +; -------------------- ------------------- +; context context +; permit permit +; deny deny +; auth auth +; secret secret +; md5secret md5secret +; dtmfmode dtmfmode +; canreinvite canreinvite +; nat nat +; callgroup callgroup +; pickupgroup pickupgroup +; language language +; allow allow +; disallow disallow +; insecure insecure +; callerid +; accountcode +; amaflags +; incominglimit +; outgoinglimit +; restrictcid +; mailbox +; username +; template +; fromdomain +; fromuser +; host +; mask +; port +; qualify +; defaultip + + +;[sip_proxy] +; For incoming calls only. Example: FWD (Free World Dialup) +;type=user +;context=from-fwd + +;[sip_proxy-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 + + +;[xlite1] +;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)! +;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed ;type=friend -;secret=blah +;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 + + +;[snom] +;type=friend ; Friends place calls and receive calls +;context=from-sip ; Context for incoming calls from this user +;secret=blah +;host=dynamic ; This peer register with us ;dtmfmode=inband ; Choices are inband, rfc2833, or info -;defaultip=192.168.0.59 -;mailbox=1234,2345 ; Mailbox for message waiting indicator +;defaultip=192.168.0.59 ; IP used until peer registers +;mailbox=1234,2345 ; Mailboxes for message waiting indicator ;restrictcid=yes ; To have the callerid restriced -> sent as ANI -;insecure=yes ; To match a peer based by IP address only and not peer -;insecure=very ; To allow registered hosts to call without re-authenticating +;disallow=all +;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw! +;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator + ;[pingtel] ;type=friend ;username=pingtel ;secret=blah ;host=dynamic +;insecure=yes ; To match a peer based by IP address only and not peer +;insecure=very ; To allow registered hosts to call without re-authenticating ;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 ; We are in caller groups 1,3,4 +;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5 +;defaultip=192.168.0.60 ; IP address to use if peer has not registred -;callgroup=1,3-4 -;pickupgroup=1,3-4 -;defaultip=192.168.0.60 - -;[cisco] +;[cisco1] ;type=friend -;username=cisco +;username=cisco1 ;secret=blah +;qualify=200 ; Qualify peer is no more than 200ms away ;nat=yes ; This phone may be natted - ; Use IP address that packet is received from - ; instead of trusting SIP headers -;host=dynamic + ; Send SIP and RTP to IP address that packet is + ; received from instead of trusting SIP headers +;host=dynamic ; This device registers with us ;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] +;[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 |