Index: configs/sip.conf.sample =================================================================== RCS file: /usr/cvsroot/asterisk/configs/sip.conf.sample,v retrieving revision 1.52 diff -u -r1.52 sip.conf.sample --- configs/sip.conf.sample 5 Jan 2005 15:15:12 -0000 1.52 +++ configs/sip.conf.sample 9 Jan 2005 10:47:48 -0000 @@ -1,5 +1,5 @@ ; -; SIP Configuration for Asterisk +; SIP Configuration example for Asterisk ; ; Syntax for specifying a SIP device in extensions.conf is ; SIP/devicename where devicename is defined in a section below. @@ -22,8 +22,8 @@ [general] context=default ; Default context for incoming calls -;allowguest=no ; Allow or reject guest calls (default is yes, this can also be set to 'osp' - ; if asterisk was compiled with OSP support. +;allowguest=no ; Allow or reject guest calls (default is yes, this can also be set to 'osp' + ; if asterisk was compiled with OSP support. ;recordhistory=yes ; Record SIP history by default ; (see sip history / sip no history) ;realm=mydomain.tld ; Realm for digest authentication @@ -42,8 +42,8 @@ ;pedantic=yes ; Enable slow, pedantic checking for Pingtel ; and multiline formatted headers for strict ; SIP compatibility (defaults to "no") -;tos=184 ; Set IP QoS to either a keyword or numeric val -;tos=lowdelay ; lowdelay,throughput,reliability,mincost,none +;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 ;defaultexpirey=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY @@ -52,7 +52,7 @@ ;disallow=all ; First disallow all codecs ;allow=ulaw ; Allow codecs in order of preference -;allow=ilbc ; Note: codec order is respected only in [general] +;allow=ilbc ; ;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 @@ -67,22 +67,16 @@ ; use 'never' to never use in-band signalling, even in cases ; where some buggy devices might not render it ;useragent=Asterisk PBX ; Allows you to change the user agent string -;nat=no ; NAT settings - ; yes = Always ignore info and assume NAT - ; no = Use NAT mode only according to RFC3581 - ; never = Never attempt NAT mode or RFC3581 support - ; route = Assume NAT, don't send rport - ; (work around more UNIDEN bugs) ;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address ; Note that promiscredir when redirects are made to the ; local system will cause loops since SIP is incapable + ; of performing a "hairpin" call. ;usereqphone = no ; If yes, ";user=phone" is added to uri that contains ; a valid phone number - ; of performing a "hairpin" call. ;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833 ; Other options: ; info : SIP INFO messages - ; inband : Inband audio + ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw) ;compactheaders = yes ; send compact sip headers. @@ -121,30 +115,47 @@ ; 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 +; (instead of type=friend) if you have calls in both directions ;registertimeout=20 ; retry registration calls every 20 seconds (default) +;---------------------------------------------- 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 = 200.201.202.203 ; 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 - ; You may add multiple local networks. A reasonable set of defaults - ; are: ;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 - ; usedl + ; used + ; You may add multiple local networks. A reasonable set of defaults + ; 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 +; 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 firewalls support of SIP+RTP ports. +; You configure Asterisk choice of RTP ports for incoming audio in rtp.conf +; +;nat=no ; Global NAT settings (Affects all peers and users) + ; yes = Always ignore info and assume NAT + ; no = Use NAT mode only according to RFC3581 + ; never = Never attempt NAT mode or RFC3581 support + ; route = Assume NAT, don't send rport + ; (work around more UNIDEN bugs) + ;----------------------------------------------------------------------------------- ; Users and peers have different settings available. Friends have all settings, ; since a friend is both a peer and a user @@ -191,29 +202,47 @@ ;[sip_proxy] ; For incoming calls only. Example: FWD (Free World Dialup) -;type=user +; We match on IP address of the proxy for incoming calls +; since we can not match on username (caller id) +;type=peer ;context=from-fwd +;host=fwd.pulver.com ;[sip_proxy-out] ;type=peer ; we only want to call out, not be called ;secret=guessit ;username=yourusername ; Authentication user for outbound proxies ;fromuser=yourusername ; Many SIP providers require this! +;fromdomain=provider.sip.domain ;host=box.provider.com ;usereqphone=yes ; This provider requires ";user=phone" on URI +;------------------------------------------------------------------------------ +; Definitions of locally connected SIP phones +; +; type = user a device that calls us +; type = peer a device we place calls to +; type = friend two configurations (peer+user) in one +; +; For local phones, type=friend works most of the time +; +; If you have one-way audio, you propably have NAT problems. +; If Asterisk is on a public IP, and the phone is inside of a NAT device +; you will need to configure nat option for those phones. +; Also, turn on qualify=yes to keep the nat session open + ;[grandstream1] -;type=friend ; either "friend" (peer+user), "peer" or "user" -;context=from-sip -;fromuser=grandstream1 ; overrides the callerid, e.g. required by FWD -;callerid=John Doe <1234> +;type=friend +;context=from-sip ; Where to start in the dialplan when this phone calls +;callerid=John Doe <1234> ; Full caller ID, to override the phones config ;host=192.168.0.23 ; we have a static but private IP address + ; No registration allowed ;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 ;incominglimit=1 ; permit only 1 outgoing call at a time ; from the phone to asterisk -;mailbox=1234@default ; mailbox 1234 in voicemail context "default" +;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! @@ -223,17 +252,16 @@ ;[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 +; 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 -;regexten=1234 ; When they register, create extension 1234 -;username=xlite1 +;regexten=1234 ; When they register, create extension 1234 ;callerid="Jane Smith" <5678> -;host=dynamic -;nat=yes ; X-Lite is behind a NAT router -;canreinvite=no ; Typically set to NO if behind NAT +;host=dynamic ; This device needs to register +;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=gsm ; GSM consumes far less bandwidth than ulaw ;allow=ulaw ;allow=alaw @@ -247,11 +275,10 @@ ;dtmfmode=inband ; Choices are inband, rfc2833, or info ;defaultip=192.168.0.59 ; IP used until peer registers ;username=snom ; Username to use in INVITE until peer registers -;mailbox=1234,2345 ; Mailboxes for message waiting indicator +;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator ;restrictcid=yes ; To have the callerid restriced -> sent as ANI ;disallow=all -;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw! -;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator +;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw! ;[polycom] @@ -271,7 +298,7 @@ ;username=pingtel ;secret=blah ;host=dynamic -;insecure=yes ; To match a peer based by IP address only and not peer +;insecure=yes ; To match a peer based by IP address only and not peer name ;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 @@ -286,7 +313,7 @@ ;secret=blah ;qualify=200 ; Qualify peer is no more than 200ms away ;nat=yes ; This phone may be natted - ; Send SIP and RTP to IP address that packet is + ; Send SIP and RTP to the 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 @@ -294,18 +321,6 @@ ; the caller to the callee. Some devices do not ; support this (especially if one of them is ; behind a NAT). -;defaultip=192.168.0.4 +;defaultip=192.168.0.4 ; IP address to use until registration +;username=goran ; Username to use when calling this device before registration -;[cisco2] -;type=friend -;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 -;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