Index: configs/extensions.conf.sample =================================================================== RCS file: /usr/cvsroot/asterisk/configs/extensions.conf.sample,v retrieving revision 1.21 diff -u -r1.21 extensions.conf.sample --- configs/extensions.conf.sample 26 Apr 2004 05:47:45 -0000 1.21 +++ configs/extensions.conf.sample 24 May 2004 07:11:04 -0000 @@ -96,6 +96,12 @@ [iaxtel700] exten => _91700XXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel) +; +; The SWITCH statement permits a server to share the dialplain with +; another server. Use with care: Reciprocal switch statements are not +; allowed (e.g. both A -> B and B -> A), and the switched server needs +; to be on-line or else dialing can be severly delayed. +; [iaxprovider] ;switch => IAX2/user:[key]@myserver/mycontext @@ -276,17 +282,29 @@ ; include => demo +; +; Extensions like the two below can be used for FWD, Nikotel, sipgate etc. +; Note that you must have a [sipprovider] section in sip.conf whereas +; the otherprovider.net example does not require such a peer definition +; +;exten => _41X.,1,Dial(SIP/${EXTEN:2}@sipprovider,,r) +;exten => _42X.,1,Dial(SIP/user:passwd@${EXTEN:2}@otherprovider.net,30,rT) -; Real extensions would go here. Generally you want real extensions to be 4 or 5 +; Real extensions would go here. Generally you want real extensions to be 4 or 5 ; digits long (although there is no such requirement) and start with a single ; digit that is fairly large (like 6 or 7) so that you have plenty of room to ; overlap extensions and menu options without conflict. You can alias them with ; names, too and use global variables - -;exten => 6275,1,Macro(stdexten,6275,${MARK}) ; assuming ${MARK} is something like Zap/2 -;exten => mark,1,Goto(6275|1) ; alias mark to 6275 -;exten => 6236,1,Macro(stdexten,6236,${WIL}) ; Ditto for wil +;exten => 6245,1,Dial(SIP/Grandstream1,20,rt) ; permit transfer +;exten => 6245,1,Dial(SIP/Grandstream1&SIP/Xlite1,20,rtT) +;exten => 6361,1,Dial(IAX2/JaneDoe,,rm) ; ring without time limit +;exten => 6389,1,Dial(MGCP/aaln/1@192.168.0.14) +;exten => 6394,1,Dial(Local/6275/n) ; this will dial ${MARK} + +;exten => 6275,1,Macro(stdexten,6275,${MARK}) ; assuming ${MARK} is something like Zap/2 +;exten => mark,1,Goto(6275|1) ; alias mark to 6275 +;exten => 6536,1,Macro(stdexten,6236,${WIL}) ; Ditto for wil ;exten => wil,1,Goto(6236|1) ; ; Some other handy things are an extension for checking voicemail via @@ -297,7 +315,7 @@ ; ; Or a conference room (you'll need to edit meetme.conf to enable this room) ; -;exten => 8600,1,Meetme,1234 +;exten => 8600,1,Meetme(1234) ; ; Or playing an announcement to the called party, as soon it answers ; Index: configs/modem.conf.sample =================================================================== RCS file: /usr/cvsroot/asterisk/configs/modem.conf.sample,v retrieving revision 1.6 diff -u -r1.6 modem.conf.sample --- configs/modem.conf.sample 22 Oct 2002 15:31:47 -0000 1.6 +++ configs/modem.conf.sample 24 May 2004 07:11:05 -0000 @@ -1,5 +1,5 @@ ; -; Internet Phone Jack +; isdn4linux ; ; Configuration file ; @@ -11,7 +11,8 @@ ; ; Modem Drivers to load ; -driver=aopen +driver=aopen ; modem by AOpen +;driver=i4l ; isdn4linux - an alternative to i4l is to use chan_capi ; ; Default language ; @@ -26,7 +27,7 @@ ; We can strip a given number of digits on outgoing dialing, so, for example ; you can have it dial "8871042" when given "98871042". ; -stripmsd=1 +stripmsd=0 ; ; Type of dialing ; @@ -45,7 +46,7 @@ ; ;device => /dev/ttyS3 ; -; ISDN example +; ISDN example (using i4l) ; ;msn=39907835 ;device => /dev/ttyI0 Index: configs/sip.conf.sample =================================================================== RCS file: /usr/cvsroot/asterisk/configs/sip.conf.sample,v retrieving revision 1.27 diff -u -r1.27 sip.conf.sample --- configs/sip.conf.sample 8 May 2004 20:58:24 -0000 1.27 +++ configs/sip.conf.sample 24 May 2004 07:11:05 -0000 @@ -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 @@ ; ;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 @@ ; 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