-; extensions.conf
-; David Rowe 4 Jan 2010
+; extensions.conf - the Asterisk dial plan
;
+; Static extension configuration file, used by
+; the pbx_config module. This is where you configure all your
+; inbound and outbound calls in Asterisk.
+;
+; This configuration file is reloaded
+; - With the "extensions reload" command in the CLI
+; - With the "reload" command (that reloads everything) in the CLI
-; Designed for Mini Asterisk GUI. However you can hand modify as much
-; as you like, as GUI scripts read/and write regular extensions.conf
-; without messing with your custom dialplan
+;
+; The "General" category is for certain variables.
+;
+[general]
+;
+; If static is set to no, or omitted, then the pbx_config will rewrite
+; this file when extensions are modified. Remember that all comments
+; made in the file will be lost when that happens.
+;
+; XXX Not yet implemented XXX
+;
+static=yes
+;
+; if static=yes and writeprotect=no, you can save dialplan by
+; CLI command 'save dialplan' too
+;
+writeprotect=no
+;
+; If autofallthrough is set, then if an extension runs out of
+; things to do, it will terminate the call with BUSY, CONGESTION
+; or HANGUP depending on Asterisk's best guess. This is the default.
+;
+; If autofallthrough is not set, then if an extension runs out of
+; things to do, Asterisk will wait for a new extension to be dialed
+; (this is the original behavior of Asterisk 1.0 and earlier).
+;
+;autofallthrough=no
+;
+; If clearglobalvars is set, global variables will be cleared
+; and reparsed on an extensions reload, or Asterisk reload.
+;
+; If clearglobalvars is not set, then global variables will persist
+; through reloads, and even if deleted from the extensions.conf or
+; one of its included files, will remain set to the previous value.
+;
+; NOTE: A complication sets in, if you put your global variables into
+; the AEL file, instead of the extensions.conf file. With clearglobalvars
+; set, a 'reload' will often leave the globals vars cleared, because it
+; is not unusual to have extensions.conf (which will have no globals)
+; load after the extensions.ael file (where the global vars are stored).
+; So, with 'reload' in this particular situation, first the AEL file will
+; clear and then set all the global vars, then, later, when the extensions.conf
+; file is loaded, the global vars are all cleared, and then not set, because
+; they are not stored in the extensions.conf file.
+;
+clearglobalvars=no
+;
+; If priorityjumping is set to 'yes', then applications that support
+; 'jumping' to a different priority based on the result of their operations
+; will do so (this is backwards compatible behavior with pre-1.2 releases
+; of Asterisk). Individual applications can also be requested to do this
+; by passing a 'j' option in their arguments.
+;
+;priorityjumping=yes
+;
+; User context is where entries from users.conf are registered. The
+; default value is 'default'
+;
+;userscontext=default
+;
+; You can include other config files, use the #include command
+; (without the ';'). Note that this is different from the "include" command
+; that includes contexts within other contexts. The #include command works
+; in all asterisk configuration files.
+;#include "filename.conf"
+; The "Globals" category contains global variables that can be referenced
+; in the dialplan with the GLOBAL dialplan function:
+; ${GLOBAL(VARIABLE)}
+; ${${GLOBAL(VARIABLE)}} or ${text${GLOBAL(VARIABLE)}} or any hybrid
+; Unix/Linux environmental variables can be reached with the ENV dialplan
+; function: ${ENV(VARIABLE)}
+;
+[globals]
+CONSOLE=Console/dsp ; Console interface for demo
+;CONSOLE=Zap/1
+;CONSOLE=Phone/phone0
+IAXINFO=guest ; IAXtel username/password
+;IAXINFO=myuser:mypass
+TRUNK=Zap/g2 ; Trunk interface
+;
+; Note the 'g2' in the TRUNK variable above. It specifies which group (defined
+; in zapata.conf) to dial, i.e. group 2, and how to choose a channel to use in
+; the specified group. The four possible options are:
+;
+; g: select the lowest-numbered non-busy Zap channel
+; (aka. ascending sequential hunt group).
+; G: select the highest-numbered non-busy Zap channel
+; (aka. descending sequential hunt group).
+; r: use a round-robin search, starting at the next highest channel than last
+; time (aka. ascending rotary hunt group).
+; R: use a round-robin search, starting at the next lowest channel than last
+; time (aka. descending rotary hunt group).
+;
+TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
+;TRUNK=IAX2/user:pass@provider
-[general]
-static = yes
-writeprotect = no
-autofallthrough = yes
-clearglobalvars = no
-priorityjumping = no
+;
+; Any category other than "General" and "Globals" represent
+; extension contexts, which are collections of extensions.
+;
+; Extension names may be numbers, letters, or combinations
+; thereof. If an extension name is prefixed by a '_'
+; character, it is interpreted as a pattern rather than a
+; literal. In patterns, some characters have special meanings:
+;
+; X - any digit from 0-9
+; Z - any digit from 1-9
+; N - any digit from 2-9
+; [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9)
+; . - wildcard, matches anything remaining (e.g. _9011. matches
+; anything starting with 9011 excluding 9011 itself)
+; ! - wildcard, causes the matching process to complete as soon as
+; it can unambiguously determine that no other matches are possible
+;
+; For example the extension _NXXXXXX would match normal 7 digit dialings,
+; while _1NXXNXXXXXX would represent an area code plus phone number
+; preceded by a one.
+;
+; Each step of an extension is ordered by priority, which must
+; always start with 1 to be considered a valid extension. The priority
+; "next" or "n" means the previous priority plus one, regardless of whether
+; the previous priority was associated with the current extension or not.
+; The priority "same" or "s" means the same as the previously specified
+; priority, again regardless of whether the previous entry was for the
+; same extension. Priorities may be immediately followed by a plus sign
+; and another integer to add that amount (most useful with 's' or 'n').
+; Priorities may then also have an alias, or label, in
+; parenthesis after their name which can be used in goto situations
+;
+; Contexts contain several lines, one for each step of each
+; extension, which can take one of two forms as listed below,
+; with the first form being preferred. One may include another
+; context in the current one as well, optionally with a
+; date and time. Included contexts are included in the order
+; they are listed.
+;
+;[context]
+;exten => someexten,{priority|label{+|-}offset}[(alias)],application(arg1,arg2,...)
+;exten => someexten,{priority|label{+|-}offset}[(alias)],application,arg1|arg2...
+;
+; Timing list for includes is
+;
+; <time range>|<days of week>|<days of month>|<months>
+;
+; Note that ranges may be specified to wrap around the ends. Also, minutes are
+; fine-grained only down to the closest even minute.
+;
+;include => daytime|9:00-17:00|mon-fri|*|*
+;include => weekend|*|sat-sun|*|*
+;include => weeknights|17:02-8:58|mon-fri|*|*
+;
+; ignorepat can be used to instruct drivers to not cancel dialtone upon
+; receipt of a particular pattern. The most commonly used example is
+; of course '9' like this:
+;
+;ignorepat => 9
+;
+; so that dialtone remains even after dialing a 9.
+;
-[default]
+;
+; Sample entries for extensions.conf
+;
+;
+[dundi-e164-canonical]
+;
+; List canonical entries here
+;
+;exten => 12564286000,1,Macro(stdexten,6000,IAX2/foo)
+;exten => _125642860XX,1,Dial(IAX2/otherbox/${EXTEN:7})
+
+[dundi-e164-customers]
+;
+; If you are an ITSP or Reseller, list your customers here.
+;
+;exten => _12564286000,1,Dial(SIP/customer1)
+;exten => _12564286001,1,Dial(IAX2/customer2)
+
+[dundi-e164-via-pstn]
+;
+; If you are freely delivering calls to the PSTN, list them here
+;
+;exten => _1256428XXXX,1,Dial(Zap/g2/${EXTEN:7}) ; Expose all of 256-428
+;exten => _1256325XXXX,1,Dial(Zap/g2/${EXTEN:7}) ; Ditto for 256-325
+
+[dundi-e164-local]
+;
+; Context to put your dundi IAX2 or SIP user in for
+; full access
+;
+include => dundi-e164-canonical
+include => dundi-e164-customers
+include => dundi-e164-via-pstn
+
+[dundi-e164-switch]
+;
+; Just a wrapper for the switch
+;
+switch => DUNDi/e164
+
+[dundi-e164-lookup]
+;
+; Locally to lookup, try looking for a local E.164 solution
+; then try DUNDi if we don't have one.
+;
+include => dundi-e164-local
+include => dundi-e164-switch
+;
+; DUNDi can also be implemented as a Macro instead of using
+; the Local channel driver.
+;
+[macro-dundi-e164]
+;
+; ARG1 is the extension to Dial
+;
+; Extension "s" is not a wildcard extension that matches "anything".
+; In macros, it is the start extension. In most other cases,
+; you have to goto "s" to execute that extension.
+;
+; For wildcard matches, see above - all pattern matches start with
+; an underscore.
+exten => s,1,Goto(${ARG1},1)
+include => dundi-e164-lookup
-exten => 555,1,Answer
-exten => 555,2,Read(ip||3)
-exten => 555,3,Ping(10.130.1.${ip})
+;
+; Here are the entries you need to participate in the IAXTEL
+; call routing system. Most IAXTEL numbers begin with 1-700, but
+; there are exceptions. For more information, and to sign
+; up, please go to www.gnophone.com or www.iaxtel.com
+;
+[iaxtel700]
+exten => _91700XXXXXXX,1,Dial(IAX2/${GLOBAL(IAXINFO)}@iaxtel.com/${EXTEN:1}@iaxtel)
-exten => 556,1,Answer
-exten => 556,2,milliwatt
+;
+; The SWITCH statement permits a server to share the dialplan 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
-exten => 557,1,Answer
-exten => 557,2,Echo
+[trunkint]
+;
+; International long distance through trunk
+;
+exten => _9011.,1,Macro(dundi-e164,${EXTEN:4})
+exten => _9011.,n,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
-; Pre-configured analog extensions, depends on IP0X model and what modules
-; are installed. Some of these may map to FXO ports
+[trunkld]
+;
+; Long distance context accessed through trunk
+;
+exten => _91NXXNXXXXXX,1,Macro(dundi-e164,${EXTEN:1})
+exten => _91NXXNXXXXXX,n,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
-; mini_ext tag causes GUI to pick these up and display on GUI
+[trunklocal]
+;
+; Local seven-digit dialing accessed through trunk interface
+;
+exten => _9NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
-exten => 6001,1,Dial(Zap/1) ; mini_ext
-exten => 6002,1,Dial(Zap/2) ; mini_ext
-exten => 6003,1,Dial(Zap/3) ; mini_ext
-exten => 6004,1,Dial(Zap/4) ; mini_ext
-exten => 6005,1,Dial(Zap/5) ; mini_ext
-exten => 6006,1,Dial(Zap/6) ; mini_ext
-exten => 6007,1,Dial(Zap/7) ; mini_ext
-exten => 6008,1,Dial(Zap/8) ; mini_ext
+[trunktollfree]
+;
+; Long distance context accessed through trunk interface
+;
+exten => _91800NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
+exten => _91888NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
+exten => _91877NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
+exten => _91866NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
+
+[international]
+;
+; Master context for international long distance
+;
+ignorepat => 9
+include => longdistance
+include => trunkint
-; Pre-configured SIP-phone extensions. Primary use case is multiple SIP
-; extensions and FXO analog Ports
+[longdistance]
+;
+; Master context for long distance
+;
+ignorepat => 9
+include => local
+include => trunkld
-exten => 6011,1,Dial(SIP/6011) ; mini_ext
-exten => 6012,1,Dial(SIP/6012) ; mini_ext
-exten => 6013,1,Dial(SIP/6013) ; mini_ext
-exten => 6014,1,Dial(SIP/6014) ; mini_ext
-exten => 6015,1,Dial(SIP/6015) ; mini_ext
-exten => 6016,1,Dial(SIP/6016) ; mini_ext
-exten => 6017,1,Dial(SIP/6017) ; mini_ext
-exten => 6018,1,Dial(SIP/6018) ; mini_ext
-exten => 6019,1,Dial(SIP/6019) ; mini_ext
-exten => 6020,1,Dial(SIP/6020) ; mini_ext
-exten => 6021,1,Dial(SIP/6021) ; mini_ext
-exten => 6022,1,Dial(SIP/6022) ; mini_ext
+[local]
+;
+; Master context for local, toll-free, and iaxtel calls only
+;
+ignorepat => 9
+include => default
+include => trunklocal
+include => iaxtel700
+include => trunktollfree
+include => iaxprovider
+
+;Include parkedcalls (or the context you define in features conf)
+;to enable call parking.
+include => parkedcalls
+;
+; You can use an alternative switch type as well, to resolve
+; extensions that are not known here, for example with remote
+; IAX switching you transparently get access to the remote
+; Asterisk PBX
+;
+; switch => IAX2/user:password@bigserver/local
+;
+; An "lswitch" is like a switch but is literal, in that
+; variable substitution is not performed at load time
+; but is passed to the switch directly (presumably to
+; be substituted in the switch routine itself)
+;
+; lswitch => Loopback/12${EXTEN}@othercontext
+;
+; An "eswitch" is like a switch but the evaluation of
+; variable substitution is performed at runtime before
+; being passed to the switch routine.
+;
+; eswitch => IAX2/context@${CURSERVER}
+
+[macro-trunkdial]
+;
+; Standard trunk dial macro (hangs up on a dialstatus that should
+; terminate call)
+; ${ARG1} - What to dial
+;
+exten => s,1,Dial(${ARG1})
+exten => s,n,Goto(s-${DIALSTATUS},1)
+exten => s-NOANSWER,1,Hangup
+exten => s-BUSY,1,Hangup
+exten => _s-.,1,NoOp
+
+[macro-stdexten];
+;
+; Standard extension macro:
+; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
+; ${ARG2} - Device(s) to ring
+;
+exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum
+exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
+
+exten => s-NOANSWER,1,Voicemail(${ARG1},u) ; If unavailable, send to voicemail w/ unavail announce
+exten => s-NOANSWER,2,Goto(default,s,1) ; If they press #, return to start
+
+exten => s-BUSY,1,Voicemail(${ARG1},b) ; If busy, send to voicemail w/ busy announce
+exten => s-BUSY,2,Goto(default,s,1) ; If they press #, return to start
+
+exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer
+
+exten => a,1,VoicemailMain(${ARG1}) ; If they press *, send the user into VoicemailMain
+
+[macro-stdPrivacyexten];
+;
+; Standard extension macro:
+; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
+; ${ARG2} - Device(s) to ring
+; ${ARG3} - Optional DONTCALL context name to jump to (assumes the s,1 extension-priority)
+; ${ARG4} - Optional TORTURE context name to jump to (assumes the s,1 extension-priority)`
+;
+exten => s,1,Dial(${ARG2},20|p) ; Ring the interface, 20 seconds maximum, call screening
+ ; option (or use P for databased call screening)
+exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
+
+exten => s-NOANSWER,1,Voicemail(${ARG1},u) ; If unavailable, send to voicemail w/ unavail announce
+exten => s-NOANSWER,2,Goto(default,s,1) ; If they press #, return to start
+
+exten => s-BUSY,1,Voicemail(${ARG1},b) ; If busy, send to voicemail w/ busy announce
+exten => s-BUSY,2,Goto(default,s,1) ; If they press #, return to start
-;; Pre-configured mini-asterisk outgoing Analog group
+exten => s-DONTCALL,1,Goto(${ARG3},s,1) ; Callee chose to send this call to a polite "Don't call again" script.
-exten => _0.,1,Dial(Zap/g1/${EXTEN:1})
+exten => s-TORTURE,1,Goto(${ARG4},s,1) ; Callee chose to send this call to a telemarketer torture script.
-;; Pre-configured mini-asterisk outgoing VOIP line
+exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer
-exten => _1.,1,Dial(SIP/user/${EXTEN:1})
+exten => a,1,VoicemailMain(${ARG1}) ; If they press *, send the user into VoicemailMain
-;; Pre-configured incoming calls
+[macro-page];
+;
+; Paging macro:
+;
+; Check to see if SIP device is in use and DO NOT PAGE if they are
+;
+; ${ARG1} - Device to page
+
+exten => s,1,ChanIsAvail(${ARG1}|js) ; j is for Jump and s is for ANY call
+exten => s,n,GoToIf([${AVAILSTATUS} = "1"]?autoanswer:fail)
+exten => s,n(autoanswer),Set(_ALERT_INFO="RA") ; This is for the PolyComs
+exten => s,n,SIPAddHeader(Call-Info: Answer-After=0) ; This is for the Grandstream, Snoms, and Others
+exten => s,n,NoOp() ; Add others here and Post on the Wiki!!!!
+exten => s,n,Dial(${ARG1}||)
+exten => s,n(fail),Hangup
+
+
+[demo]
+;
+; We start with what to do when a call first comes in.
+;
+exten => s,1,Wait(1) ; Wait a second, just for fun
+exten => s,n,Answer ; Answer the line
+exten => s,n,Set(TIMEOUT(digit)=5) ; Set Digit Timeout to 5 seconds
+exten => s,n,Set(TIMEOUT(response)=10) ; Set Response Timeout to 10 seconds
+exten => s,n(restart),BackGround(demo-congrats) ; Play a congratulatory message
+exten => s,n(instruct),BackGround(demo-instruct) ; Play some instructions
+exten => s,n,WaitExten ; Wait for an extension to be dialed.
+
+exten => 2,1,BackGround(demo-moreinfo) ; Give some more information.
+exten => 2,n,Goto(s,instruct)
+
+exten => 3,1,Set(LANGUAGE()=fr) ; Set language to french
+exten => 3,n,Goto(s,restart) ; Start with the congratulations
+
+exten => 1000,1,Goto(default,s,1)
+;
+; We also create an example user, 1234, who is on the console and has
+; voicemail, etc.
+;
+exten => 1234,1,Playback(transfer,skip) ; "Please hold while..."
+ ; (but skip if channel is not up)
+exten => 1234,n,Macro(stdexten,1234,${GLOBAL(CONSOLE)})
+
+exten => 1235,1,Voicemail(1234,u) ; Right to voicemail
+
+exten => 1236,1,Dial(Console/dsp) ; Ring forever
+exten => 1236,n,Voicemail(1234,b) ; Unless busy
+
+;
+; # for when they're done with the demo
+;
+exten => #,1,Playback(demo-thanks) ; "Thanks for trying the demo"
+exten => #,n,Hangup ; Hang them up.
+
+;
+; A timeout and "invalid extension rule"
+;
+exten => t,1,Goto(#,1) ; If they take too long, give up
+exten => i,1,Playback(invalid) ; "That's not valid, try again"
-exten => s,1,Dial(SIP/6011) ;; mini-asterisk - don't remove this comment
+;
+; Create an extension, 500, for dialing the
+; Asterisk demo.
+;
+exten => 500,1,Playback(demo-abouttotry); Let them know what's going on
+exten => 500,n,Dial(IAX2/guest@misery.digium.com/s@default) ; Call the Asterisk demo
+exten => 500,n,Playback(demo-nogo) ; Couldn't connect to the demo site
+exten => 500,n,Goto(s,6) ; Return to the start over message.
+
+;
+; Create an extension, 600, for evaluating echo latency.
+;
+exten => 600,1,Playback(demo-echotest) ; Let them know what's going on
+exten => 600,n,Echo ; Do the echo test
+exten => 600,n,Playback(demo-echodone) ; Let them know it's over
+exten => 600,n,Goto(s,6) ; Start over
+
+;
+; You can use the Macro Page to intercom a individual user
+exten => 76245,1,Macro(page,SIP/Grandstream1)
+; or if your peernames are the same as extensions
+exten => _7XXX,1,Macro(page,SIP/${EXTEN})
+;
+;
+; System Wide Page at extension 7999
+;
+exten => 7999,1,Set(TIMEOUT(absolute)=60)
+exten => 7999,2,Page(Local/Grandstream1@page&Local/Xlite1@page&Local/1234@page/n|d)
+
+; Give voicemail at extension 8500
+;
+exten => 8500,1,VoicemailMain
+exten => 8500,n,Goto(s,6)
+;
+; Here's what a phone entry would look like (IXJ for example)
+;
+;exten => 1265,1,Dial(Phone/phone0,15)
+;exten => 1265,n,Goto(s,5)
+
+;
+; The page context calls up the page macro that sets variables needed for auto-answer
+; It is in is own context to make calling it from the Page() application as simple as
+; Local/{peername}@page
+;
+[page]
+exten => _X.,1,Macro(page,SIP/${EXTEN})
+
+;[mainmenu]
+;
+; Example "main menu" context with submenu
+;
+;exten => s,1,Answer
+;exten => s,n,Background(thanks) ; "Thanks for calling press 1 for sales, 2 for support, ..."
+;exten => s,n,WaitExten
+;exten => 1,1,Goto(submenu,s,1)
+;exten => 2,1,Hangup
+;include => default
+;
+;[submenu]
+;exten => s,1,Ringing ; Make them comfortable with 2 seconds of ringback
+;exten => s,n,Wait,2
+;exten => s,n,Background(submenuopts) ; "Thanks for calling the sales department. Press 1 for steve, 2 for..."
+;exten => s,n,WaitExten
+;exten => 1,1,Goto(default,steve,1)
+;exten => 2,1,Goto(default,mark,2)
+
+[default]
+;
+; By default we include the demo. In a production system, you
+; probably don't want to have the demo there.
+;
+include => demo
+
+exten => 6010,1,Dial(SIP/6010) ; mini_ext
+exten => 6011,1,Dial(SIP/6011) ; mini_ext
+
+;
+; An extension like the one below can be used for FWD, Nikotel, sipgate etc.
+; Note that you must have a [sipprovider] section in sip.conf
+;
+;exten => _41X.,1,Dial(SIP/${EXTEN:2}@sipprovider,,r)
+
+; 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 => 6245,hint,SIP/Grandstream1&SIP/Xlite1,Joe Schmoe ; Channel hints for presence
+;exten => 6245,1,Dial(SIP/Grandstream1,20,rt) ; permit transfer
+;exten => 6245,n(dial),Dial(${HINT},20,rtT) ; Use hint as listed
+;exten => 6245,n,Voicemail(6245,u) ; Voicemail (unavailable)
+;exten => 6245,s+1,Hangup ; s+1, same as n
+;exten => 6245,dial+101,Voicemail(6245,b) ; Voicemail (busy)
+;exten => 6361,1,Dial(IAX2/JaneDoe,,rm) ; ring without time limit
+;exten => 6389,1,Dial(MGCP/aaln/1@192.168.0.14)
+;exten => 6390,1,Dial(JINGLE/caller/callee) ; Dial via jingle using labels
+;exten => 6391,1,Dial(JINGLE/asterisk@digium.com/mogorman@astjab.org) ;Dial via jingle using asterisk as the transport and calling mogorman.
+;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)
+
+;If you want to subscribe to the status of a parking space, this is
+;how you do it. Subscribe to extension 6600 in sip, and you will see
+;the status of the first parking lot with this extensions' help
+;exten => 6600,hint,park:701@parkedcalls
+;exten => 6600,1,noop
+;
+; Some other handy things are an extension for checking voicemail via
+; voicemailmain
+;
+;exten => 8500,1,VoicemailMain
+;exten => 8500,n,Hangup
+;
+; Or a conference room (you'll need to edit meetme.conf to enable this room)
+;
+;exten => 8600,1,Meetme(1234)
+;
+; Or playing an announcement to the called party, as soon it answers
+;
+;exten = 8700,1,Dial(${MARK},30,A(/path/to/my/announcemsg))
+;
+; For more information on applications, just type "show applications" at your
+; friendly Asterisk CLI prompt.
+;
+; 'show application <command>' will show details of how you
+; use that particular application in this file, the dial plan.
+; 'show functions" will list all dialplan functions
+; 'show function <COMMAND>' will show you more information about
+; one function. Remember that function names are UPPER CASE.
--- /dev/null
+; extensions.conf
+; David Rowe 4 Jan 2010
+;
+
+; Designed for Mini Asterisk GUI. However you can hand modify as much
+; as you like, as GUI scripts read/and write regular extensions.conf
+; without messing with your custom dialplan
+
+
+[general]
+static = yes
+writeprotect = no
+autofallthrough = yes
+clearglobalvars = no
+priorityjumping = no
+
+[default]
+
+exten => 555,1,Answer
+exten => 555,2,Read(ip||3)
+exten => 555,3,Ping(10.130.1.${ip})
+
+exten => 556,1,Answer
+exten => 556,2,milliwatt
+
+exten => 557,1,Answer
+exten => 557,2,Echo
+
+; Pre-configured analog extensions, depends on IP0X model and what modules
+; are installed. Some of these may map to FXO ports
+
+; mini_ext tag causes GUI to pick these up and display on GUI
+
+exten => 6001,1,Dial(Zap/1) ; mini_ext
+exten => 6002,1,Dial(Zap/2) ; mini_ext
+exten => 6003,1,Dial(Zap/3) ; mini_ext
+exten => 6004,1,Dial(Zap/4) ; mini_ext
+exten => 6005,1,Dial(Zap/5) ; mini_ext
+exten => 6006,1,Dial(Zap/6) ; mini_ext
+exten => 6007,1,Dial(Zap/7) ; mini_ext
+exten => 6008,1,Dial(Zap/8) ; mini_ext
+
+; Pre-configured SIP-phone extensions. Primary use case is multiple SIP
+; extensions and FXO analog Ports
+
+exten => 6011,1,Dial(SIP/6011) ; mini_ext
+exten => 6012,1,Dial(SIP/6012) ; mini_ext
+exten => 6013,1,Dial(SIP/6013) ; mini_ext
+exten => 6014,1,Dial(SIP/6014) ; mini_ext
+exten => 6015,1,Dial(SIP/6015) ; mini_ext
+exten => 6016,1,Dial(SIP/6016) ; mini_ext
+exten => 6017,1,Dial(SIP/6017) ; mini_ext
+exten => 6018,1,Dial(SIP/6018) ; mini_ext
+exten => 6019,1,Dial(SIP/6019) ; mini_ext
+exten => 6020,1,Dial(SIP/6020) ; mini_ext
+exten => 6021,1,Dial(SIP/6021) ; mini_ext
+exten => 6022,1,Dial(SIP/6022) ; mini_ext
+
+;; Pre-configured mini-asterisk outgoing Analog group
+
+exten => _0.,1,Dial(Zap/g1/${EXTEN:1})
+
+;; Pre-configured mini-asterisk outgoing VOIP line
+
+exten => _1.,1,Dial(SIP/user/${EXTEN:1})
+
+;; Pre-configured incoming calls
+
+exten => s,1,Dial(SIP/6011) ;; mini-asterisk - don't remove this comment
--- /dev/null
+
+; Inter-Asterisk eXchange driver definition
+;
+; This configuration is re-read at reload
+; or with the CLI command
+; reload chan_iax2.so
+;
+; General settings, like port number to bind to, and
+; an option address (the default is to bind to all
+; local addresses).
+;
+[general]
+;bindport=4569 ; bindport and bindaddr may be specified
+; ; NOTE: bindport must be specified BEFORE
+ ; bindaddr or may be specified on a specific
+ ; bindaddr if followed by colon and port
+ ; (e.g. bindaddr=192.168.0.1:4569)
+;bindaddr=192.168.0.1 ; more than once to bind to multiple
+; ; addresses, but the first will be the
+; ; default
+;
+; Set iaxcompat to yes if you plan to use layered switches or
+; some other scenario which may cause some delay when doing a
+; lookup in the dialplan. It incurs a small performance hit to
+; enable it. This option causes Asterisk to spawn a separate thread
+; when it receives an IAX DPREQ (Dialplan Request) instead of
+; blocking while it waits for a response.
+;
+;iaxcompat=yes
+;
+; Disable UDP checksums (if nochecksums is set, then no checkums will
+; be calculated/checked on systems supporting this feature)
+;
+;nochecksums=no
+;
+;
+; For increased security against brute force password attacks
+; enable "delayreject" which will delay the sending of authentication
+; reject for REGREQ or AUTHREP if there is a password.
+;
+;delayreject=yes
+;
+; You may specify a global default AMA flag for iaxtel calls. It must be
+; one of 'default', 'omit', 'billing', or 'documentation'. These flags
+; are used in the generation of call detail records.
+;
+;amaflags=default
+;
+; ADSI (Analog Display Services Interface) can be enabled if you have
+; (or may have) ADSI compatible CPE equipment
+;
+;adsi=no
+;
+; You may specify a default account for Call Detail Records in addition
+; to specifying on a per-user basis
+;
+;accountcode=lss0101
+;
+; You may specify a global default language for users.
+; Can be specified also on a per-user basis
+; If omitted, will fallback to english
+;
+;language=en
+;
+; This option specifies a preference for which music on hold class this channel
+; should listen to when put on hold if the music class has not been set on the
+; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
+; channel putting this one on hold did not suggest a music class.
+;
+; If this option is set to "passthrough", then the hold message will always be
+; passed through as signalling instead of generating hold music locally.
+;
+; This option may be specified globally, or on a per-user or per-peer basis.
+;
+;mohinterpret=default
+;
+; This option specifies which music on hold class to suggest to the peer channel
+; when this channel places the peer on hold. It may be specified globally or on
+; a per-user or per-peer basis.
+;
+;mohsuggest=default
+;
+; Specify bandwidth of low, medium, or high to control which codecs are used
+; in general.
+;
+bandwidth=low
+;
+; You can also fine tune codecs here using "allow" and "disallow" clauses
+; with specific codecs. Use "all" to represent all formats.
+;
+;allow=all ; same as bandwidth=high
+;disallow=g723.1 ; Hm... Proprietary, don't use it...
+disallow=lpc10 ; Icky sound quality... Mr. Roboto.
+;allow=gsm ; Always allow GSM, it's cool :)
+;
+
+; You can adjust several parameters relating to the jitter buffer.
+; The jitter buffer's function is to compensate for varying
+; network delay.
+;
+; All the jitter buffer settings except dropcount are in milliseconds.
+; The jitter buffer works for INCOMING audio - the outbound audio
+; will be dejittered by the jitter buffer at the other end.
+;
+; jitterbuffer=yes|no: global default as to whether you want
+; the jitter buffer at all.
+;
+; forcejitterbuffer=yes|no: in the ideal world, when we bridge VoIP channels
+; we don't want to do jitterbuffering on the switch, since the endpoints
+; can each handle this. However, some endpoints may have poor jitterbuffers
+; themselves, so this option will force * to always jitterbuffer, even in this
+; case.
+;
+; maxjitterbuffer: a maximum size for the jitter buffer.
+; Setting a reasonable maximum here will prevent the call delay
+; from rising to silly values in extreme situations; you'll hear
+; SOMETHING, even though it will be jittery.
+;
+; resyncthreshold: when the jitterbuffer notices a significant change in delay
+; that continues over a few frames, it will resync, assuming that the change in
+; delay was caused by a timestamping mix-up. The threshold for noticing a
+; change in delay is measured as twice the measured jitter plus this resync
+; threshold.
+; Resyncing can be disabled by setting this parameter to -1.
+;
+; maxjitterinterps: the maximum number of interpolation frames the jitterbuffer
+; should return in a row. Since some clients do not send CNG/DTX frames to
+; indicate silence, the jitterbuffer will assume silence has begun after
+; returning this many interpolations. This prevents interpolating throughout
+; a long silence.
+;
+
+jitterbuffer=no
+forcejitterbuffer=no
+;dropcount=2
+;maxjitterbuffer=1000
+;maxjitterinterps=10
+;resyncthreshold=1000
+;maxexcessbuffer=80
+;minexcessbuffer=10
+;jittershrinkrate=1
+
+;trunkfreq=20 ; How frequently to send trunk msgs (in ms)
+
+; Should we send timestamps for the individual sub-frames within trunk frames?
+; There is a small bandwidth use for these (less than 1kbps/call), but they
+; ensure that frame timestamps get sent end-to-end properly. If both ends of
+; all your trunks go directly to TDM, _and_ your trunkfreq equals the frame
+; length for your codecs, you can probably suppress these. The receiver must
+; also support this feature, although they do not also need to have it enabled.
+;
+; trunktimestamps=yes
+;
+; Minimum and maximum amounts of time that IAX peers can request as
+; a registration expiration interval (in seconds).
+; minregexpire = 60
+; maxregexpire = 60
+;
+; IAX helper threads
+; Establishes the number of iax helper threads to handle I/O.
+; iaxthreadcount = 10
+; Establishes the number of extra dynamic threads that may be spawned to handle I/O
+; iaxmaxthreadcount = 100
+;
+; We can register with another IAX server to let him know where we are
+; in case we have a dynamic IP address for example
+;
+; Register with tormenta using username marko and password secretpass
+;
+;register => marko:secretpass@tormenta.linux-support.net
+;
+; Register joe at remote host with no password
+;
+;register => joe@remotehost:5656
+;
+; Register marko at tormenta.linux-support.net using RSA key "torkey"
+;
+;register => marko:[torkey]@tormenta.linux-support.net
+;
+; Sample Registration for iaxtel
+;
+; Visit http://www.iaxtel.com to register with iaxtel. Replace "user"
+; and "pass" with your username and password for iaxtel. Incoming
+; calls arrive at the "s" extension of "default" context.
+;
+;register => user:pass@iaxtel.com
+;
+; Sample Registration for IAX + FWD
+;
+; To register using IAX with FWD, it must be enabled by visiting the URL
+; http://www.fwdnet.net/index.php?section_id=112
+;
+; Note that you need an extension in you default context which matches
+; your free world dialup number. Please replace "FWDNumber" with your
+; FWD number and "passwd" with your password.
+;
+;register => FWDNumber:passwd@iax.fwdnet.net
+;
+;
+; You can disable authentication debugging to reduce the amount of
+; debugging traffic.
+;
+;authdebug=no
+;
+; See doc/README.tos for a description of the tos parameters.
+;tos=ef
+;
+; If regcontext is specified, Asterisk will dynamically create and destroy
+; a NoOp priority 1 extension for a given peer who registers or unregisters
+; with us. The actual extension is the 'regexten' parameter of the registering
+; peer or its name if 'regexten' is not provided. More than one regexten
+; may be supplied if they are separated by '&'. Patterns may be used in
+; regexten.
+;
+;regcontext=iaxregistrations
+;
+; If we don't get ACK to our NEW within 2000ms, and autokill is set to yes,
+; then we cancel the whole thing (that's enough time for one retransmission
+; only). This is used to keep things from stalling for a long time for a host
+; that is not available, but would be ill advised for bad connections. In
+; addition to 'yes' or 'no' you can also specify a number of milliseconds.
+; See 'qualify' for individual peers to turn on for just a specific peer.
+;
+autokill=yes
+;
+; codecpriority controls the codec negotiation of an inbound IAX call.
+; This option is inherited to all user entities. It can also be defined
+; in each user entity separately which will override the setting in general.
+;
+; The valid values are:
+;
+; caller - Consider the callers preferred order ahead of the host's.
+; host - Consider the host's preferred order ahead of the caller's.
+; disabled - Disable the consideration of codec preference altogether.
+; (this is the original behaviour before preferences were added)
+; reqonly - Same as disabled, only do not consider capabilities if
+; the requested format is not available the call will only
+; be accepted if the requested format is available.
+;
+; The default value is 'host'
+;
+;codecpriority=host
+
+;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
+ ; just like friends added from the config file only on a
+ ; as-needed basis? (yes|no)
+
+;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
+ ; If set to yes, when a IAX2 peer registers successfully,
+ ; the ip address, the origination port, the registration period,
+ ; and the username of the peer will be set to database via realtime.
+ ; If not present, defaults to 'yes'.
+
+;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
+ ; as if it had just registered? (yes|no|<seconds>)
+ ; If set to yes, when the registration expires, the friend will
+ ; vanish from the configuration until requested again.
+ ; If set to an integer, friends expire within this number of
+ ; seconds instead of the registration interval.
+
+;rtignoreregexpire=yes ; When reading a peer from Realtime, if the peer's registration
+ ; has expired based on its registration interval, used the stored
+ ; address information regardless. (yes|no)
+
+; Guest sections for unauthenticated connection attempts. Just specify an
+; empty secret, or provide no secret section.
+;
+[guest]
+type=user
+context=default
+callerid="Guest IAX User"
+
+;
+; Trust Caller*ID Coming from iaxtel.com
+;
+[iaxtel]
+type=user
+context=default
+auth=rsa
+inkeys=iaxtel
+
+;
+; Trust Caller*ID Coming from iax.fwdnet.net
+;
+[iaxfwd]
+type=user
+context=default
+auth=rsa
+inkeys=freeworlddialup
+
+;
+; Trust callerid delivered over DUNDi/e164
+;
+;
+;[dundi]
+;type=user
+;dbsecret=dundi/secret
+;context=dundi-e164-local
+
+;
+; Further user sections may be added, specifying a context and a secret used
+; for connections with that given authentication name. Limited IP based
+; access control is allowed by use of "allow" and "deny" keywords. Multiple
+; rules are permitted. Multiple permitted contexts may be specified, in
+; which case the first will be the default. You can also override caller*ID
+; so that when you receive a call you set the Caller*ID to be what you want
+; instead of trusting what the remote user provides
+;
+; There are three authentication methods that are supported: md5, plaintext,
+; and rsa. The least secure is "plaintext", which sends passwords cleartext
+; across the net. "md5" uses a challenge/response md5 sum arrangement, but
+; still requires both ends have plain text access to the secret. "rsa" allows
+; unidirectional secret knowledge through public/private keys. If "rsa"
+; authentication is used, "inkeys" is a list of acceptable public keys on the
+; local system that can be used to authenticate the remote peer, separated by
+; the ":" character. "outkey" is a single, private key to use to authenticate
+; to the other side. Public keys are named /var/lib/asterisk/keys/<name>.pub
+; while private keys are named /var/lib/asterisk/keys/<name>.key. Private
+; keys should always be 3DES encrypted.
+;
+;
+; NOTE: All hostnames and IP addresses in this file are for example purposes
+; only; you should not expect any of them to actually be available for
+; your use.
+;
+;
+;[markster]
+;type=user
+;context=default
+;context=local
+;auth=md5,plaintext,rsa
+;secret=markpasswd
+;setvar=foo=bar
+;dbsecret=mysecrets/place ; Secrets can be stored in astdb, too
+;transfer=no ; Disable IAX native transfer
+;transfer=mediaonly ; When doing IAX native transfers, transfer
+ ; only media stream
+;jitterbuffer=yes ; Override global setting an enable jitter buffer
+; ; for this user
+;maxauthreq=10 ; Set maximum number of outstanding AUTHREQs waiting for replies. Any further authentication attempts will be blocked
+; ; if this limit is reached until they expire or a reply is received.
+;callerid="Mark Spencer" <(256) 428-6275>
+;deny=0.0.0.0/0.0.0.0
+;accountcode=markster0101
+;permit=209.16.236.73/255.255.255.0
+;language=en ; Use english as default language
+;
+; Peers may also be specified, with a secret and
+; a remote hostname.
+;
+[demo]
+type=peer
+username=asterisk
+secret=supersecret
+host=216.207.245.47
+;sendani=no
+;host=asterisk.linux-support.net
+;port=5036
+;mask=255.255.255.255
+;qualify=yes ; Make sure this peer is alive
+;qualifysmoothing = yes ; use an average of the last two PONG
+ ; results to reduce falsely detected LAGGED hosts
+ ; Default: Off
+;qualifyfreqok = 60000 ; how frequently to ping the peer when
+ ; everything seems to be ok, in milliseconds
+;qualifyfreqnotok = 10000 ; how frequently to ping the peer when it's
+ ; either LAGGED or UNAVAILABLE, in milliseconds
+;jitterbuffer=no ; Turn off jitter buffer for this peer
+
+;
+; Peers can remotely register as well, so that they can be mobile. Default
+; IP's can also optionally be given but are not required. Caller*ID can be
+; suggested to the other side as well if it is for example a phone instead of
+; another PBX.
+;
+
+;[dynamichost]
+;host=dynamic
+;secret=mysecret
+;mailbox=1234 ; Notify about mailbox 1234
+;inkeys=key1:key2
+;peercontext=local ; Default context to request for calls to peer
+;defaultip=216.207.245.34
+;callerid="Some Host" <(256) 428-6011>
+;
+
+;
+;[biggateway]
+;type=peer
+;host=192.168.0.1
+;context=*
+;secret=myscret
+;trunk=yes ; Use IAX2 trunking with this host
+;timezone=America/New_York ; Set a timezone for the date/time IE
+;
+
+;
+; Friends are a short cut for creating a user and
+; a peer with the same values.
+;
+;[marko]
+;type=friend
+;host=dynamic
+;regexten=1234
+;secret=moofoo ; Multiple secrets may be specified. For a "user", all
+;secret=foomoo ; specified entries will be accepted as valid. For a "peer",
+;secret=shazbot ; only the last specified secret will be used.
+;context=default
+;permit=0.0.0.0/0.0.0.0
+
--- /dev/null
+;
+; 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.
+;
+; 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
+;
+; reload chan_sip.so Reload configuration file
+; Active SIP peers will not be reconfigured
+;
+
+[general]
+context=default ; Default context for incoming calls
+allowguest=no ; Allow or reject guest calls (default is yes)
+allowoverlap=no ; Disable overlap dialing support. (Default is yes)
+;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
+ ; Default is enabled
+;realm=mydomain.tld ; Realm for digest authentication
+ ; defaults to "asterisk". If you set a system name in
+ ; asterisk.conf, it defaults to that system name
+ ; Realms MUST be globally unique according to RFC 3261
+ ; Set this to your host name or domain name
+bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
+ ; bindport is the local UDP port that Asterisk will listen on
+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
+ ; Disabling DNS SRV lookups disables the
+ ; ability to place SIP calls based on domain
+ ; names to some other SIP users on the Internet
+
+;domain=mydomain.tld ; Set default domain for this host
+ ; If configured, Asterisk will only allow
+ ; INVITE and REFER to non-local domains
+ ; Use "sip show domains" to list local domains
+;pedantic=yes ; Enable checking of tags in headers,
+ ; international character conversions in URIs
+ ; and multiline formatted headers for strict
+ ; SIP compatibility (defaults to "no")
+
+; See doc/README.tos for a description of these parameters.
+;tos_sip=cs3 ; Sets TOS for SIP packets.
+;tos_audio=ef ; Sets TOS for RTP audio packets.
+;tos_video=af41 ; Sets TOS for RTP video packets.
+
+;maxexpiry=3600 ; Maximum allowed time of incoming registrations
+ ; and subscriptions (seconds)
+;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
+;defaultexpiry=120 ; Default length of incoming/outgoing registration
+;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
+ ; Defaults to 100 ms
+;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
+;checkmwi=10 ; Default time between mailbox checks for peers
+;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC
+ ; fully. Enable this option to not get error messages
+ ; when sending MWI to phones with this bug.
+;vmexten=voicemail ; dialplan extension to reach mailbox sets the
+ ; Message-Account in the MWI notify message
+ ; defaults to "asterisk"
+;disallow=all ; First disallow all codecs
+;allow=ulaw ; Allow codecs in order of preference
+;allow=ilbc ; see doc/rtp-packetization for framing options
+;
+; This option specifies a preference for which music on hold class this channel
+; should listen to when put on hold if the music class has not been set on the
+; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
+; channel putting this one on hold did not suggest a music class.
+;
+; This option may be specified globally, or on a per-user or per-peer basis.
+;
+;mohinterpret=default
+;
+; This option specifies which music on hold class to suggest to the peer channel
+; when this channel places the peer on hold. It may be specified globally or on
+; a per-user or per-peer basis.
+;
+;mohsuggest=default
+;
+;language=en ; Default language setting for all users/peers
+ ; This may also be set for individual users/peers
+;relaxdtmf=yes ; Relax dtmf handling
+;trustrpid = no ; If Remote-Party-ID should be trusted
+;sendrpid = yes ; If Remote-Party-ID should be sent
+;progressinband=never ; If we should generate in-band ringing always
+ ; use 'never' to never use in-band signalling, even in cases
+ ; where some buggy devices might not render it
+ ; Valid values: yes, no, never Default: never
+;useragent=Asterisk PBX ; Allows you to change the user agent string
+;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 Asterisk is incapable
+ ; of performing a "hairpin" call.
+;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
+ ; a valid phone number
+;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
+ ; Other options:
+ ; info : SIP INFO messages
+ ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
+ ; auto : Use rfc2833 if offered, inband otherwise
+
+;compactheaders = yes ; send compact sip headers.
+;
+;videosupport=yes ; Turn on support for SIP video. You need to turn this on
+ ; in the this section to get any video support at all.
+ ; You can turn it off on a per peer basis if the general
+ ; video support is enabled, but you can't enable it for
+ ; one peer only without enabling in the general section.
+;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
+ ; Videosupport and maxcallbitrate is settable
+ ; for peers and users as well
+;callevents=no ; generate manager events when sip ua
+ ; performs events (e.g. hold)
+;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
+ ; for any reason, always reject with '401 Unauthorized'
+ ; instead of letting the requester know whether there was
+ ; a matching user or peer for their request
+
+;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
+ ; order instead of RFC3551 packing order (this is required
+ ; for Sipura and Grandstream ATAs, among others). This is
+ ; contrary to the RFC3551 specification, the peer _should_
+ ; be negotiating AAL2-G726-32 instead :-(
+
+;matchexterniplocally = yes ; Only substitute the externip or externhost setting if it matches
+ ; your localnet setting. Unless you have some sort of strange network
+ ; setup you will not need to enable this.
+
+;
+; If regcontext is specified, Asterisk will dynamically create and destroy a
+; NoOp priority 1 extension for a given peer who registers or unregisters with
+; us and have a "regexten=" configuration item.
+; Multiple contexts may be specified by separating them with '&'. The
+; actual extension is the 'regexten' parameter of the registering peer or its
+; name if 'regexten' is not provided. If more than one context is provided,
+; the context must be specified within regexten by appending the desired
+; context after '@'. More than one regexten may be supplied if they are
+; separated by '&'. Patterns may be used in regexten.
+;
+;regcontext=sipregistrations
+;
+;--------------------------- RTP timers ----------------------------------------------------
+; These timers are currently used for both audio and video streams. The RTP timeouts
+; are only applied to the audio channel.
+; The settings are settable in the global section as well as per device
+;
+;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
+ ; on the audio channel
+ ; when we're not on hold. This is to be able to hangup
+ ; a call in the case of a phone disappearing from the net,
+ ; like a powerloss or grandma tripping over a cable.
+;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
+ ; on the audio channel
+ ; when we're on hold (must be > rtptimeout)
+;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open
+ ; (default is off - zero)
+;--------------------------- SIP DEBUGGING ---------------------------------------------------
+;sipdebug = yes ; Turn on SIP debugging by default, from
+ ; the moment the channel loads this configuration
+;recordhistory=yes ; Record SIP history by default
+ ; (see sip history / sip no history)
+;dumphistory=yes ; Dump SIP history at end of SIP dialogue
+ ; SIP history is output to the DEBUG logging channel
+
+
+;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
+; You can subscribe to the status of extensions with a "hint" priority
+; (See extensions.conf.sample for examples)
+; chan_sip support two major formats for notifications: dialog-info and SIMPLE
+;
+; You will get more detailed reports (busy etc) if you have a call limit set
+; for a device. When the call limit is filled, we will indicate busy. Note that
+; you need at least 2 in order to be able to do attended transfers.
+;
+; For queues, you will need this level of detail in status reporting, regardless
+; if you use SIP subscriptions. Queues and manager use the same internal interface
+; for reading status information.
+;
+; Note: Subscriptions does not work if you have a realtime dialplan and use the
+; realtime switch.
+;
+;allowsubscribe=no ; Disable support for subscriptions. (Default is yes)
+;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
+ ; Useful to limit subscriptions to local extensions
+ ; Settable per peer/user also
+;notifyringing = yes ; Notify subscriptions on RINGING state (default: no)
+;notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
+ ; Turning on notifyringing and notifyhold will add a lot
+ ; more database transactions if you are using realtime.
+;limitonpeers = yes ; Apply call limits on peers only. This will improve
+ ; status notification when you are using type=friend
+ ; Inbound calls, that really apply to the user part
+ ; of a friend will now be added to and compared with
+ ; the peer limit instead of applying two call limits,
+ ; one for the peer and one for the user.
+
+;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT -----------------------
+;
+; This setting is available in the [general] section as well as in device configurations.
+; Setting this to yes, enables T.38 fax (UDPTL) passthrough on SIP to SIP calls, provided
+; both parties have T38 support enabled in their Asterisk configuration
+; This has to be enabled in the general section for all devices to work. You can then
+; disable it on a per device basis.
+;
+; T.38 faxing only works in SIP to SIP calls, with no local or agent channel being used.
+;
+; t38pt_udptl = yes ; Default false
+;
+;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------
+; 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 => 1234:password@mysipprovider.com
+;
+; This will pass incoming calls to the 's' extension
+;
+;
+;register => 2345:password@sip_proxy/1234
+;
+; Register 2345 at sip provider 'sip_proxy'. Calls from this provider
+; connect to local extension 1234 in extensions.conf, default 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
+
+;registertimeout=20 ; retry registration calls every 20 seconds (default)
+;registerattempts=10 ; Number of registration attempts before we give up
+ ; 0 = continue forever, hammering the other server
+ ; until it accepts the registration
+ ; Default is 0 tries, continue forever
+
+;----------------------------------------- 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
+;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
+ ; 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
+; firewall's 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 (;rport)
+ ; never = Never attempt NAT mode or RFC3581 support
+ ; route = Assume NAT, don't send rport
+ ; (work around more UNIDEN bugs)
+
+;----------------------------------- MEDIA HANDLING --------------------------------
+; By default, Asterisk tries to re-invite the audio to an optimal path. If there's
+; no reason for Asterisk to stay in the media path, the media will be redirected.
+; This does not really work with in the case where Asterisk is outside and have
+; clients on the inside of a NAT. In that case, you want to set canreinvite=nonat
+;
+;canreinvite=yes ; 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).
+ ; The default setting is YES. If you have all clients
+ ; behind a NAT, or for some other reason wants Asterisk to
+ ; stay in the audio path, you may want to turn this off.
+
+ ; In Asterisk 1.4 this setting also affect direct RTP
+ ; at call setup (a new feature in 1.4 - setting up the
+ ; call directly between the endpoints instead of sending
+ ; a re-INVITE).
+
+;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
+ ; the call directly with media peer-2-peer without re-invites.
+ ; Will not work for video and cases where the callee sends
+ ; RTP payloads and fmtp headers in the 200 OK that does not match the
+ ; callers INVITE.
+
+;canreinvite=nonat ; An additional option is to allow media path redirection
+ ; (reinvite) but only when the peer where the media is being
+ ; sent is known to not be behind a NAT (as the RTP core can
+ ; determine it based on the apparent IP address the media
+ ; arrives from).
+
+;canreinvite=update ; Yet a third option... use UPDATE for media path redirection,
+ ; instead of INVITE. This can be combined with 'nonat', as
+ ; 'canreinvite=update,nonat'. It implies 'yes'.
+
+;----------------------------------------- REALTIME SUPPORT ------------------------
+; For additional information on ARA, the Asterisk Realtime Architecture,
+; please read realtime.txt and extconfig.txt in the /doc directory of the
+; source code.
+;
+;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
+ ; just like friends added from the config file only on a
+ ; as-needed basis? (yes|no)
+
+;rtsavesysname=yes ; Save systemname in realtime database at registration
+ ; Default= no
+
+;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
+ ; If set to yes, when a SIP UA registers successfully, the ip address,
+ ; the origination port, the registration period, and the username of
+ ; the UA will be set to database via realtime.
+ ; If not present, defaults to 'yes'.
+;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
+ ; as if it had just registered? (yes|no|<seconds>)
+ ; If set to yes, when the registration expires, the friend will
+ ; vanish from the configuration until requested again. If set
+ ; to an integer, friends expire within this number of seconds
+ ; instead of the registration interval.
+
+;ignoreregexpire=yes ; Enabling this setting has two functions:
+ ;
+ ; For non-realtime peers, when their registration expires, the
+ ; information will _not_ be removed from memory or the Asterisk database
+ ; if you attempt to place a call to the peer, the existing information
+ ; will be used in spite of it having expired
+ ;
+ ; For realtime peers, when the peer is retrieved from realtime storage,
+ ; the registration information will be used regardless of whether
+ ; it has expired or not; if it expires while the realtime peer
+ ; is still in memory (due to caching or other reasons), the
+ ; information will not be removed from realtime storage
+
+;----------------------------------------- SIP DOMAIN SUPPORT ------------------------
+; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
+; domains, each of which can direct the call to a specific context if desired.
+; By default, all domains are accepted and sent to the default context or the
+; context associated with the user/peer placing the call.
+; Domains can be specified using:
+; domain=<domain>[,<context>]
+; Examples:
+; domain=myasterisk.dom
+; domain=customer.com,customer-context
+;
+; In addition, all the 'default' domains associated with a server should be
+; added if incoming request filtering is desired.
+; autodomain=yes
+;
+; To disallow requests for domains not serviced by this server:
+; allowexternaldomains=no
+
+;domain=mydomain.tld,mydomain-incoming
+ ; Add domain and configure incoming context
+ ; for external calls to this domain
+;domain=1.2.3.4 ; Add IP address as local domain
+ ; You can have several "domain" settings
+;allowexternalinvites=no ; Disable INVITE and REFER to non-local domains
+ ; Default is yes
+;autodomain=yes ; Turn this on to have Asterisk add local host
+ ; name and local IP to domain list.
+
+; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
+ ; non-peers, use your primary domain "identity"
+ ; for From: headers instead of just your IP
+ ; address. This is to be polite and
+ ; it may be a mandatory requirement for some
+ ; destinations which do not have a prior
+ ; account relationship with your server.
+
+;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
+; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
+ ; SIP channel. Defaults to "no". An enabled jitterbuffer will
+ ; be used only if the sending side can create and the receiving
+ ; side can not accept jitter. The SIP channel can accept jitter,
+ ; thus a jitterbuffer on the receive SIP side will be used only
+ ; if it is forced and enabled.
+
+; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
+ ; channel. Defaults to "no".
+
+; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
+
+; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
+ ; resynchronized. Useful to improve the quality of the voice, with
+ ; big jumps in/broken timestamps, usually sent from exotic devices
+ ; and programs. Defaults to 1000.
+
+; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
+ ; channel. Two implementations are currently available - "fixed"
+ ; (with size always equals to jbmaxsize) and "adaptive" (with
+ ; variable size, actually the new jb of IAX2). Defaults to fixed.
+
+; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
+;-----------------------------------------------------------------------------------
+
+[authentication]
+; Global credentials for outbound calls, i.e. when a proxy challenges your
+; Asterisk server for authentication. These credentials override
+; any credentials in peer/register definition if realm is matched.
+;
+; This way, Asterisk can authenticate for outbound calls to other
+; realms. We match realm on the proxy challenge and pick an set of
+; credentials from this list
+; Syntax:
+; auth = <user>:<secret>@<realm>
+; auth = <user>#<md5secret>@<realm>
+; Example:
+;auth=mark:topsecret@digium.com
+;
+; You may also add auth= statements to [peer] definitions
+; Peer auth= override all other authentication settings if we match on realm
+
+;------------------------------------------------------------------------------
+; 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
+; callingpres callingpres
+; permit permit
+; deny deny
+; secret secret
+; md5secret md5secret
+; dtmfmode dtmfmode
+; canreinvite canreinvite
+; nat nat
+; callgroup callgroup
+; pickupgroup pickupgroup
+; language language
+; allow allow
+; disallow disallow
+; insecure insecure
+; trustrpid trustrpid
+; progressinband progressinband
+; promiscredir promiscredir
+; useclientcode useclientcode
+; accountcode accountcode
+; setvar setvar
+; callerid callerid
+; amaflags amaflags
+; call-limit call-limit
+; allowoverlap allowoverlap
+; allowsubscribe allowsubscribe
+; allowtransfer allowtransfer
+; subscribecontext subscribecontext
+; videosupport videosupport
+; maxcallbitrate maxcallbitrate
+; rfc2833compensate mailbox
+; username
+; template
+; fromdomain
+; regexten
+; fromuser
+; host
+; port
+; qualify
+; defaultip
+; rtptimeout
+; rtpholdtimeout
+; sendrpid
+; outboundproxy
+; rfc2833compensate
+
+;[sip_proxy]
+; For incoming calls only. Example: FWD (Free World Dialup)
+; 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
+;call-limit=5 ; permit only 5 simultaneous outgoing calls to this peer
+;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer
+ ; Call-limits will not be enforced on real-time peers,
+ ; since they are not stored in-memory
+;port=80 ; The port number we want to connect to on the remote side
+ ; Also used as "defaultport" in combination with "defaultip" settings
+
+;------------------------------------------------------------------------------
+; Definitions of locally connected SIP devices
+;
+; type = user a device that authenticates to us by "from" field to place calls
+; type = peer a device we place calls to or that calls us and we match by host
+; type = friend two configurations (peer+user) in one
+;
+; For device names, we recommend using only a-z, numerics (0-9) and underscore
+;
+; For local phones, type=friend works most of the time
+;
+; If you have one-way audio, you probably 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
+;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
+ ; on incoming calls to Asterisk
+;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
+;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time
+ ; from the phone to asterisk
+ ; 1 for the explicit peer, 1 for the explicit user,
+ ; remember that a friend equals 1 peer and 1 user in
+ ; memory
+ ; This will affect your subscriptions as well.
+ ; There is no combined call counter for a "friend"
+ ; so there's currently no way in sip.conf to limit
+ ; to one inbound or outbound call per phone. Use
+ ; the group counters in the dial plan for that.
+ ;
+;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
+;callingpres=allowed_passed_screen ; Set caller ID presentation
+ ; See README.callingpres for more information
+
+
+;[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
+;regexten=1234 ; When they register, create extension 1234
+;callerid="Jane Smith" <5678>
+;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=ulaw
+;allow=alaw
+;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes
+
+
+;[snom]
+;type=friend ; Friends place calls and receive calls
+;context=from-sip ; Context for incoming calls from this user
+;secret=blah
+;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions
+;language=de ; Use German prompts for this user
+;host=dynamic ; This peer register with us
+;dtmfmode=inband ; Choices are inband, rfc2833, or info
+;defaultip=192.168.0.59 ; IP used until peer registers
+;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator
+;subscribemwi=yes ; Only send notifications if this phone
+ ; subscribes for mailbox notification
+;vmexten=voicemail ; dialplan extension to reach mailbox
+ ; sets the Message-Account in the MWI notify message
+ ; defaults to global vmexten which defaults to "asterisk"
+;disallow=all
+;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
+
+
+;[polycom]
+;type=friend ; Friends place calls and receive calls
+;context=from-sip ; Context for incoming calls from this user
+;secret=blahpoly
+;host=dynamic ; This peer register with us
+;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
+;username=polly ; Username to use in INVITE until peer registers
+ ; Normally you do NOT need to set this parameter
+;disallow=all
+;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
+;progressinband=no ; Polycom phones don't work properly with "never"
+
+
+;[pingtel]
+;type=friend
+;secret=blah
+;host=dynamic
+;insecure=port ; Allow matching of peer by IP address without
+ ; matching port number
+;insecure=invite ; Do not require authentication of incoming INVITEs
+;insecure=port,invite ; (both)
+;qualify=1000 ; Consider it down if it's 1 second to reply
+ ; Helps with NAT session
+ ; qualify=yes uses default value
+;
+; Call group and Pickup group should be in the range from 0 to 63
+;
+;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 registered
+;deny=0.0.0.0/0.0.0.0 ; ACL: Control access to this account based on IP address
+;permit=192.168.0.60/255.255.255.0
+
+;[cisco1]
+;type=friend
+;secret=blah
+;qualify=200 ; Qualify peer is no more than 200ms away
+;nat=yes ; This phone may be natted
+ ; 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
+ ; 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).
+;defaultip=192.168.0.4 ; IP address to use until registration
+;username=goran ; Username to use when calling this device before registration
+ ; Normally you do NOT need to set this parameter
+;setvar=CUSTID=5678 ; Channel variable to be set for all calls from this device
+
+;[pre14-asterisk]
+;type=friend
+;secret=digium
+;host=dynamic
+;rfc2833compensate=yes ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
+ ; You must have this turned on or DTMF reception will work improperly.
+
+[6010]
+type=friend
+context=default
+host=dynamic
+secret=6010
+canreinvite=no
+callerid=6010
+disallow=all
+allow=ulaw,alaw
+qualify=yes
+
+[6011]
+type=friend
+context=default
+host=dynamic
+secret=6011
+canreinvite=no
+callerid=6011
+disallow=all
+allow=ulaw,alaw
+qualify=yes
+
+[6012]
+type=friend
+context=default
+host=dynamic
+secret=6012
+canreinvite=no
+callerid=6012
+disallow=all
+allow=ulaw,alaw
+qualify=yes
+
+[6013]
+type=friend
+context=default
+host=dynamic
+secret=6013
+canreinvite=no
+callerid=6013
+disallow=all
+allow=ulaw,alaw
+qualify=yes
+
+[6014]
+type=friend
+context=default
+host=dynamic
+secret=6014
+canreinvite=no
+callerid=6014
+disallow=all
+allow=ulaw,alaw
+qualify=yes
+
+[6015]
+type=friend
+context=default
+host=dynamic
+secret=6015
+canreinvite=no
+callerid=6015
+disallow=all
+allow=ulaw,alaw
+qualify=yes
+
+[6016]
+type=friend
+context=default
+host=dynamic
+secret=6016
+canreinvite=no
+callerid=6016
+disallow=all
+allow=ulaw,alaw
+qualify=yes
+
+[6017]
+type=friend
+context=default
+host=dynamic
+secret=6017
+canreinvite=no
+callerid=6017
+disallow=all
+allow=ulaw,alaw
+qualify=yes
+
+[6018]
+type=friend
+context=default
+host=dynamic
+secret=6018
+canreinvite=no
+callerid=6018
+disallow=all
+allow=ulaw,alaw
+qualify=yes
+
+[6019]
+type=friend
+context=default
+host=dynamic
+secret=6019
+canreinvite=no
+callerid=6019
+disallow=all
+allow=ulaw,alaw
+qualify=yes
+
+[6020]
+type=friend
+context=default
+host=dynamic
+secret=6020
+canreinvite=no
+callerid=6020
+disallow=all
+allow=ulaw,alaw
+qualify=yes
+
--- /dev/null
+;
+; Zapata telephony interface
+;
+; Configuration file
+;
+; You need to restart Asterisk to re-configure the Zap channel
+; CLI> reload chan_zap.so
+; will reload the configuration file,
+; but not all configuration options are
+; re-configured during a reload.
+
+
+
+[trunkgroups]
+;
+; Trunk groups are used for NFAS or GR-303 connections.
+;
+; Group: Defines a trunk group.
+; trunkgroup => <trunkgroup>,<dchannel>[,<backup1>...]
+;
+; trunkgroup is the numerical trunk group to create
+; dchannel is the zap channel which will have the
+; d-channel for the trunk.
+; backup1 is an optional list of backup d-channels.
+;
+;trunkgroup => 1,24,48
+;trunkgroup => 1,24
+;
+; Spanmap: Associates a span with a trunk group
+; spanmap => <zapspan>,<trunkgroup>[,<logicalspan>]
+;
+; zapspan is the zap span number to associate
+; trunkgroup is the trunkgroup (specified above) for the mapping
+; logicalspan is the logical span number within the trunk group to use.
+; if unspecified, no logical span number is used.
+;
+;spanmap => 1,1,1
+;spanmap => 2,1,2
+;spanmap => 3,1,3
+;spanmap => 4,1,4
+
+[channels]
+;
+; Default language
+;
+;language=en
+;
+; Default context
+;
+context=default
+;
+; Switchtype: Only used for PRI.
+;
+; national: National ISDN 2 (default)
+; dms100: Nortel DMS100
+; 4ess: AT&T 4ESS
+; 5ess: Lucent 5ESS
+; euroisdn: EuroISDN
+; ni1: Old National ISDN 1
+; qsig: Q.SIG
+;
+switchtype=national
+;
+; Some switches (AT&T especially) require network specific facility IE
+; supported values are currently 'none', 'sdn', 'megacom', 'tollfreemegacom', 'accunet'
+;
+;nsf=none
+;
+; PRI Dialplan: Only RARELY used for PRI.
+;
+; unknown: Unknown
+; private: Private ISDN
+; local: Local ISDN
+; national: National ISDN
+; international: International ISDN
+; dynamic: Dynamically selects the appropriate dialplan
+;
+;pridialplan=national
+;
+; PRI Local Dialplan: Only RARELY used for PRI (sets the calling number's numbering plan)
+;
+; unknown: Unknown
+; private: Private ISDN
+; local: Local ISDN
+; national: National ISDN
+; international: International ISDN
+; dynamic: Dynamically selects the appropriate dialplan
+;
+;prilocaldialplan=national
+;
+; PRI callerid prefixes based on the given TON/NPI (dialplan)
+; This is especially needed for euroisdn E1-PRIs
+;
+; sample 1 for Germany
+;internationalprefix = 00
+;nationalprefix = 0
+;localprefix = 0711
+;privateprefix = 07115678
+;unknownprefix =
+;
+; sample 2 for Germany
+;internationalprefix = +
+;nationalprefix = +49
+;localprefix = +49711
+;privateprefix = +497115678
+;unknownprefix =
+;
+; PRI resetinterval: sets the time in seconds between restart of unused
+; channels, defaults to 3600; minimum 60 seconds. Some PBXs don't like
+; channel restarts. so set the interval to a very long interval e.g. 100000000
+; or 'never' to disable *entirely*.
+;
+;resetinterval = 3600
+;
+; Overlap dialing mode (sending overlap digits)
+;
+;overlapdial=yes
+;
+; PRI Out of band indications.
+; Enable this to report Busy and Congestion on a PRI using out-of-band
+; notification. Inband indication, as used by Asterisk doesn't seem to work
+; with all telcos.
+;
+; outofband: Signal Busy/Congestion out of band with RELEASE/DISCONNECT
+; inband: Signal Busy/Congestion using in-band tones
+;
+; priindication = outofband
+;
+; If you need to override the existing channels selection routine and force all
+; PRI channels to be marked as exclusively selected, set this to yes.
+; priexclusive = yes
+;
+; ISDN Timers
+; All of the ISDN timers and counters that are used are configurable. Specify
+; the timer name, and its value (in ms for timers).
+; K: Layer 2 max number of outstanding unacknowledged I frames (default 7)
+; N200: Layer 2 max number of retransmissions of a frame (default 3)
+; T200: Layer 2 max time before retransmission of a frame (default 1000 ms)
+; T203: Layer 2 max time without frames being exchanged (default 10000 ms)
+; T305: Wait for DISCONNECT acknowledge (default 30000 ms)
+; T308: Wait for RELEASE acknowledge (default 4000 ms)
+; T309: Maintain active calls on Layer 2 disconnection (default -1, Asterisk clears calls)
+; EuroISDN: 6000 to 12000 ms, according to (N200 + 1) x T200 + 2s
+; May vary in other ISDN standards (Q.931 1993 : 90000 ms)
+; T313: Wait for CONNECT acknowledge, CPE side only (default 3000 ms)
+;
+; pritimer => t200,1000
+; pritimer => t313,4000
+;
+; To enable transmission of facility-based ISDN supplementary services (such
+; as caller name from CPE over facility), enable this option.
+; facilityenable = yes
+;
+;
+; Signalling method (default is fxs). Valid values:
+; em: E & M
+; em_w: E & M Wink
+; featd: Feature Group D (The fake, Adtran style, DTMF)
+; featdmf: Feature Group D (The real thing, MF (domestic, US))
+; featdmf_ta: Feature Group D (The real thing, MF (domestic, US)) through
+; a Tandem Access point
+; featb: Feature Group B (MF (domestic, US))
+; fgccama Feature Group C-CAMA (DP DNIS, MF ANI)
+; fgccamamf Feature Group C-CAMA MF (MF DNIS, MF ANI)
+; fxs_ls: FXS (Loop Start)
+; fxs_gs: FXS (Ground Start)
+; fxs_ks: FXS (Kewl Start)
+; fxo_ls: FXO (Loop Start)
+; fxo_gs: FXO (Ground Start)
+; fxo_ks: FXO (Kewl Start)
+; pri_cpe: PRI signalling, CPE side
+; pri_net: PRI signalling, Network side
+; gr303fxoks_net: GR-303 Signalling, FXO Loopstart, Network side
+; gr303fxsks_cpe: GR-303 Signalling, FXS Loopstart, CPE side
+; sf: SF (Inband Tone) Signalling
+; sf_w: SF Wink
+; sf_featd: SF Feature Group D (The fake, Adtran style, DTMF)
+; sf_featdmf: SF Feature Group D (The real thing, MF (domestic, US))
+; sf_featb: SF Feature Group B (MF (domestic, US))
+; e911: E911 (MF) style signalling
+;
+; The following are used for Radio interfaces:
+; fxs_rx: Receive audio/COR on an FXS kewlstart interface (FXO at the
+; channel bank)
+; fxs_tx: Transmit audio/PTT on an FXS loopstart interface (FXO at the
+; channel bank)
+; fxo_rx: Receive audio/COR on an FXO loopstart interface (FXS at the
+; channel bank)
+; fxo_tx: Transmit audio/PTT on an FXO groundstart interface (FXS at
+; the channel bank)
+; em_rx: Receive audio/COR on an E&M interface (1-way)
+; em_tx: Transmit audio/PTT on an E&M interface (1-way)
+; em_txrx: Receive audio/COR AND Transmit audio/PTT on an E&M interface
+; (2-way)
+; em_rxtx: Same as em_txrx (for our dyslexic friends)
+; sf_rx: Receive audio/COR on an SF interface (1-way)
+; sf_tx: Transmit audio/PTT on an SF interface (1-way)
+; sf_txrx: Receive audio/COR AND Transmit audio/PTT on an SF interface
+; (2-way)
+; sf_rxtx: Same as sf_txrx (for our dyslexic friends)
+;
+signalling=fxo_ls
+;
+; If you have an outbound signalling format that is different from format
+; specified above (but compatible), you can specify outbound signalling format,
+; (see below). The 'signalling' format specified will be the inbound signalling
+; format. If you only specify 'signalling', then it will be the format for
+; both inbound and outbound.
+;
+; signalling=featdmf
+; outsignalling=featb
+;
+; For Feature Group D Tandem access, to set the default CIC and OZZ use these
+; parameters:
+;defaultozz=0000
+;defaultcic=303
+;
+; A variety of timing parameters can be specified as well
+; Including:
+; prewink: Pre-wink time (default 50ms)
+; preflash: Pre-flash time (default 50ms)
+; wink: Wink time (default 150ms)
+; flash: Flash time (default 750ms)
+; start: Start time (default 1500ms)
+; rxwink: Receiver wink time (default 300ms)
+; rxflash: Receiver flashtime (default 1250ms)
+; debounce: Debounce timing (default 600ms)
+;
+rxwink=300 ; Atlas seems to use long (250ms) winks
+;
+; How long generated tones (DTMF and MF) will be played on the channel
+; (in milliseconds)
+;toneduration=100
+;
+; Whether or not to do distinctive ring detection on FXO lines
+;
+;usedistinctiveringdetection=yes
+;distinctiveringaftercid=yes ; enable dring detection after callerid for those countries like Australia
+ ; where the ring cadence is changed *after* the callerid spill.
+;
+; Whether or not to use caller ID
+;
+usecallerid=yes
+;
+; Type of caller ID signalling in use
+; bell = bell202 as used in US
+; v23 = v23 as used in the UK
+; v23_jp = v23 as used in Japan
+; dtmf = DTMF as used in Denmark, Sweden and Netherlands
+; smdi = Use SMDI for callerid. Requires SMDI to be enabled (usesmdi).
+;
+;cidsignalling=bell
+;
+; What signals the start of caller ID
+; ring = a ring signals the start
+; polarity = polarity reversal signals the start
+;
+;cidstart=ring
+;
+; Whether or not to hide outgoing caller ID (Override with *67 or *82)
+;
+hidecallerid=no
+;
+; Whether or not to enable call waiting on internal extensions
+; With this set to 'yes', busy extensions will hear the call-waiting
+; tone, and can use hook-flash to switch between callers. The Dial()
+; app will not return the "BUSY" result for extensions.
+;
+callwaiting=yes
+;
+; Whether or not restrict outgoing caller ID (will be sent as ANI only, not
+; available for the user)
+; Mostly use with FXS ports
+;
+;restrictcid=no
+;
+; Whether or not use the caller ID presentation for the outgoing call that the
+; calling switch is sending.
+; See README.callingpres
+;
+usecallingpres=yes
+;
+; Some countries (UK) have ring tones with different ring tones (ring-ring),
+; which means the callerid needs to be set later on, and not just after
+; the first ring, as per the default.
+;
+;sendcalleridafter=1
+;
+;
+; Support Caller*ID on Call Waiting
+;
+callwaitingcallerid=yes
+;
+; Support three-way calling
+;
+threewaycalling=yes
+;
+; Support flash-hook call transfer (requires three way calling)
+; Also enables call parking (overrides the 'canpark' parameter)
+;
+transfer=yes
+;
+; Allow call parking
+; ('canpark=no' is overridden by 'transfer=yes')
+;
+canpark=yes
+;
+; Support call forward variable
+;
+cancallforward=yes
+;
+; Whether or not to support Call Return (*69)
+;
+callreturn=yes
+;
+; Stutter dialtone support: If a mailbox is specified without a voicemail
+; context, then when voicemail is received in a mailbox in the default
+; voicemail context in voicemail.conf, taking the phone off hook will cause a
+; stutter dialtone instead of a normal one.
+;
+; If a mailbox is specified *with* a voicemail context, the same will result
+; if voicemail received in mailbox in the specified voicemail context.
+;
+; for default voicemail context, the example below is fine:
+;
+;mailbox=1234
+;
+; for any other voicemail context, the following will produce the stutter tone:
+;
+;mailbox=1234@context
+;
+; Enable echo cancellation
+; Use either "yes", "no", or a power of two from 32 to 256 if you wish to
+; actually set the number of taps of cancellation.
+;
+; Note that if any of your Zaptel cards have hardware echo cancellers,
+; then this setting only turns them on and off; numeric settings will
+; be treated as "yes". There are no special settings required for
+; hardware echo cancellers; when present and enabled in their kernel
+; modules, they take precedence over the software echo canceller compiled
+; into Zaptel automatically.
+;
+echocancel=yes
+;
+; Generally, it is not necessary (and in fact undesirable) to echo cancel when
+; the circuit path is entirely TDM. You may, however, change this behavior
+; by enabling the echo cancel during pure TDM bridging below.
+;
+echocancelwhenbridged=yes
+;
+; In some cases, the echo canceller doesn't train quickly enough and there
+; is echo at the beginning of the call. Enabling echo training will cause
+; asterisk to briefly mute the channel, send an impulse, and use the impulse
+; response to pre-train the echo canceller so it can start out with a much
+; closer idea of the actual echo. Value may be "yes", "no", or a number of
+; milliseconds to delay before training (default = 400)
+;
+; WARNING: In some cases this option can make echo worse! If you are
+; trying to debug an echo problem, it is worth checking to see if your echo
+; is better with the option set to yes or no. Use whatever setting gives
+; the best results.
+;
+; Note that these parameters do not apply to hardware echo cancellers.
+;
+;echotraining=yes
+;echotraining=800
+;
+; If you are having trouble with DTMF detection, you can relax the DTMF
+; detection parameters. Relaxing them may make the DTMF detector more likely
+; to have "talkoff" where DTMF is detected when it shouldn't be.
+;
+;relaxdtmf=yes
+;
+; You may also set the default receive and transmit gains (in dB)
+;
+rxgain=0.0
+txgain=0.0
+;
+; Logical groups can be assigned to allow outgoing rollover. Groups range
+; from 0 to 63, and multiple groups can be specified.
+;
+group=1
+;
+; Ring groups (a.k.a. call groups) and pickup groups. If a phone is ringing
+; and it is a member of a group which is one of your pickup groups, then
+; you can answer it by picking up and dialling *8#. For simple offices, just
+; make these both the same. Groups range from 0 to 63.
+;
+callgroup=1
+pickupgroup=1
+
+;
+; Specify whether the channel should be answered immediately or if the simple
+; switch should provide dialtone, read digits, etc.
+;
+immediate=no
+;
+; Specify whether flash-hook transfers to 'busy' channels should complete or
+; return to the caller performing the transfer (default is yes).
+;
+;transfertobusy=no
+;
+; CallerID can be set to "asreceived" or a specific number if you want to
+; override it. Note that "asreceived" only applies to trunk interfaces.
+;
+;callerid=2564286000
+;
+; AMA flags affects the recording of Call Detail Records. If specified
+; it may be 'default', 'omit', 'billing', or 'documentation'.
+;
+;amaflags=default
+;
+; Channels may be associated with an account code to ease
+; billing
+;
+;accountcode=lss0101
+;
+; ADSI (Analog Display Services Interface) can be enabled on a per-channel
+; basis if you have (or may have) ADSI compatible CPE equipment
+;
+;adsi=yes
+;
+; SMDI (Simplified Message Desk Interface) can be enabled on a per-channel
+; basis if you would like that channel to behave like an SMDI message desk.
+; The SMDI port specified should have already been defined in smdi.conf. The
+; default port is /dev/ttyS0.
+;
+;usesmdi=yes
+;smdiport=/dev/ttyS0
+;
+; On trunk interfaces (FXS) and E&M interfaces (E&M, Wink, Feature Group D
+; etc, it can be useful to perform busy detection either in an effort to
+; detect hangup or for detecting busies. This enables listening for
+; the beep-beep busy pattern.
+;
+;busydetect=yes
+;
+; If busydetect is enabled, it is also possible to specify how many busy tones
+; to wait for before hanging up. The default is 4, but better results can be
+; achieved if set to 6 or even 8. Mind that the higher the number, the more
+; time that will be needed to hangup a channel, but lowers the probability
+; that you will get random hangups.
+;
+;busycount=4
+;
+; If busydetect is enabled, it is also possible to specify the cadence of your
+; busy signal. In many countries, it is 500msec on, 500msec off. Without
+; busypattern specified, we'll accept any regular sound-silence pattern that
+; repeats <busycount> times as a busy signal. If you specify busypattern,
+; then we'll further check the length of the sound (tone) and silence, which
+; will further reduce the chance of a false positive.
+;
+;busypattern=500,500
+;
+; NOTE: In the Asterisk Makefile you'll find further options to tweak the busy
+; detector. If your country has a busy tone with the same length tone and
+; silence (as many countries do), consider defining the
+; -DBUSYDETECT_COMPARE_TONE_AND_SILENCE option.
+;
+; Use a polarity reversal to mark when a outgoing call is answered by the
+; remote party.
+;
+;answeronpolarityswitch=yes
+;
+; In some countries, a polarity reversal is used to signal the disconnect of a
+; phone line. If the hanguponpolarityswitch option is selected, the call will
+; be considered "hung up" on a polarity reversal.
+;
+;hanguponpolarityswitch=yes
+;
+; On trunk interfaces (FXS) it can be useful to attempt to follow the progress
+; of a call through RINGING, BUSY, and ANSWERING. If turned on, call
+; progress attempts to determine answer, busy, and ringing on phone lines.
+; This feature is HIGHLY EXPERIMENTAL and can easily detect false answers,
+; so don't count on it being very accurate.
+;
+; Few zones are supported at the time of this writing, but may be selected
+; with "progzone"
+;
+; This feature can also easily detect false hangups. The symptoms of this is
+; being disconnected in the middle of a call for no reason.
+;
+;callprogress=yes
+;progzone=us
+;
+; FXO (FXS signalled) devices must have a timeout to determine if there was a
+; hangup before the line was answered. This value can be tweaked to shorten
+; how long it takes before Zap considers a non-ringing line to have hungup.
+;
+;ringtimeout=8000
+;
+; For FXO (FXS signalled) devices, whether to use pulse dial instead of DTMF
+;
+;pulsedial=yes
+;
+; For fax detection, uncomment one of the following lines. The default is *OFF*
+;
+;faxdetect=both
+;faxdetect=incoming
+;faxdetect=outgoing
+;faxdetect=no
+;
+; This option specifies a preference for which music on hold class this channel
+; should listen to when put on hold if the music class has not been set on the
+; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
+; channel putting this one on hold did not suggest a music class.
+;
+; If this option is set to "passthrough", then the hold message will always be
+; passed through as signalling instead of generating hold music locally. This
+; setting is only valid when used on a channel that uses digital signalling.
+;
+; This option may be specified globally, or on a per-user or per-peer basis.
+;
+;mohinterpret=default
+;
+; This option specifies which music on hold class to suggest to the peer channel
+; when this channel places the peer on hold. It may be specified globally or on
+; a per-user or per-peer basis.
+;
+;mohsuggest=default
+;
+; PRI channels can have an idle extension and a minunused number. So long as
+; at least "minunused" channels are idle, chan_zap will try to call "idledial"
+; on them, and then dump them into the PBX in the "idleext" extension (which
+; is of the form exten@context). When channels are needed the "idle" calls
+; are disconnected (so long as there are at least "minidle" calls still
+; running, of course) to make more channels available. The primary use of
+; this is to create a dynamic service, where idle channels are bundled through
+; multilink PPP, thus more efficiently utilizing combined voice/data services
+; than conventional fixed mappings/muxings.
+;
+;idledial=6999
+;idleext=6999@dialout
+;minunused=2
+;minidle=1
+;
+; Configure jitter buffers in zapata (each one is 20ms, default is 4)
+;
+;jitterbuffers=4
+;
+;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
+; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
+ ; ZAP channel. Defaults to "no". An enabled jitterbuffer will
+ ; be used only if the sending side can create and the receiving
+ ; side can not accept jitter. The ZAP channel can't accept jitter,
+ ; thus an enabled jitterbuffer on the receive ZAP side will always
+ ; be used if the sending side can create jitter.
+
+; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
+
+; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
+ ; resynchronized. Useful to improve the quality of the voice, with
+ ; big jumps in/broken timestamps, usually sent from exotic devices
+ ; and programs. Defaults to 1000.
+
+; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a ZAP
+ ; channel. Two implementations are currently available - "fixed"
+ ; (with size always equals to jbmax-size) and "adaptive" (with
+ ; variable size, actually the new jb of IAX2). Defaults to fixed.
+
+; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
+;-----------------------------------------------------------------------------------
+;
+; You can define your own custom ring cadences here. You can define up to 8
+; pairs. If the silence is negative, it indicates where the callerid spill is
+; to be placed. Also, if you define any custom cadences, the default cadences
+; will be turned off.
+;
+; Syntax is: cadence=ring,silence[,ring,silence[...]]
+;
+; These are the default cadences:
+;
+;cadence=125,125,2000,-4000
+;cadence=250,250,500,1000,250,250,500,-4000
+;cadence=125,125,125,125,125,-4000
+;cadence=1000,500,2500,-5000
+;
+; Each channel consists of the channel number or range. It inherits the
+; parameters that were specified above its declaration.
+;
+; For GR-303, CRV's are created like channels except they must start with the
+; trunk group followed by a colon, e.g.:
+;
+; crv => 1:1
+; crv => 2:1-2,5-8
+;
+;
+;callerid="Green Phone"<(256) 428-6121>
+;channel => 1
+;callerid="Black Phone"<(256) 428-6122>
+;channel => 2
+;callerid="CallerID Phone" <(256) 428-6123>
+;callerid="CallerID Phone" <(630) 372-1564>
+;callerid="CallerID Phone" <(256) 704-4666>
+;channel => 3
+;callerid="Pac Tel Phone" <(256) 428-6124>
+;channel => 4
+;callerid="Uniden Dead" <(256) 428-6125>
+;channel => 5
+;callerid="Cortelco 2500" <(256) 428-6126>
+;channel => 6
+;callerid="Main TA 750" <(256) 428-6127>
+;channel => 44
+;
+; For example, maybe we have some other channels which start out in a
+; different context and use E & M signalling instead.
+;
+;context=remote
+;sigalling=em
+;channel => 15
+;channel => 16
+
+;signalling=em_w
+;
+; All those in group 0 I'll use for outgoing calls
+;
+; Strip most significant digit (9) before sending
+;
+;stripmsd=1
+;callerid=asreceived
+;group=0
+;signalling=fxs_ls
+;channel => 45
+
+;signalling=fxo_ls
+;group=1
+;callerid="Joe Schmoe" <(256) 428-6131>
+;channel => 25
+;callerid="Megan May" <(256) 428-6132>
+;channel => 26
+;callerid="Suzy Queue" <(256) 428-6233>
+;channel => 27
+;callerid="Larry Moe" <(256) 428-6234>
+;channel => 28
+;
+; Sample PRI (CPE) config: Specify the switchtype, the signalling as either
+; pri_cpe or pri_net for CPE or Network termination, and generally you will
+; want to create a single "group" for all channels of the PRI.
+;
+; switchtype = national
+; signalling = pri_cpe
+; group = 2
+; channel => 1-23
+
+;
+
+; Used for distinctive ring support for x100p.
+; You can see the dringX patterns is to set any one of the dringXcontext fields
+; and they will be printed on the console when an inbound call comes in.
+;
+;dring1=95,0,0
+;dring1context=internal1
+;dring2=325,95,0
+;dring2context=internal2
+; If no pattern is matched here is where we go.
+;context=default
+;channel => 1
+
--- /dev/null
+# /usr/share/doc/lighttpd-doc/authentication.txt.gz
+
+server.modules += ( "mod_auth" )
+
+# auth.backend = "plain"
+# auth.backend.plain.userfile = "lighttpd.user"
+# auth.backend.plain.groupfile = "lighttpd.group"
+
+# auth.backend.ldap.hostname = "localhost"
+# auth.backend.ldap.base-dn = "dc=my-domain,dc=com"
+# auth.backend.ldap.filter = "(uid=$)"
+
+# auth.require = ( "/server-status" =>
+# (
+# "method" => "digest",
+# "realm" => "download archiv",
+# "require" => "group=www|user=jan|host=192.168.2.10"
+# ),
+# "/server-info" =>
+# (
+# "method" => "digest",
+# "realm" => "download archiv",
+# "require" => "group=www|user=jan|host=192.168.2.10"
+# )
+# )
--- /dev/null
+server.modules += ( "mod_accesslog" )
+
+accesslog.filename = "/var/log/lighttpd/access.log"
--- /dev/null
+# /usr/share/doc/lighttpd-doc/cgi.txt
+
+server.modules += ( "mod_cgi" )
+
+$HTTP["url"] =~ "^/cgi-bin/" {
+ cgi.assign = ( "" => "" )
+}
+
+## Warning this represents a security risk, as it allow to execute any file
+## with a .pl/.py even outside of /usr/lib/cgi-bin.
+#
+cgi.assign = (
+ ".pl" => "/usr/bin/perl",
+ ".sh" => "/bin/sh",
+)
--- /dev/null
+server.modules += ( "mod_evasive" )
--- /dev/null
+# http://redmine.lighttpd.net/wiki/1/Docs:ModEVhost
+
+server.modules += ( "mod_evhost" )
+
+evhost.path-pattern = "/srv/%_/htdocs"
--- /dev/null
+# http://redmine.lighttpd.net/projects/lighttpd/wiki/Docs:ModExpire
+
+server.modules += ( "mod_expire" )
--- /dev/null
+# /usr/share/doc/lighttpd-doc/fastcgi.txt.gz
+# http://redmine.lighttpd.net/projects/lighttpd/wiki/Docs:ConfigurationOptions#mod_fastcgi-fastcgi
+
+server.modules += ( "mod_fastcgi" )
--- /dev/null
+server.modules += ( "mod_flv_streaming" )
--- /dev/null
+$HTTP["host"] =~ "^www\.(.*)" {
+ url.redirect = ( "^/(.*)" => "http://%1/$1" )
+}
--- /dev/null
+# /usr/share/doc/lighttpd-doc/proxy.txt
+
+server.modules += ( "mod_proxy" )
+
+## Balance algorithm, possible values are: "hash", "round-robin" or "fair" (default)
+# proxy.balance = "hash"
+
+
+## Redirect all queries to files ending with ".php" to 192.168.0.101:80
+#proxy.server = ( ".php" =>
+# (
+# ( "host" => "192.168.0.101",
+# "port" => 80
+# )
+# )
+# )
+
+## Redirect all connections on www.example.com to 10.0.0.1{0,1,2,3}
+#$HTTP["host"] == "www.example.com" {
+# proxy.balance = "hash"
+# proxy.server = ( "" => ( ( "host" => "10.0.0.10" ),
+# ( "host" => "10.0.0.11" ),
+# ( "host" => "10.0.0.12" ),
+# ( "host" => "10.0.0.13" ) ) )
+#}
--- /dev/null
+# /usr/share/doc/lighttpd-doc/rrdtool.txt
+
+server.modules += ( "mod_rrdtool" )
+
+## path to the rrdtool binary
+rrdtool.binary = "/usr/bin/rrdtool"
+
+## file to store the rrd database, will be created by lighttpd
+rrdtool.db-name = "/var/www/lighttpd.rrd"
+
--- /dev/null
+# /usr/share/doc/lighttpd-doc/simple-vhost.txt
+
+server.modules += ( "mod_simple_vhost" )
+
+## The document root of a virtual host is document-root =
+## simple-vhost.server-root + $HTTP["host"] + simple-vhost.document-root
+simple-vhost.server-root = "/srv"
+simple-vhost.document-root = "htdocs"
+
+## the default host if no host is sent
+simple-vhost.default-host = "www.example.com"
--- /dev/null
+# /usr/share/doc/lighttpd-doc/ssi.txt
+
+server.modules += ( "mod_ssi" )
+
+ssi.extension = ( ".shtml" )
--- /dev/null
+# /usr/share/doc/lighttpd-doc/ssl.txt
+
+$SERVER["socket"] == "0.0.0.0:443" {
+ ssl.engine = "enable"
+ ssl.pemfile = "/etc/lighttpd/server.pem"
+}
--- /dev/null
+# /usr/share/doc/lighttpd-doc/status.txt
+# http://trac.lighttpd.net/trac/wiki/Docs%3AModStatus
+
+server.modules += ( "mod_status" )
+
+# status.status-url = "/server-status"
+
+# status.config-url = "/server-config"
+
+## relative URL for a plain-text page containing the internal statistics
+# status.statistics-url = "/server-statistics"
+
+## add JavaScript which allows client-side sorting for the connection overview
+## default: enable
+# status.enable-sort = "disable"
--- /dev/null
+## The userdir module provides a simple way to link user-based directories into
+## the global namespace of the webserver.
+##
+# /usr/share/doc/lighttpd-doc/userdir.txt
+
+server.modules += ( "mod_userdir" )
+
+## the subdirectory of a user's home dir which should be accessible
+## under http://$host/~$user
+userdir.path = "public_html"
+
+## The users whose home directories should not be accessible
+userdir.exclude-user = ( "root", "postmaster" )
--- /dev/null
+server.modules += ( "mod_usertrack" )
--- /dev/null
+# /usr/share/doc/lighttpd-doc/fastcgi.txt.gz
+# http://redmine.lighttpd.net/projects/lighttpd/wiki/Docs:ConfigurationOptions#mod_fastcgi-fastcgi
+
+## Start an FastCGI server for php (needs the php5-cgi package)
+fastcgi.server += ( ".php" =>
+ ((
+ "bin-path" => "/usr/bin/php-cgi",
+ "socket" => "/tmp/php.socket",
+ "max-procs" => 1,
+ "bin-environment" => (
+ "PHP_FCGI_CHILDREN" => "4",
+ "PHP_FCGI_MAX_REQUESTS" => "10000"
+ ),
+ "bin-copy-environment" => (
+ "PATH", "SHELL", "USER"
+ ),
+ "broken-scriptfilename" => "enable"
+ ))
+)
--- /dev/null
+#### handle Debian Policy Manual, Section 11.5. urls
+## by default allow them only from localhost
+$HTTP["remoteip"] =~ "^127\.0\.0\.1$|^::1$" {
+ alias.url += (
+ "/cgi-bin/" => "/usr/lib/cgi-bin/",
+ "/doc/" => "/usr/share/doc/",
+ "/images/" => "/usr/share/images/"
+ )
+ $HTTP["url"] =~ "^/doc/|^/images/" {
+ dir-listing.activate = "enable"
+ }
+ $HTTP["url"] =~ "^/cgi-bin/" {
+ cgi.assign = ( "" => "" )
+ }
+}
--- /dev/null
+ligghttpd Configuration under Debian GNU/Linux
+==============================================
+
+Files and Directories in /etc/lighttpd:
+---------------------------------------
+
+lighttpd.conf:
+ main configuration file
+
+conf-available/
+ This directory contains a series of .conf files. These files contain
+ configuration directives necessary to load and run webserver modules.
+ If you want to create your own files they names should be
+ build as nn-name.conf where "nn" is two digit number (number
+ is used to find order for loading files)
+
+conf-enabled/
+ To actually enable a module for lighttpd, it is necessary to create a
+ symlink in this directory to the .conf file in conf-available/.
+
+Enabling and disabling modules could be done by provided
+/usr/sbin/lighty-enable-mod and /usr/sbin/lighty-disable-mod scripts.
--- /dev/null
+../conf-available/10-cgi.conf
\ No newline at end of file
--- /dev/null
+server.modules = (
+ "mod_access",
+ "mod_alias",
+ "mod_compress",
+ "mod_redirect",
+# "mod_rewrite",
+)
+
+server.document-root = "/home/ian/mini-asterisk-gui2"
+server.upload-dirs = ( "/var/cache/lighttpd/uploads" )
+server.errorlog = "/var/log/lighttpd/error.log"
+server.pid-file = "/var/run/lighttpd.pid"
+server.username = "www-data"
+server.groupname = "www-data"
+#server.username = "wwwrun"
+#server.groupname = "wwwrun"
+
+index-file.names = ( "index.php", "index.html",
+ "index.htm", "default.htm",
+ " index.lighttpd.html" )
+
+url.access-deny = ( "~", ".inc" )
+
+static-file.exclude-extensions = ( ".php", ".pl", ".fcgi" )
+
+## Use ipv6 if available
+#include_shell "/usr/share/lighttpd/use-ipv6.pl"
+
+dir-listing.encoding = "utf-8"
+server.dir-listing = "enable"
+
+compress.cache-dir = "/var/cache/lighttpd/compress/"
+compress.filetype = ( "application/x-javascript", "text/css", "text/html", "text/plain" )
+
+include_shell "/usr/share/lighttpd/create-mime.assign.pl"
+include_shell "/usr/share/lighttpd/include-conf-enabled.pl"
--- /dev/null
+loadzone = us
+defaultzone=us
+
+fxoks=2,3,4
// kick off cascading CGIs, initialisePage() will be called when complete
loadExtensions();
+ //initialisePage();
}
// fire off Ifconfig IP address to get IP Address
-// downloadUrl("/cgi-bin/ifconfig.cgi?arg=eth0", processIfconfig);
- downloadUrl("/cgi-bin/ifconfig.cgi?arg=eth1", processIfconfig);
+ downloadUrl("/cgi-bin/ifconfig.cgi?arg=eth0", processIfconfig);
+// downloadUrl("/cgi-bin/ifconfig.cgi?arg=eth1", processIfconfig);
}
projects / freetel-svn-tracking.git / commitdiff