configs: Move sample config files into a subdirectory of configs

This moves all samples configs from configs/ to configs/samples. This allows
for additional sets of sample configuration files to be added in the future.

Review: https://reviewboard.asterisk.org/r/3804/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418870 65c4cc65-6c06-0410-ace0-fbb531ad65f3

1 files changed, 857 insertions, 0 deletions

diff --git a/configs/samples/extensions.conf.sample b/configs/samples/extensions.conf.sample
new file mode 100644
index 000000000..df91223f0
--- /dev/null
+++ b/configs/samples/extensions.conf.sample
@@ -0,0 +1,857 @@
+; 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 "dialplan reload" command in the CLI
+; - With the "reload" command (that reloads everything) in the CLI
+
+;
+; 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 "dialplan save" 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 extenpatternmatchnew is set (true, yes, etc), then a new algorithm that uses
+; a Trie to find the best matching pattern is used. In dialplans
+; with more than about 20-40 extensions in a single context, this
+; new algorithm can provide a noticeable speedup.
+; With 50 extensions, the speedup is 1.32x
+; with 88 extensions, the speedup is 2.23x
+; with 138 extensions, the speedup is 3.44x
+; with 238 extensions, the speedup is 5.8x
+; with 438 extensions, the speedup is 10.4x
+; With 1000 extensions, the speedup is ~25x
+; with 10,000 extensions, the speedup is 374x
+; Basically, the new algorithm provides a flat response
+; time, no matter the number of extensions.
+;
+; By default, the old pattern matcher is used.
+;
+; ****This is a new feature! *********************
+; The new pattern matcher is for the brave, the bold, and
+; the desperate. If you have large dialplans (more than about 50 extensions
+; in a context), and/or high call volume, you might consider setting
+; this value to "yes" !!
+; Please, if you try this out, and are forced to return to the
+; old pattern matcher, please report your reasons in a bug report
+; on https://issues.asterisk.org. We have made good progress in providing
+; something compatible with the old matcher; help us finish the job!
+;
+; This value can be switched at runtime using the cli command "dialplan set extenpatternmatchnew true"
+; or "dialplan set extenpatternmatchnew false", so you can experiment to your hearts content.
+;
+;extenpatternmatchnew=no
+;
+; If clearglobalvars is set, global variables will be cleared
+; and reparsed on a dialplan 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
+;
+; 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"
+;#include <filename.conf>
+;#include filename.conf
+;
+; You can execute a program or script that produces config files, and they
+; will be inserted where you insert the #exec command. The #exec command
+; works on all asterisk configuration files.  However, you will need to
+; activate them within asterisk.conf with the "execincludes" option.  They
+; are otherwise considered a security risk.
+;#exec /opt/bin/build-extra-contexts.sh
+;#exec /opt/bin/build-extra-contexts.sh --foo="bar"
+;#exec </opt/bin/build-extra-contexts.sh --foo="bar">
+;#exec "/opt/bin/build-extra-contexts.sh --foo=\"bar\""
+;
+
+; 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=DAHDI/1
+;CONSOLE=Phone/phone0
+IAXINFO=guest					; IAXtel username/password
+;IAXINFO=myuser:mypass
+TRUNK=DAHDI/G2					; Trunk interface
+;
+; Note the 'G2' in the TRUNK variable above. It specifies which group (defined
+; in chan_dahdi.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 DAHDI channel
+;    (aka. ascending sequential hunt group).
+; G: select the highest-numbered non-busy DAHDI 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
+
+;FREENUMDOMAIN=mydomain.com                     ; domain to send on outbound
+                                                ; freenum calls (uses outbound-freenum
+                                                ; context)
+
+;
+; WARNING WARNING WARNING WARNING
+; If you load any other extension configuration engine, such as pbx_ael.so,
+; your global variables may be overridden by that file.  Please take care to
+; use only one location to set global variables, and you will likely save
+; yourself a ton of grief.
+; WARNING WARNING WARNING WARNING
+;
+; 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 parentheses after their name which can be used in goto situations.
+;
+; Contexts contain several lines, one for each step of each extension.  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.
+; Switches may also be included within a context.  The order of matching within
+; a context is always exact extensions, pattern match extensions, includes, and
+; switches.  Includes are always processed depth-first.  So for example, if you
+; would like a switch "A" to match before context "B", simply put switch "A" in
+; an included context "C", where "C" is included in your original context
+; before "B".
+;
+;[context]
+;exten => someexten,{priority|label{+|-}offset}[(alias)],application(arg1,arg2,...)
+;
+; Timing list for includes is
+;
+;   <time range>,<days of week>,<days of month>,<months>[,<timezone>]
+;
+; 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.  Please note that ignorepat
+; only works with channels which receive dialtone from the PBX, such as DAHDI,
+; Phone, and VPB.  Other channels, such as SIP and MGCP, which generate their
+; own dialtone and converse with the PBX only after a number is complete, are
+; generally unaffected by ignorepat (unless DISA or another method is used to
+; generate a dialtone after answering the channel).
+;
+
+;
+; Sample entries for extensions.conf
+;
+;
+[dundi-e164-canonical]
+;include => stdexten
+;
+; List canonical entries here
+;
+;exten => 12564286000,1,Gosub(6000,stdexten(IAX2/foo))
+;exten => 12564286000,n,Goto(default,s,1)	; exited Voicemail
+;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(DAHDI/G2/${EXTEN:7}) ; Expose all of 256-428
+;exten => _1256325XXXX,1,Dial(DAHDI/G2/${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.
+;
+; Note: In old versions of Asterisk the PBX in some cases defaulted to
+; extension "s" when a given extension was wrong (like in AMI originate).
+; This is no longer the case.
+;
+; For wildcard matches, see above - all pattern matches start with
+; an underscore.
+exten => s,1,Goto(${ARG1},1)
+include => dundi-e164-lookup
+
+;
+; Here are the entries you need to participate in the IAXTEL
+; call routing system.  Most IAXTEL numbers begin with 1-700, but
+; there are exceptions.  For more information, and to sign
+; up, please go to www.gnophone.com or www.iaxtel.com
+;
+[iaxtel700]
+exten => _91700XXXXXXX,1,Dial(IAX2/${GLOBAL(IAXINFO)}@iaxtel.com/${EXTEN:1}@iaxtel)
+
+;
+; The SWITCH statement permits a server to share the dialplan with
+; another server. Use with care: Reciprocal switch statements are not
+; allowed (e.g. both A -> B and B -> A), and the switched server needs
+; to be on-line or else dialing can be severly delayed.
+;
+[iaxprovider]
+;switch => IAX2/user:[key]@myserver/mycontext
+
+[trunkint]
+;
+; International long distance through trunk
+;
+exten => _9011.,1,Macro(dundi-e164,${EXTEN:4})
+exten => _9011.,n,Dial(${GLOBAL(TRUNK)}/${FILTER(0-9,${EXTEN:${GLOBAL(TRUNKMSD)}})})
+
+[trunkld]
+;
+; Long distance context accessed through trunk
+;
+exten => _91NXXNXXXXXX,1,Macro(dundi-e164,${EXTEN:1})
+exten => _91NXXNXXXXXX,n,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
+
+[trunklocal]
+;
+; Local seven-digit dialing accessed through trunk interface
+;
+exten => _9NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
+
+[trunktollfree]
+;
+; Long distance context accessed through trunk interface
+;
+exten => _91800NXXXXXX,1,Dial(${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:${GLOBAL(TRUNKMSD)}})
+
+[international]
+;
+; Master context for international long distance
+;
+ignorepat => 9
+include => longdistance
+include => trunkint
+
+[longdistance]
+;
+; Master context for long distance
+;
+ignorepat => 9
+include => local
+include => trunkld
+
+[local]
+;
+; Master context for local, toll-free, and iaxtel calls only
+;
+ignorepat => 9
+include => default
+include => trunklocal
+include => iaxtel700
+include => trunktollfree
+include => iaxprovider
+
+;Include parkedcalls (or the context you define in features conf)
+;to enable call parking.
+include => parkedcalls
+;
+; You can use an alternative switch type as well, to resolve
+; extensions that are not known here, for example with remote
+; IAX switching you transparently get access to the remote
+; Asterisk PBX
+;
+; switch => IAX2/user:password@bigserver/local
+;
+; An "lswitch" is like a switch but is literal, in that
+; variable substitution is not performed at load time
+; but is passed to the switch directly (presumably to
+; be substituted in the switch routine itself)
+;
+; lswitch => Loopback/12${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@${CURSERVER}
+
+; The following two contexts are a template to enable the ability to dial
+; ISN numbers. For more information about what an ISN number is, please see
+; http://www.freenum.org.
+;
+; This is the dialing hook.  use:
+; include => outbound-freenum
+
+[outbound-freenum]
+; We'll add more digits as needed. The purpose is to dial things
+; like extension numbers at domains (ITAD number) so we're matching
+; on lengths of 1 through 6 prior to the separator (the asterisk [*])
+;
+exten => _X*X!,1,Goto(outbound-freenum2,${EXTEN},1)
+exten => _XX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
+exten => _XXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
+exten => _XXXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
+exten => _XXXXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
+exten => _XXXXXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
+
+[outbound-freenum2]
+; This is the handler which performs the dialing logic. It is called
+; from the [outbound-freenum] context
+;
+exten => _X!,1,Verbose(2,Performing ISN lookup for ${EXTEN})
+same => n,Set(SUFFIX=${CUT(EXTEN,*,2-)})                                ; make sure the suffix is all digits as well
+same => n,GotoIf($["${FILTER(0-9,${SUFFIX})}" != "${SUFFIX}"]?fn-CONGESTION,1)
+                                                                        ; filter out bad characters per the README-SERIOUSLY.best-practices.txt document
+same => n,Set(TIMEOUT(absolute)=10800)
+same => n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org)})     ; perform our lookup with freenum.org
+same => n,GotoIf($["${isnresult}" != ""]?from)
+same => n,Set(DIALSTATUS=CONGESTION)
+same => n,Goto(fn-CONGESTION,1)
+same => n(from),Set(__SIPFROMUSER=${CALLERID(num)})
+same => n,GotoIf($["${GLOBAL(FREENUMDOMAIN)}" = ""]?dial)               ; check if we set the FREENUMDOMAIN global variable in [global]
+same => n,Set(__SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)})                 ;    if we did set it, then we'll use it for our outbound dialing domain
+same => n(dial),Dial(SIP/${isnresult},40)
+same => n,Goto(fn-${DIALSTATUS},1)
+
+exten => fn-BUSY,1,Busy()
+
+exten => _f[n]-.,1,NoOp(ISN: ${DIALSTATUS})
+same => n,Congestion()
+
+[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
+
+[stdexten]
+;
+; Standard extension subroutine:
+;   ${EXTEN} - Extension
+;   ${ARG1} - Device(s) to ring
+;   ${ARG2} - Optional context in Voicemail
+;
+; Note that the current version will drop through to the next priority in the
+; case of their pressing '#'.  This gives more flexibility in what do to next:
+; you can prompt for a new extension, or drop the call, or send them to a
+; general delivery mailbox, or...
+;
+; The use of the LOCAL() function is purely for convenience.  Any variable
+; initially declared as LOCAL() will disappear when the innermost Gosub context
+; in which it was declared returns.  Note also that you can declare a LOCAL()
+; variable on top of an existing variable, and its value will revert to its
+; previous value (before being declared as LOCAL()) upon Return.
+;
+exten => _X.,50000(stdexten),NoOp(Start stdexten)
+exten => _X.,n,Set(LOCAL(ext)=${EXTEN})
+exten => _X.,n,Set(LOCAL(dev)=${ARG1})
+exten => _X.,n,Set(LOCAL(cntx)=${ARG2})
+exten => _X.,n,Set(LOCAL(mbx)=${ext}${IF($[!${ISNULL(${cntx})}]?@${cntx})})
+exten => _X.,n,Dial(${dev},20)				; Ring the interface, 20 seconds maximum
+exten => _X.,n,Goto(stdexten-${DIALSTATUS},1)		; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
+
+exten => stdexten-NOANSWER,1,Voicemail(${mbx},u)	; If unavailable, send to voicemail w/ unavail announce
+exten => stdexten-NOANSWER,n,Return()			; If they press #, return to start
+
+exten => stdexten-BUSY,1,Voicemail(${mbx},b)		; If busy, send to voicemail w/ busy announce
+exten => stdexten-BUSY,n,Return()			; If they press #, return to start
+
+exten => _stde[x]te[n]-.,1,Goto(stdexten-NOANSWER,1)	; Treat anything else as no answer
+
+exten => a,1,VoicemailMain(${mbx})			; If they press *, send the user into VoicemailMain
+exten => a,n,Return()
+
+[stdPrivacyexten]
+;
+; Standard extension subroutine:
+;   ${ARG1} - Extension
+;   ${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)`
+;   ${ARG5} - Context in voicemail (if empty, then "default")
+;
+; See above note in stdexten about priority handling on exit.
+;
+exten => _X.,60000(stdPrivacyexten),NoOp(Start stdPrivacyexten)
+exten => _X.,n,Set(LOCAL(ext)=${ARG1})
+exten => _X.,n,Set(LOCAL(dev)=${ARG2})
+exten => _X.,n,Set(LOCAL(dontcntx)=${ARG3})
+exten => _X.,n,Set(LOCAL(tortcntx)=${ARG4})
+exten => _X.,n,Set(LOCAL(cntx)=${ARG5})
+
+exten => _X.,n,Set(LOCAL(mbx)="${ext}"$["${cntx}" ? "@${cntx}" :: ""])
+exten => _X.,n,Dial(${dev},20,p)			; Ring the interface, 20 seconds maximum, call screening
+						; option (or use P for databased call _X.creening)
+exten => _X.,n,Goto(stdexten-${DIALSTATUS},1)		; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
+
+exten => stdexten-NOANSWER,1,Voicemail(${mbx},u)	; If unavailable, send to voicemail w/ unavail announce
+exten => stdexten-NOANSWER,n,NoOp(Finish stdPrivacyexten NOANSWER)
+exten => stdexten-NOANSWER,n,Return()			; If they press #, return to start
+
+exten => stdexten-BUSY,1,Voicemail(${mbx},b)		; If busy, send to voicemail w/ busy announce
+exten => stdexten-BUSY,n,NoOp(Finish stdPrivacyexten BUSY)
+exten => stdexten-BUSY,n,Return()			; If they press #, return to start
+
+exten => stdexten-DONTCALL,1,Goto(${dontcntx},s,1)	; Callee chose to send this call to a polite "Don't call again" script.
+
+exten => stdexten-TORTURE,1,Goto(${tortcntx},s,1)	; Callee chose to send this call to a telemarketer torture script.
+
+exten => _stde[x]te[n]-.,1,Goto(stdexten-NOANSWER,1)	; Treat anything else as no answer
+
+exten => a,1,VoicemailMain(${mbx})		; If they press *, send the user into VoicemailMain
+exten => a,n,Return
+
+[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},s)			; 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(${ARG1})
+exten => s,n(fail),Hangup
+
+
+[demo]
+include => stdexten
+;
+; We start with what to do when a call first comes in.
+;
+exten => s,1,Wait(1)			; Wait a second, just for fun
+exten => s,n,Answer			; Answer the line
+exten => s,n,Set(TIMEOUT(digit)=5)	; Set Digit Timeout to 5 seconds
+exten => s,n,Set(TIMEOUT(response)=10)	; Set Response Timeout to 10 seconds
+exten => s,n(restart),BackGround(demo-congrats)	; Play a congratulatory message
+exten => s,n(instruct),BackGround(demo-instruct)	; Play some instructions
+exten => s,n,WaitExten			; Wait for an extension to be dialed.
+
+exten => 2,1,BackGround(demo-moreinfo)	; Give some more information.
+exten => 2,n,Goto(s,instruct)
+
+exten => 3,1,Set(LANGUAGE()=fr)		; Set language to french
+exten => 3,n,Goto(s,restart)		; Start with the congratulations
+
+exten => 1000,1,Goto(default,s,1)
+;
+; We also create an example user, 1234, who is on the console and has
+; voicemail, etc.
+;
+exten => 1234,1,Playback(transfer,skip)		; "Please hold while..."
+					; (but skip if channel is not up)
+exten => 1234,n,Gosub(${EXTEN},stdexten(${GLOBAL(CONSOLE)}))
+exten => 1234,n,Goto(default,s,1)		; exited Voicemail
+
+exten => 1235,1,Voicemail(1234,u)		; Right to voicemail
+
+exten => 1236,1,Dial(Console/dsp)		; Ring forever
+exten => 1236,n,Voicemail(1234,b)		; Unless busy
+
+;
+; # for when they're done with the demo
+;
+exten => #,1,Playback(demo-thanks)	; "Thanks for trying the demo"
+exten => #,n,Hangup			; Hang them up.
+
+;
+; A timeout and "invalid extension rule"
+;
+exten => t,1,Goto(#,1)			; If they take too long, give up
+exten => i,1,Playback(invalid)		; "That's not valid, try again"
+
+;
+; Create an extension, 500, for dialing the
+; Asterisk demo.
+;
+exten => 500,1,Playback(demo-abouttotry); Let them know what's going on
+exten => 500,n,Dial(IAX2/guest@pbx.digium.com/s@default)	; Call the Asterisk demo
+exten => 500,n,Playback(demo-nogo)	; Couldn't connect to the demo site
+exten => 500,n,Goto(s,6)		; Return to the start over message.
+
+;
+; Create an extension, 600, for evaluating echo latency.
+;
+exten => 600,1,Playback(demo-echotest)	; Let them know what's going on
+exten => 600,n,Echo			; Do the echo test
+exten => 600,n,Playback(demo-echodone)	; Let them know it's over
+exten => 600,n,Goto(s,6)		; Start over
+
+;
+;	You can use the Macro Page to intercom a individual user
+exten => 76245,1,Macro(page,SIP/Grandstream1)
+; or if your peernames are the same as extensions
+exten => _7XXX,1,Macro(page,SIP/${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/${EXTEN})
+
+;[mainmenu]
+;
+; Example "main menu" context with submenu
+;
+;exten => s,1,Answer
+;exten => s,n,Background(thanks)		; "Thanks for calling press 1 for sales, 2 for support, ..."
+;exten => s,n,WaitExten
+;exten => 1,1,Goto(submenu,s,1)
+;exten => 2,1,Hangup
+;include => default
+;
+;[submenu]
+;exten => s,1,Ringing					; Make them comfortable with 2 seconds of ringback
+;exten => s,n,Wait,2
+;exten => s,n,Background(submenuopts)	; "Thanks for calling the sales department.  Press 1 for steve, 2 for..."
+;exten => s,n,WaitExten
+;exten => 1,1,Goto(default,steve,1)
+;exten => 2,1,Goto(default,mark,2)
+
+[public]
+;
+; ATTENTION: If your Asterisk is connected to the internet and you do
+; not have allowguest=no in sip.conf, everybody out there may use your
+; public context without authentication.  In that case you want to
+; double check which services you offer to the world.
+;
+include => demo
+
+[default]
+;
+; By default we include the demo.  In a production system, you
+; probably don't want to have the demo there.
+;
+include => demo
+
+;
+; An extension like the one below can be used for FWD, Nikotel, sipgate etc.
+; Note that you must have a [sipprovider] section in sip.conf
+;
+;exten => _41X.,1,Dial(SIP/${FILTER(0-9,${EXTEN:2})}@sipprovider,,r)
+
+; Real extensions would go here. Generally you want real extensions to be
+; 4 or 5 digits long (although there is no such requirement) and start with a
+; single digit that is fairly large (like 6 or 7) so that you have plenty of
+; room to overlap extensions and menu options without conflict.  You can alias
+; them with names, too, and use global variables
+
+;exten => 6245,hint,SIP/Grandstream1&SIP/Xlite1(Joe Schmoe) ; Channel hints for presence
+;exten => 6245,1,Dial(SIP/Grandstream1,20,rt)	; permit transfer
+;exten => 6245,n(dial),Dial(${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 ${MARK}
+
+;exten => 6275,1,Gosub(${EXTEN},stdexten(${MARK}))
+						; assuming ${MARK} is something like DAHDI/2
+;exten => 6275,n,Goto(default,s,1)		; exited Voicemail
+;exten => mark,1,Goto(6275,1)			; alias mark to 6275
+;exten => 6536,1,Gosub(${EXTEN},stdexten(${WIL}))
+						; Ditto for wil
+;exten => 6536,n,Goto(default,s,1)		; exited Voicemail
+;exten => wil,1,Goto(6236,1)
+
+;If you want to subscribe to the status of a parking space, this is
+;how you do it. Subscribe to extension 6600 in sip, and you will see
+;the status of the first parking lot with this extensions' help
+;exten => 6600,hint,park:701@parkedcalls
+;exten => 6600,1,noop
+;
+;To subscribe to the availability of a free member in the 'markq' queue.
+;Note: '_avail' is added to the QueueName
+;exten => 8501,hint,Queue:markq_avail
+;exten => 8501,1,Queue(markq)
+;
+; You can also monitor the status of a queue by providing a hint for a
+; particular queue name.
+;exten => 8502,hint,Queue:markq
+;exten => 8502,1,Queue(markq)
+;
+; Some other handy things are an extension for checking voicemail via
+; voicemailmain
+;
+;exten => 8500,1,VoicemailMain
+;exten => 8500,n,Hangup
+;
+; Or a conference room (you'll need to edit meetme.conf to enable this room)
+;
+;exten => 8600,1,Meetme(1234)
+;
+; Or playing an announcement to the called party, as soon it answers
+;
+;exten = 8700,1,Dial(${MARK},30,A(/path/to/my/announcemsg))
+;
+
+; example of a compartmentalized company called "acme"
+;
+; this is the context that your incoming IAX/SIP trunk dumps you in...
+;[acme-incoming]
+;exten => s,1,Wait(1)
+;exten => s,n,Answer()
+;exten => s,n(menu),Playback(acme/vm-brief-menu)
+;exten => s,n(exten),Background(vm-enter-num-to-call)
+;exten => s,n,WaitExten(5)
+;exten => s,n(goodbye),Playback(vm-goodbye)
+;exten => s,n(end),Hangup()
+;
+;include  => acme-extens
+;
+;exten => i,1,Playback(vm-invalid)
+;exten => i,n,Goto(s,exten)			; optionally, transfer to operator
+;
+;exten => t,1,Goto(s,goodbye)
+;
+; this is the context our internal SIP hardphones use (see sip.conf)
+;
+;[acme-internal]
+;exten => s,1,Answer()
+;exten => s,n(exten),Background(vm-enter-num-to-call)
+;exten => s,n,WaitExten(5)
+;exten => s,n(goodbye),Playback(vm-goodbye)
+;exten => s,n(end),Hangup()
+;
+;include => trunkint
+;include => trunkld
+;include => trunklocal
+;
+;include => acme-extens
+;
+; you can test what your system sounds like to outside callers by dialing this
+;exten => 777,1,DISA(no-password,acme-incoming)
+;
+; grouping of acme's extensions... never used directly, always included.
+;
+;[acme-extens]
+;include => stdexten
+;exten => 111,1,Gosub(111,stdexten(SIP/pete_1,acme))
+;exten => 111,n,Goto(s,exten)
+;
+;exten => 112,1,Gosub(112,stdexten(SIP/nancy_1,acme))
+;exten => 112,n,Goto(s,end)
+;
+; end of acme example
+
+;
+; Time context: you can patch this in via the following.
+;
+; [acme-internal]
+; ...
+; exten => 777,1,Gosub(time)
+; exten => 777,n,Hangup()
+;
+; ...
+; include => time
+;
+; Note: if you're geographically spread out, you can have SIP extensions
+; specify their own local timezone in sip.conf as:
+;
+; [boi]
+; type=friend
+; context=acme-internal
+; callerid="Boise Ofc. <2083451111>"
+; ...
+; ; use system-wide default timezone of MST7MDT
+;
+; [lws]
+; type=friend
+; context=acme-internal
+; callerid="Lewiston Ofc. <2087431111>"
+; ...
+; setvar=timezone=PST8PDT
+;
+; "timezone" isn't a 'reserved' name in any way, and other places where
+; the timezone is significant (e.g. calls to "SayUnixTime()", etc) will
+; require modification as well.  Note that voicemail.conf already has
+; a mechanism for timezones.
+;
+
+[time]
+exten => _X.,30000(time),NoOp(Time: ${EXTEN} ${timezone})
+exten => _X.,n,Wait(0.25)
+exten => _X.,n,Answer()
+; the amount of delay is set for English; you may need to adjust this time
+; for other languages if there's no pause before the synchronizing beep.
+exten => _X.,n,Set(FUTURETIME=$[${EPOCH} + 12])
+exten => _X.,n,SayUnixTime(${FUTURETIME},Zulu,HNS)
+exten => _X.,n,SayPhonetic(z)
+; use the timezone associated with the extension (sip only), or system-wide
+; default if one hasn't been set.
+exten => _X.,n,SayUnixTime(${FUTURETIME},${timezone},HNS)
+exten => _X.,n,Playback(spy-local)
+exten => _X.,n,WaitUntil(${FUTURETIME})
+exten => _X.,n,Playback(beep)
+exten => _X.,n,Return()
+
+;
+; ANI context: use in the same way as "time" above
+;
+
+[ani]
+exten => _X.,40000(ani),NoOp(ANI: ${EXTEN})
+exten => _X.,n,Wait(0.25)
+exten => _X.,n,Answer()
+exten => _X.,n,Playback(vm-from)
+exten => _X.,n,SayDigits(${CALLERID(ani)})
+exten => _X.,n,Wait(1.25)
+exten => _X.,n,SayDigits(${CALLERID(ani)})	; playback again in case of missed digit
+exten => _X.,n,Return()
+
+; For more information on applications, just type "core show applications" at your
+; friendly Asterisk CLI prompt.
+;
+; "core show application <command>" will show details of how you
+; use that particular application in this file, the dial plan.
+; "core show functions" will list all dialplan functions
+; "core show function <COMMAND>" will show you more information about
+; one function. Remember that function names are UPPER CASE.