;typical NCS cable modem definition
;[x1-6-00-50-bf-b8-18-67.mtt.com]
;[x1-6-00-22-2d-62-cd-97.mtt.com]
;host=10.15.51.36
;dtmfmode= inband;rfc2833
;context = default
;
;ncs=1
;;nat=no
;wcardep= aaln/*
;callerid ="David_1" <85592926>
;;"Anonymous" <Anonymous>;"Private" <Private>;"David_1" <85592926>
;line => aaln/1
;callwaiting=yes
;cancallforward=yes
;canreinvite=no
;transfer=yes
;;callerid ="David_23" <85592272>
;line => aaln/2

[DUT1-112233.sh.useasp.net]     -------------[HostName+DomainName]
host=dynamic
dtmfmode= inband;rfc2833
context = default
ncs=1
;nat=no
wcardep= aaln/*
callerid ="dut1-112233"
line => aaln/1
callwaiting=yes
cancallforward=yes
canreinvite=no
transfer=yes
line => aaln/2

;;;;;;;;;;;;;;;;;;;;;;
; Custom patch for "template" cable modem.
; Any undefined NCS gateway can register with *
; It will get this template definition after first RSIP (or any other message)
; warning - this setup has no security ACLs !
; 'mgcp reload' deletes all 'template cloned' entries that are not in use
;[template]
;host=dynamic
;dtmfmode=rfc2833
;ncs=1
;nat=no
;canreinvite=no
;wcardep= aaln/*

;context = default
;immediate= no
;line => aaln/1

;context = default
;immediate= no
;line => aaln/2

;;;kb add below
;;[001ceaa51302]
;;;context=ext-local
;;context=default
;;host=dynamic
;;dtmfmode=rfc2833
;;line => aaln/1
;;callwaiting=yes
;;cancallforward=yes
;;canreinvite=no
;;transfer=yes
;;line => aaln/2
;;;end kb add
;
;
;;kb add below
;;[10.15.80.12]
;[10.15.81.185]
;;context=ext-local
;wcardep=aaln/*
;;context=default
;;host=dynamic
;host=10.15.81.185
;dtmfmode=rfc2833
;line => aaln/1
;callwaiting=yes
;cancallforward=yes
;canreinvite=no
;transfer=yes
;line => aaln/2
;;end kb add

[root@sipserver ~]#vim /etc/asterisk/sip.conf 配置sip

;[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.
;t38pt_usertpsource=yes         ; Use the source IP address of RTP as the destination IP address for UDPTL packets
                                ; if the nat option is enabled. If a single RTP packet is received Asterisk will know the
                                ; external IP address of the remote device. If port forwarding is done at the client side
                                ; then UDPTL will flow to the remote device.

[11112]
type=friend                     ; Friends place calls and receive calls
context=from-sip                ; Context for incoming calls from this user
secret=11112
host=dynamic                    ; This peer register with us
dtmfmode=rfc2833                ; Choices are inband, rfc2833, or info
                                 ; 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"
 

[11111]
type=friend                     ; Friends place calls and receive calls
context=from-sip                ; Context for incoming calls from this user
secret=11111
host=dynamic                    ; This peer register with us
dtmfmode=rfc2833                ; Choices are inband, rfc2833, or info
                                 ; 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

[root@sipserver ~]# vim /etc/asterisk/extensions.conf(设置拨号规则)
; 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 bugs.digium.com. 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
;
; 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=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 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

;
; 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,...)
;
; 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.
;

;
; Sample entries for extensions.conf
;
;
[dundi-e164-canonical]
;
; List canonical entries here
;
;exten => 12564286000,1,Gosub(stdexten,s,1(6000,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.
;
; 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)}/${EXTEN:${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}

[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:
;   ${ARG1} - Extension
;   ${ARG2} - Device(s) to ring
;   ${ARG3} - Optional context in Voicemail (if empty, then "default")
;
; Note that the current version will drop through to the next priority in the
; case of their pressing '#'.  This gives more flexibility in what do to next:
; you can prompt for a new extension, or drop the call, or send them to a
; general delivery mailbox, or...
;
; The use of the LOCAL() function is purely for convenience.  Any variable
; initially declared as LOCAL() will disappear when the innermost Gosub context
; in which it was declared returns.  Note also that you can declare a LOCAL()
; variable on top of an existing variable, and its value will revert to its
; previous value (before being declared as LOCAL()) upon Return.
;
exten => s,1,NoOp(Start stdexten)
exten => s,n,Set(LOCAL(ext)=${ARG1})
exten => s,n,Set(LOCAL(dev)=${ARG2})
exten => s,n,Set(LOCAL(cntx)=${ARG3})

exten => s,n,Set(LOCAL(mbx)="${ext}"$["${cntx}" ? "@${cntx}" :: ""])
exten => s,n,Dial(${dev},20)                    ; Ring the interface, 20 seconds maximum
exten => s,n,Goto(s-${DIALSTATUS},1)            ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

exten => s-NOANSWER,1,Voicemail(${mbx},u)       ; If unavailable, send to voicemail w/ unavail announce
exten => s-NOANSWER,n,NoOp(Finish stdexten NOANSWER)
exten => s-NOANSWER,n,Return()                  ; If they press #, return to start

exten => s-BUSY,1,Voicemail(${mbx},b)
                                                ; If busy, send to voicemail w/ busy announce
exten => s-BUSY,n,NoOp(Finish stdexten BUSY)
exten => s-BUSY,n,Return()                      ; If they press #, return to start

exten => _s-.,1,Goto(s-NOANSWER,1)              ; Treat anything else as no answer

exten => a,1,VoicemailMain(${mbx})              ; If they press *, send the user into VoicemailMain
                                                ; does this ever 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)`
;   ${ARG5} - Context in voicemail (if empty, then "default")
;
; See above note in stdexten about priority handling on exit.
;
exten => s,1,NoOp(Start stdPrivacyexten)
exten => s,n,Set(LOCAL(ext)=${ARG1})
exten => s,n,Set(LOCAL(dev)=${ARG2})
exten => s,n,Set(LOCAL(dontcntx)=${ARG3})
exten => s,n,Set(LOCAL(tortcntx)=${ARG4})
exten => s,n,Set(LOCAL(cntx)=${ARG5})

exten => s,n,Set(LOCAL(mbx)="${ext}"$["${cntx}" ? "@${cntx}" :: ""])
exten => s,n,Dial(${dev},20,p)                  ; Ring the interface, 20 seconds maximum, call screening
                                                ; option (or use P for databased call screening)
exten => s,n,Goto(s-${DIALSTATUS},1)            ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

exten => s-NOANSWER,1,Voicemail(${mbx},u)       ; If unavailable, send to voicemail w/ unavail announce
exten => s-NOANSWER,n,NoOp(Finish stdPrivacyexten NOANSWER)
exten => s-NOANSWER,n,Return()                  ; If they press #, return to start

exten => s-BUSY,1,Voicemail(${mbx},b)           ; If busy, send to voicemail w/ busy announce
exten => s-BUSY,n,NoOp(Finish stdPrivacyexten BUSY)
exten => s-BUSY,n,Return()                      ; If they press #, return to start

exten => s-DONTCALL,1,Goto(${dontcntx},s,1)     ; Callee chose to send this call to a polite "Don't call again" script.

exten => s-TORTURE,1,Goto(${tortcntx},s,1)      ; Callee chose to send this call to a telemarketer torture script.

exten => _s-.,1,Goto(s-NOANSWER,1)              ; Treat anything else as no answer

exten => a,1,VoicemailMain(${mbx})              ; If they press *, send the user into VoicemailMain
                                                ; does this ever 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(${ARG1},js)                    ; j is for Jump and s is for ANY call
exten => s,n,GoToIf([${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,Gosub(stdexten,s,1(1234,${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)

[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/${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

;add for useasp test
exten => 8101,1,Dial(MGCP/aaln/1@dut1-112233.sh.useasp.net)
exten => 8102,1,Dial(MGCP/aaln/2@dut1-112233.sh.useasp.net)

exten => 8888,1,Dial(MGCP/aaln/1@jerry-pc)
exten => 8899,1,Dial(MGCP/aaln/1@lc-pc)

exten => 11112,1,Dial(SIP/11112)
exten => 11111,1,Dial(SIP/11111)

;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(stdexten,s,1(6275,${MARK}))
                                                ; assuming ${MARK} is something like DAHDI/2
;exten => 6275,n,Goto(default,s,1)              ; exited Voicemail
;exten => mark,1,Goto(6275,1)                   ; alias mark to 6275
;exten => 6536,1,Gosub(stdexten,s,1(6236,${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
;
; 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"

[root@sipserver ~]# asterisk 启用asterisk

[root@sipserver ~]# asterisk –r （进入asterisk的cli）
Asterisk 1.6.0.1, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 1.6.0.1 currently running on sipserver (pid = 17802)
Verbosity is at least 3
sipserver*CLI>
sipserver*CLI> restart now（重启 Server）
sipserver*CLI> mgcp show endpoints