! Attention ! BOUNTY 80$ who HELP with ooh323.conf

Hello,
my name is Rudolf Jurco. I am working in IPEC Company as IT Manager, and i have problem with integrate Asterisk and GNUGK.So who help me he has got reward 80$ dollars.I am sorry for my bad English.So whats ago. I have installed AsteriskNOW 1.5 and on other server is installed GNUGK with registred H.323 extensions.I have registred Trixbox on GNUGK as gateway with extensions 100,101,105 when i dial from GNUGK 100,101 or 105 asterisk automaticaly answer “Goodbye” AND this is a problem i need most this H.323 extensions to SIP.

This is my configuration files.:

OOH323.conf

; ---------------------------------------------------------------------------------
; — ******* IMPORTANT NOTE ***********
; —
; — This module is currently unsupported. Use it at your own risk.
; —
; ---------------------------------------------------------------------------------

; Objective System’s H323 Configuration example for Asterisk
; ooh323c driver configuration
;
; [general] section defines global parameters
;
; This is followed by profiles which can be of three types - user/peer/friend
; Name of the user profile should match with the h323id of the user device.
; For peer/friend profiles, host ip address must be provided as “dynamic” is
; not supported as of now.
;
; Syntax for specifying a H323 device in extensions.conf is
; For Registered peers/friends profiles:
; OOH323/name where name is the name of the peer/friend profile.
;
; For unregistered H.323 phones:
; OOH323/ip[:port] OR if gk is used OOH323/alias where alias can be any H323
; alias
;
; For dialing into another asterisk peer at a specific exten
; OOH323/exten/peer OR OOH323/exten@ip
;
; Domain name resolution is not yet supported.
;
; When a H.323 user calls into asterisk, his H323ID is matched with the profile
; name and context is determined to route the call
;
; The channel driver will register all global aliases and aliases defined in
; peer profiles with the gatekeeper, if one exists. So, that when someone
; outside our pbx (non-user) calls an extension, gatekeeper will route that
; call to our asterisk box, from where it will be routed as per dial plan.

[general]
;Define the asetrisk server h323 endpoint

;The port asterisk should listen for incoming H323 connections.
;Default - 1720
port=1720

;The dotted IP address asterisk should listen on for incoming H323
;connections
;Default - tries to find out local ip address on it’s own
bindaddr=10.0.0.91

;This parameter indicates whether channel driver should register with
;gatekeeper as a gateway or an endpoint.
;Default - no
gateway=yes

;Whether asterisk should use fast-start and tunneling for H323 connections.
;Default - yes
;faststart=no
;h245tunneling=no

;Whether media wait for connect
;Default - No
;mediawaitforconnect=yes

;H323-ID to be used for asterisk server
;Default - Asterisk PBX
h323id=trixbox
e164=100

;CallerID to use for calls
;Default - Same as h323id
callerid=trixbox

;Whether this asterisk server will use gatekeeper.
;Default - DISABLE
;gatekeeper = 10.0.0.81
gatekeeper = 10.0.0.81

;Location for H323 log file
;Default - /var/log/asterisk/h323_log
;logfile=/var/log/asterisk/h323_log

;Following values apply to all users/peers/friends defined below, unless
;overridden within their client definition

;Sets default context all clients will be placed in.
;Default - default
context=vidsoft

;Sets rtptimeout for all clients, unless overridden
;Default - 60 seconds
;rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity
; when we’re not on hold

;Type of Service
;Default - none (lowdelay, thoughput, reliability, mincost, none)
;tos=lowdelay

;amaflags = default

;The account code used by default for all clients.
;accountcode=h3230101

;The codecs to be used for all clients.Only ulaw and gsm supported as of now.
;Default - ulaw
; ONLY ulaw, gsm, g729 and g7231 supported as of now
allow=all ;Note order of disallow/allow is important.
;allow=gsm
;allow=ulaw
;allow=g729
;allow=g7321

; dtmf mode to be used by default for all clients. Supports rfc2833, q931keypad
; h245alphanumeric, h245signal.
;Default - rfc 2833
dtmfmode=rfc2833

; User/peer/friend definitions:
; User config options Peer config options
; ------------------ -------------------
; context
; disallow disallow
; allow allow
; accountcode accountcode
; amaflags amaflags
; dtmfmode dtmfmode
; rtptimeout ip
; port
; h323id
; email
; url
; e164
; rtptimeout

;

;Define users here
;Section header is extension
;[myuser1]
;type=user
;context=context1
;disallow=all
;allow=all
;allow=ulaw

[gatekeeper]
type=peer
context=vidsoft
ip=10.0.0.81 ; UPDATE with appropriate ip address
port=1720 ; UPDATE with appropriate port
e164=101

[myfriend1]
type=friend
context=default
ip=10.0.0.117 ; UPDATE with appropriate ip address
port=1720 ; UPDATE with appropriate port
allow=all
allow=gsm
allow=ulaw
allow=g729
allow=g7321
e164=105
rtptimeout=60
dtmfmode=rfc2833

SIP.conf

;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ;
; this file must be done via the web gui. There are alternative files to make ;
; custom modifications, details at: freepbx.org/configuration_files ;
;--------------------------------------------------------------------------------;
;

[general]

; These files will all be included in the [general] context
;
#include sip_general_additional.conf

;sip_general_custom.conf is the proper file location for placing any sip general
;options that you might need set. For example: enable and force the sip jitterbuffer.
;If these settings are desired they should be set the sip_general_custom.conf file.
;
; jbenable=yes
; jbforce=yes
;
;It is also the proper place to add the lines needed for sip nat’ing when going
;through a firewall. For nat’ing you’d need to add the following lines:
; nat=yes , externip= , localhost= , and optionally fromdomain= .
;
#include sip_general_custom.conf

;sip_nat.conf is here for legacy support reasons and for those that upgrade
;from previous versions. If you have this file with lines in it please make
;sure they are not duplicated in sip_general_custom.conf, if so remove them
;from sip_nat.conf as sip_general_custom.conf will have precedence.
#include sip_nat.conf

;sip_registrations_custom.conf is for any customizations you might need to do to
;the automatically generated registrations that FreePBX makes.
;
#include sip_registrations_custom.conf
#include sip_registrations.conf

; These files should all be expected to come after the [general] context
;
#include sip_custom.conf
#include sip_additional.conf

;sip_custom_post.conf If you have extra parameters that are needed for a
;extension to work to for example, those go here. So you have extension
;1000 defined in your system you start by creating a line 1000 in this
;file. Then on the next line add the extra parameter that is needed.
;When the sip.conf is loaded it will append your additions to the end of
;that extension.
;
#include sip_custom_post.conf

Extensions.conf

;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ;
; this file must be done via the web gui. There are alternative files to make ;
; custom modifications, details at: freepbx.org/configuration_files ;
;--------------------------------------------------------------------------------;
;

; FreePBX
; Copyright © 2004 Coalescent Systems Inc (Canada)
; Copyright © 2006 Why Pay More 4 Less Pty Ltd (Australia)
; Copyright © 2007 Astrogen LLC (USA)
; Released under the GNU GPL Licence version 2.

; dialparties.agi (sprackett.com/asterisk/)
; Asterisk::AGI (asterisk.gnuinter.net/)
; gsm (ibiblio.org/pub/Linux/utils/ … short.html)
; loligo sounds (loligo.com/asterisk/sounds/)
; mpg123 (voip-info.org/wiki-Asterisk+conf … nhold.conf)

;************************** -WARNING- ****************************************
; *
; This include file is to be used with extreme caution. In almost all cases *
; any custom dialplan SHOULD be put in extensions_custom.conf which will *
; not hurt a FreePBX generated dialplan. In some very rare and custom *
; situations users may have a need to override what freepbx automatically *
; generates. If so anything in this file will do that. If you come up with a *
; situation where you need to modify the existing dialplan or macro, please *
; put it here and also notify the FreePBX development team so they can take it *
; into account in the future. *
; *
#include extensions_override_freepbx.conf
; *
;************************** -WARNING- ****************************************

; include extension contexts generated from AMP
#include extensions_additional.conf

; Customizations to this dialplan should be made in extensions_custom.conf
; See extensions_custom.conf.sample for an example.
; If you need to use [macro-dialout-trunk-predial-hook], [ext-did-custom], or
; [from-internal-custom] for example, place these in this file or they will get overwritten.
;
#include extensions_custom.conf

[from-trunk] ; just an alias since VoIP shouldn’t be called PSTN
include => from-pstn

[from-pstn]
include => from-pstn-custom ; create this context in extensions_custom.conf to include customizations
include => ext-did
include => ext-did-post-custom
include => from-did-direct ; MODIFICATOIN (PL) for findmefollow if enabled, should be bofore ext-local
include => ext-did-catchall ; THIS MUST COME AFTER ext-did
exten => fax,1,Goto(ext-fax,in_fax,1)

; MODIFICATION (PL)
;
; Required to assure that direct dids go to personal ring group before local extension.
; This could be auto-generated however I it is prefered to be put here and hard coded
; so that it can be modified if ext-local should take precedence in certain situations.
; will have to decide what to do later.
;
[from-did-direct]
include => ext-findmefollow
include => ext-local

; ############################################################################
; Macros [macro]
; ############################################################################

; Rings one or more extensions. Handles things like call forwarding and DND
; We don’t call dial directly for anything internal anymore.
; ARGS: $TIMER, $OPTIONS, $EXT1, $EXT2, $EXT3, …
; Use a Macro call such as the following:
; Macro(dial,$DIAL_TIMER,$DIAL_OPTIONS,$EXT1,$EXT2,$EXT3,…)
[macro-dial]
exten => s,1,GotoIf($["${MOHCLASS}" = “”]?dial)
exten => s,n,SetMusicOnHold(${MOHCLASS})
exten => s,n(dial),AGI(dialparties.agi)
exten => s,n,NoOp(Returned from dialparties with no extensions to call and DIALSTATUS: ${DIALSTATUS})

exten => s,n+2(normdial),Dial(${ds}) ; dialparties will set the priority to 10 if $ds is not null
exten => s,n,Set(DIALSTATUS=${IF($["${DIALSTATUS_CW}"!="" ]?${DIALSTATUS_CW}:${DIALSTATUS})})
exten => s,n,GosubIf($["${SCREEN}" != “”]?${DIALSTATUS},1)

exten => s,20(huntdial),NoOp(Returned from dialparties with hunt groups to dial )
exten => s,n,Set(HuntLoop=0)
exten => s,n(a22),GotoIf($[${HuntMembers} >= 1]?a30) ; if this is from rg-group, don’t strip prefix
exten => s,n,NoOp(Returning there are no members left in the hunt group to ring)

; dialparties.agi has setup the dialstring for each hunt member in a variable labeled HuntMember0, HuntMember1 etc for each iteration
; and The total number in HuntMembers. So for each iteration, we will update the CALLTRACE Data.
;
exten => s,n+2(a30),Set(HuntMember=HuntMember${HuntLoop})
exten => s,n,GotoIf($[$["${CALLTRACE_HUNT}" != “” ] & $[$["${RingGroupMethod}" = “hunt” ] | $["${RingGroupMethod}" = “firstavailable”] | $["${RingGroupMethod}" = “firstnotonphone”]]]?a32:a35)

exten => s,n(a32),Set(CT_EXTEN=${CUT(FILTERED_DIAL,$[${HuntLoop} + 1])})
exten => s,n,Set(DB(CALLTRACE/${CT_EXTEN})=${CALLTRACE_HUNT})
exten => s,n,Goto(s,a42)

;Set Call Trace for each hunt member we are going to call “Memory groups have multiple members to set CALL TRACE For hence the loop
;
exten => s,n(a35),GotoIf($[$[”${CALLTRACE_HUNT}" != “” ] & $["${RingGroupMethod}" = “memoryhunt” ]]?a36:a50)
exten => s,n(a36),Set(CTLoop=0)
exten => s,n(a37),GotoIf($[${CTLoop} > ${HuntLoop}]?a42) ; if this is from rg-group, don’t strip prefix
exten => s,n,Set(CT_EXTEN=${CUT(FILTERED_DIAL,$[${CTLoop} + 1])})
exten => s,n,Set(DB(CALLTRACE/${CT_EXTEN})=${CALLTRACE_HUNT})
exten => s,n,Set(CTLoop=$[1 + ${CTLoop}])
exten => s,n,Goto(s,a37)

exten => s,n(a42),Dial(${${HuntMember}}${ds})
exten => s,n,Set(HuntLoop=$[1 + ${HuntLoop}])
exten => s,n,GotoIf($[$[$[“foo${RingGroupMethod}” != “foofirstavailable”] & $[“foo${RingGroupMethod}” != “foofirstnotonphone”]] | $[“foo${DialStatus}” = “fooBUSY”]]?a46)
exten => s,n,Set(HuntMembers=0)
exten => s,n(a46),Set(HuntMembers=$[${HuntMembers} - 1])
exten => s,n,Goto(s,a22)

exten => s,n(a50),DBdel(CALLTRACE/${CT_EXTEN})
exten => s,n,Goto(s,a42)

; For call screening
exten => NOANSWER,1,Macro(vm,${SCREEN_EXTEN},BUSY,${IVR_RETVM})
exten => NOANSWER,n,GotoIf($["${IVR_RETVM}" != “RETURN” | “${IVR_CONTEXT}” = “”]?bye)
exten => NOANSWER,n,Return
exten => NOANSWER,n(bye),Macro(hangupcall)
exten => TORTURE,1,Goto(app-blackhole,musiconhold,1)
exten => TORTURE,n,Macro(hangupcall)
exten => DONTCALL,1,Answer
exten => DONTCALL,n,Wait(1)
exten => DONTCALL,n,Zapateller()
exten => DONTCALL,n,Playback(ss-noservice)
exten => DONTCALL,n,Macro(hangupcall)

; make sure hungup calls go here so that proper cleanup occurs from call confirmed calls and the like
;
exten => h,1,Macro(hangupcall)

; Ring an extension, if the extension is busy or there is no answer send it
; to voicemail
; ARGS: $VMBOX, $EXT
[macro-exten-vm]
exten => s,1,Macro(user-callerid)

exten => s,n,Set(RingGroupMethod=none)
exten => s,n,Set(VMBOX=${ARG1})
exten => s,n,Set(EXTTOCALL=${ARG2})
exten => s,n,Set(CFUEXT=${DB(CFU/${EXTTOCALL})})
exten => s,n,Set(CFBEXT=${DB(CFB/${EXTTOCALL})})
exten => s,n,Set(RT=${IF($[$["${VMBOX}"!=“novm”] | $[“foo${CFUEXT}”!=“foo”]]?${RINGTIMER}:"")})
exten => s,n,Macro(record-enable,${EXTTOCALL},IN)
exten => s,n,Macro(dial,${RT},${DIAL_OPTIONS},${EXTTOCALL})
exten => s,n,GotoIf($[ $["${VMBOX}" != “novm”] & $["${SCREEN}" != “”] & $["${DIALSTATUS}" = “NOANSWER”] ]?exit,return)
exten => s,n,Set(SV_DIALSTATUS=${DIALSTATUS})
exten => s,n,GosubIf($[$["${SV_DIALSTATUS}"=“NOANSWER”] & $["${CFUEXT}"!=""] & $["${SCREEN}" = “”]]?docfu,1) ; check for CFU in use on no answer
exten => s,n,GosubIf($[$["${SV_DIALSTATUS}"=“BUSY”] & $["${CFBEXT}"!=""]]?docfb,1) ; check for CFB in use on busy
exten => s,n,Set(DIALSTATUS=${SV_DIALSTATUS})
exten => s,n,NoOp(Voicemail is ‘${VMBOX}’)
exten => s,n,GotoIf($["${VMBOX}" = “novm”]?s-${DIALSTATUS},1) ; no voicemail in use for this extension
exten => s,n,NoOp(Sending to Voicemail box ${EXTTOCALL})
exten => s,n,Macro(vm,${VMBOX},${DIALSTATUS},${IVR_RETVM})

; Try the Call Forward on No Answer / Unavailable number
exten => docfu,1,Set(RTCFU=${IF($["${VMBOX}"!=“novm”]?${RINGTIMER}:"")})
exten => docfu,n,Dial(Local/${CFUEXT}@from-internal/n,${RTCFU},${DIAL_OPTIONS})
exten => docfu,n,Return

; Try the Call Forward on Busy number
exten => docfb,1,Set(RTCFB=${IF($["${VMBOX}"!=“novm”]?${RINGTIMER}:"")})
exten => docfb,n,Dial(Local/${CFBEXT}@from-internal/n,${RTCFB},${DIAL_OPTIONS})
exten => docfb,n,Return

; Extensions with no Voicemail box reporting BUSY come here
exten => s-BUSY,1,NoOp(Extension is reporting BUSY and not passing to Voicemail)
exten => s-BUSY,n,GotoIf($["${IVR_RETVM}" = “RETURN” & “${IVR_CONTEXT}” != “”]?exit,1)
exten => s-BUSY,n,Playtones(busy)
exten => s-BUSY,n,Busy(20)

; Anything but BUSY comes here
exten => _s-.,1,Noop(IVR_RETVM: ${IVR_RETVM} IVR_CONTEXT: ${IVR_CONTEXT})
exten => _s-.,n,GotoIf($["${IVR_RETVM}" = “RETURN” & “${IVR_CONTEXT}” != “”]?exit,1)
exten => _s-.,n,Playtones(congestion)
exten => _s-.,n,Congestion(10)

; Short burst of tones then return
exten => exit,1,Playback(beep&line-busy-transfer-menu&silence/1)
exten => exit,n(return),MacroExit()

;------------------------------------------------------------------------
; [macro-simple-dial]
;------------------------------------------------------------------------
; This macro was derived from macro-exten-vm, which is what is normally used to
; ring an extension. It has been simplified and designed to never go to voicemail
; and always return regardless of the DIALSTATUS for any incomplete call.
;
; It’s current primary purpose is to allow findmefollow ring an extension prior
; to trying the follow-me ringgroup that is provided.
;
; Ring an extension, if the extension is busy or there is no answer, return
; ARGS: $EXTENSION, $RINGTIME
;------------------------------------------------------------------------
[macro-simple-dial]
exten => s,1,Set(EXTTOCALL=${ARG1})
exten => s,n,Set(RT=${ARG2})
exten => s,n,Set(CFUEXT=${DB(CFU/${EXTTOCALL})})
exten => s,n,Set(CFBEXT=${DB(CFB/${EXTTOCALL})})

exten => s,n,Set(CWI_TMP=${CWIGNORE})
exten => s,n,Macro(dial,${RT},${DIAL_OPTIONS},${EXTTOCALL})
exten => s,n,Set(__CWIGNORE=${CWI_TMP})

exten => s,n,Set(PR_DIALSTATUS=${DIALSTATUS})

; if we return, thus no answer, and they have a CFU setting, then we try that next
;
exten => s,n,GosubIf($[$["${PR_DIALSTATUS}"=“NOANSWER”] & $[“foo${CFUEXT}”!=“foo”]]?docfu,1) ; check for CFU in use on no answer
exten => s,n,GosubIf($[$["${PR_DIALSTATUS}"=“BUSY”] & $[“foo${CFBEXT}”!=“foo”]]?docfb,1) ; check for CFB in use on busy
exten => s,n,Set(DIALSTATUS=${PR_DIALSTATUS})

; Nothing yet, then go to the end (which will just return, but in case we decide to do something with certain
; return situations, this is left in.
;
exten => s,n,Goto(s-${DIALSTATUS},1)

; Try the Call Forward on No Answer / Unavailable number.
; We want to try CFU if set, but we want the same ring timer as was set to our call (or do we want the
; system ringtimer? - probably not). Then if no answer there (assuming it doesn’t drop into their vm or
; something we return, which will have the net effect of returning to the followme setup.)
;
; want to avoid going to other follow-me settings here. So check if the CFUEXT is a user and if it is
; then direct it straight to ext-local (to avoid getting intercepted by findmefollow) otherwise send it
; to from-internal since it may be an outside line.
;
exten => docfu,1,GotoIf( $[ “foo${DB(AMPUSER/${CFUEXT}/device)}” = “foo” ]?chlocal)
exten => docfu,n,Dial(Local/${CFUEXT}@ext-local,${RT},${DIAL_OPTIONS})
exten => docfu,n,Return
exten => docfu,n(chlocal),Dial(Local/${CFUEXT}@from-internal/n,${RT},${DIAL_OPTIONS})
exten => docfu,n,Return

; Try the Call Forward on Busy number
exten => docfb,1,GotoIf( $[ “foo${DB(AMPUSER/${CFBEXT}/device)}” = “foo” ]?chlocal)
exten => docfb,n,Dial(Local/${CFBEXT}@ext-local,${RT},${DIAL_OPTIONS})
exten => docfb,n,Return
exten => docfb,n(chlocal),Dial(Local/${CFBEXT}@from-internal/n,${RT},${DIAL_OPTIONS})
exten => docfb,n,Return

; In all cases of no connection, come here and simply return, since the calling dialplan will
; decide what to do next
exten => _s-.,1,NoOp(Extension is reporting ${EXTEN})
;------------------------------------------------------------------------

; get the voicemail context for the user in ARG1
[macro-get-vmcontext]
exten => s,1,Set(VMCONTEXT=${DB(AMPUSER/${ARG1}/voicemail)})
exten => s,2,GotoIf($[“foo${VMCONTEXT}” = “foo”]?200:300)
exten => s,200,Set(VMCONTEXT=default)
exten => s,300,NoOp()

; For some reason, if I don’t run setCIDname, CALLERID(name) will be blank in my AGI
; ARGS: none
[macro-fixcid]
exten => s,1,Set(CALLERID(name)=${CALLERID(name)})

; Ring groups of phones
; ARGS: comma separated extension list
; 1 - Ring Group Strategy
; 2 - ringtimer
; 3 - prefix
; 4 - extension list
[macro-rg-group]
exten => s,1,Macro(user-callerid,SKIPTTL) ; already called from ringgroup
exten => s,2,GotoIf($["${CALLERID(name):0:${LEN(${RGPREFIX})}}" != “${RGPREFIX}”]?4:3) ; check for old prefix
exten => s,3,Set(CALLERID(name)=${CALLERID(name):${LEN(${RGPREFIX})}}) ; strip off old prefix
exten => s,4,Set(RGPREFIX=${ARG3}) ; set new prefix
exten => s,5,Set(CALLERID(name)=${RGPREFIX}${CALLERID(name)}) ; add prefix to callerid name
exten => s,6,Set(RecordMethod=Group) ; set new prefix
exten => s,7,Macro(record-enable,${MACRO_EXTEN},${RecordMethod})
exten => s,8,Set(RingGroupMethod=${ARG1}) ;
exten => s,9,Macro(dial,${ARG2},${DIAL_OPTIONS},${ARG4})
exten => s,10,Set(RingGroupMethod=’’) ;

;
; Outgoing channel(s) are busy … inform the client
; but use noanswer features like ringgroups don’t break by being answered
; just to play the message.
;
[macro-outisbusy]
exten => s,1,Playback(all-circuits-busy-now,noanswer)
exten => s,n,Playback(pls-try-call-later,noanswer)
exten => s,n,Macro(hangupcall)

; What to do on hangup.
[macro-hangupcall]
exten => s,1,ResetCDR(vw)
exten => s,n,NoCDR()

; Cleanup any remaining RG flag
;
exten => s,n,GotoIf($[ “x${USE_CONFIRMATION}” = “x” | “x${RINGGROUP_INDEX}” = “x” | “${CHANNEL}” != “${UNIQCHAN}”]?skiprg)
exten => s,n,Noop(Cleaning Up Confirmation Flag: RG/${RINGGROUP_INDEX}/${CHANNEL})
exten => s,n,DBDel(RG/${RINGGROUP_INDEX}/${CHANNEL})

; Cleanup any remaining BLKVM flag
;
exten => s,n(skiprg),GotoIf($[ “x${BLKVM_BASE}” = “x” | “BLKVM/${BLKVM_BASE}/${CHANNEL}” != “${BLKVM_OVERRIDE}” ]?skipblkvm)
exten => s,n,Noop(Cleaning Up Block VM Flag: ${BLKVM_OVERRIDE})
exten => s,n,DBDel(${BLKVM_OVERRIDE})

; Cleanup any remaining FollowMe DND flags
;
exten => s,n(skipblkvm),GotoIf($[ “x${FMGRP}” = “x” | “x${FMUNIQUE}” = “x” | “${CHANNEL}” != “${FMUNIQUE}” ]?theend)
exten => s,n,DBDel(FM/DND/${FMGRP}/${CHANNEL})

exten => s,n(theend),Hangup

[macro-faxreceive]
exten => s,1,Set(FAXFILE=${ASTSPOOLDIR}/fax/${UNIQUEID}.tif)
exten => s,2,Set(EMAILADDR=${FAX_RX_EMAIL})
exten => s,3,rxfax(${FAXFILE})
exten => s,103,Set(EMAILADDR=${FAX_RX_EMAIL})
exten => s,104,Goto(3)

; dialout and strip the prefix
[macro-dialout]
exten => s,1,Macro(user-callerid,SKIPTTL)
exten => s,2,GotoIf($["${ECID${CALLERID(number)}}" = “”]?5) ;check for CID override for exten
exten => s,3,Set(CALLERID(all)=${ECID${CALLERID(number)}})
exten => s,4,Goto(7)
exten => s,5,GotoIf($["${OUTCID_${ARG1}}" = “”]?7) ;check for CID override for trunk
exten => s,6,Set(CALLERID(all)=${OUTCID_${ARG1}})
exten => s,7,Set(length=${LEN(${DIAL_OUT_${ARG1}})})
exten => s,8,Dial(${OUT_${ARG1}}/${ARG2:${length}})
exten => s,9,Playtones(congestion)
exten => s,10,Congestion(5)
exten => s,109,Macro(outisbusy)

; dialout using default OUT trunk - no prefix
[macro-dialout-default]
exten => s,1,Macro(user-callerid,SKIPTTL)
exten => s,2,Macro(record-enable,${CALLERID(number)},OUT)
exten => s,3,Macro(outbound-callerid,${ARG1})
exten => s,4,Dial(${OUT}/${ARG1})
exten => s,5,Playtones(congestion)
exten => s,6,Congestion(5)
exten => s,105,Macro(outisbusy)

[macro-dialout-trunk-predial-hook]
; this macro intentially left blank so it may be safely overwritten for any custom
; requirements that an installation may have.
;
; MACRO RETURN CODE: ${PREDIAL_HOOK_RET}
; if set to “BYPASS” then this trunk will be skipped
;
exten => s,1,MacroExit()

; This macro is for dev purposes and just dumps channel/app variables. Useful when designing new contexts.
[macro-dumpvars]
exten => s,1,Noop(ACCOUNTCODE=${ACCOUNTCODE})
exten => s,2,Noop(ANSWEREDTIME=${ANSWEREDTIME})
exten => s,3,Noop(BLINDTRANSFER=${BLINDTRANSFER})
exten => s,4,Noop(CALLERID=${CALLERID(all)})
exten => s,5,Noop(CALLERID(name)=${CALLERID(name)})
exten => s,6,Noop(CALLERID(number)=${CALLERID(number)})
exten => s,7,Noop(CALLINGPRES=${CALLINGPRES})
exten => s,8,Noop(CHANNEL=${CHANNEL})
exten => s,9,Noop(CONTEXT=${CONTEXT})
exten => s,10,Noop(DATETIME=${DATETIME})
exten => s,11,Noop(DIALEDPEERNAME=${DIALEDPEERNAME})
exten => s,12,Noop(DIALEDPEERNUMBER=${DIALEDPEERNUMBER})
exten => s,13,Noop(DIALEDTIME=${DIALEDTIME})
exten => s,14,Noop(DIALSTATUS=${DIALSTATUS})
exten => s,15,Noop(DNID=${DNID})
exten => s,16,Noop(EPOCH=${EPOCH})
exten => s,17,Noop(EXTEN=${EXTEN})
exten => s,18,Noop(HANGUPCAUSE=${HANGUPCAUSE})
exten => s,19,Noop(INVALID_EXTEN=${INVALID_EXTEN})
exten => s,20,Noop(LANGUAGE=${LANGUAGE})
exten => s,21,Noop(MEETMESECS=${MEETMESECS})
exten => s,22,Noop(PRIORITY=${PRIORITY})
exten => s,23,Noop(RDNIS=${RDNIS})
exten => s,24,Noop(SIPDOMAIN=${SIPDOMAIN})
exten => s,25,Noop(SIP_CODEC=${SIP_CODEC})
exten => s,26,Noop(SIPCALLID=${SIPCALLID})
exten => s,27,Noop(SIPUSERAGENT=${SIPUSERAGENT})
exten => s,29,Noop(TXTCIDNAME=${TXTCIDNAME})
exten => s,30,Noop(UNIQUEID=${UNIQUEID})
exten => s,31,Noop(TOUCH_MONITOR=${TOUCH_MONITOR})
exten => s,32,Noop(MACRO_CONTEXT=${MACRO_CONTEXT})
exten => s,33,Noop(MACRO_EXTEN=${MACRO_EXTEN})
exten => s,34,Noop(MACRO_PRIORITY=${MACRO_PRIORITY})

[macro-user-logon]
; check device type
;
exten => s,1,Set(DEVICETYPE=${DB(DEVICE/${CALLERID(number)}/type)})
exten => s,n,Answer()
exten => s,n,Wait(1)
exten => s,n,GotoIf($["${DEVICETYPE}" = “fixed”]?s-FIXED,1)
; get user’s extension
;
exten => s,n,Set(AMPUSER=${ARG1})
exten => s,n,GotoIf($["${AMPUSER}" != “”]?gotpass)
exten => s,n(playagain),Read(AMPUSER,please-enter-your-extension-then-press-pound,4)
; get user’s password and authenticate
;
exten => s,n,GotoIf($["${AMPUSER}" = “”]?s-MAXATTEMPTS,1)
exten => s,n(gotpass),GotoIf($["${DB_EXISTS(AMPUSER/${AMPUSER}/password)}" = “0”]?s-NOUSER,1)
exten => s,n,Set(AMPUSERPASS=${DB_RESULT})
exten => s,n,GotoIf($[${LEN(${AMPUSERPASS})} = 0]?s-NOPASSWORD,1)
; do not continue if the user has already logged onto this device
;
exten => s,n,Set(DEVICEUSER=${DB(DEVICE/${CALLERID(number)}/user)})
exten => s,n,GotoIf($["${DEVICEUSER}" = “${AMPUSER}”]?s-ALREADYLOGGEDON,1)
exten => s,n,Authenticate(${AMPUSERPASS})
exten => s,n,DeadAGI(user_login_out.agi,login,${CALLERID(number)},${AMPUSER})
exten => s,n,Playback(vm-goodbye)

exten => s-FIXED,1,NoOp(Device is FIXED and cannot be logged into)
exten => s-FIXED,n,Playback(ha/phone)
exten => s-FIXED,n,SayDigits(${CALLERID(number)})
exten => s-FIXED,n,Playback(is-curntly-unavail&vm-goodbye)
exten => s-FIXED,n,Hangup ;TODO should play msg indicated device cannot be logged into

exten => s-ALREADYLOGGEDON,1,NoOp(This device has already been logged into by this user)
exten => s-ALREADYLOGGEDON,n,Playback(vm-goodbye)
exten => s-ALREADYLOGGEDON,n,Hangup ;TODO should play msg indicated device is already logged into

exten => s-NOPASSWORD,1,NoOp(This extension does not exist or no password is set)
exten => s-NOPASSWORD,n,Playback(pbx-invalid)
exten => s-NOPASSWORD,n,Goto(s,playagain)

exten => s-MAXATTEMPTS,1,NoOp(Too many login attempts)
exten => s-MAXATTEMPTS,n,Playback(vm-goodbye)
exten => s-MAXATTEMPTS,n,Hangup

exten => s-NOUSER,1,NoOp(Invalid extension ${AMPUSER} entered)
exten => s-NOUSER,n,Playback(pbx-invalid)
exten => s-NOUSER,n,Goto(s,playagain)

[macro-user-logoff]
; check device type
;
exten => s,1,Set(DEVICETYPE=${DB(DEVICE/${CALLERID(number)}/type)})
exten => s,n,GotoIf($["${DEVICETYPE}" = “fixed”]?s-FIXED,1)
exten => s,n,DeadAGI(user_login_out.agi,logout,${CALLERID(number)})
exten => s,n(done),Playback(vm-goodbye)

exten => s-FIXED,1,NoOp(Device is FIXED and cannot be logged out of)
exten => s-FIXED,n,Playback(an-error-has-occured&vm-goodbye)
exten => s-FIXED,n,Hangup ;TODO should play msg indicated device cannot be logged into

; Privacy Manager Macro makes sure that any calls that don’t pass the privacy manager are presented
; with congestion since there have been observed cases of the call continuing if not stopped with a
; congestion, and this provides a slightly more friendly ‘sorry’ message in case the user is
; legitamately trying to be cooperative.
;
; Note: the following options are configurable in privacy.conf:
;
; maxretries = 3 ; default value, number of retries before failing
; minlength = 10 ; default value, number of digits to be accepted as valid CID
;
[macro-privacy-mgr]
exten => s,1,Set(KEEPCID=${CALLERID(num)})
exten => s,n,GotoIf($[“foo${CALLERID(num):0:1}”=“foo+”]?CIDTEST2:CIDTEST1)
exten => s,n(CIDTEST1),Set(TESTCID=${MATH(1+${CALLERID(num)})})
exten => s,n,Goto(TESTRESULT)
exten => s,n(CIDTEST2),Set(TESTCID=${MATH(1+${CALLERID(num):1})})
exten => s,n(TESTRESULT),GotoIf($[“foo${TESTCID}”=“foo”]?CLEARCID:PRIVMGR)
exten => s,n(CLEARCID),Set(CALLERID(num)=)
exten => s,n(PRIVMGR),PrivacyManager()
exten => s,n,GotoIf($["${PRIVACYMGRSTATUS}"=“FAILED”]?fail)
exten => s,n,SetCallerPres(allowed_passed_screen); stop gap until app_privacy.c clears unavailble bit

; Text-To-Speech related macros
; These all follow common actions. First try to playback a file “tts/custom-md5”
; where “md5” is the md5() of whatever is going to be played. If that doesn’t exist,
; try to playback using macro-tts-sayXXXXX (where XXXXX is text/digits/etc, same as
; the macro below). If that macro exits with MACRO_OFFSET=100, then it’s done,
; therwise, fallback to the default asterisk method.
;
; say text is purely for text-to-speech, there is no fallback
[macro-saytext]
exten => s,1,Noop(Trying custom SayText playback for “${ARG1}”)
exten => s,n,Playback(tts/custom-${MD5(${ARG1})})
exten => s,n,GotoIf($["${PLAYBACKSTATUS}"=“SUCCESS”]?done)
; call tts-saytext. This should set MACRO_OFFSET=101 if it was successful
exten => s,n(tts),Macro(tts-saytext,${ARG1},${ARG2},${ARG3})
exten => s,n,Noop(No text-to-speech handler for SayText, cannot say “${ARG1}”)
exten => s,n,Goto(done)
exten => s,tts+101,Noop(tts handled saytext)

; say name is for saying names typically, but fallsback to using SayAlpha
; (saying the word letter-by-letter)
[macro-sayname]
exten => s,1,Noop(Trying custom SayName playback for “${ARG1}”)
exten => s,n,Playback(tts/custom-${MD5(${ARG1})})
exten => s,n,GotoIf($["${PLAYBACKSTATUS}"=“SUCCESS”]?done)
; call tts-sayalpha. This should set MACRO_OFFSET=101 if it was successful
exten => s,n(tts),Macro(tts-sayalpha,${ARG1},${ARG2},${ARG3})
exten => s,n,SayAlpha(${ARG1})
exten => s,n,Goto(done)
exten => s,tts+101,Noop(tts handled sayname)

; Say number is for saying numbers (eg “one thousand forty six”)
[macro-saynumber]
exten => s,1,Noop(Trying custom SayNumber playback for “${ARG1}”)
exten => s,n,Playback(tts/custom-${MD5(${ARG1})})
exten => s,n,GotoIf($["${PLAYBACKSTATUS}"=“SUCCESS”]?done)
; call tts-saynumber. This should set MACRO_OFFSET=101 if it was successful
exten => s,n(tts),Macro(tts-saynumber,${ARG1},${ARG2},${ARG3})
exten => s,n,SayNumber(${ARG1})
exten => s,n,Goto(done)
exten => s,tts+101,Noop(tts handled saynumber)

; Say digits is for saying digits one-by-one (eg, “one zero four six”)
[macro-saydigits]
exten => s,1,Noop(Trying custom SayDigits playback for “${ARG1}”)
exten => s,n,Playback(tts/custom-${MD5(${ARG1})})
exten => s,n,GotoIf($["${PLAYBACKSTATUS}"=“SUCCESS”]?done)
; call tts-saydigits. This should set MACRO_OFFSET=101 if it was successful
exten => s,n(tts),Macro(tts-saydigits,${ARG1},${ARG2},${ARG3})
exten => s,n,SayDigits(${ARG1})
exten => s,n,Goto(done)

;
; ############################################################################
; Inbound Contexts [from]
; ############################################################################

[from-sip-external]
;give external sip users congestion and hangup
; Yes. This is really meant to be _. - I know asterisk whinges about it, but
; I do know what I’m doing. This is correct.
exten => _.,1,NoOp(Received incoming SIP connection from unknown peer to ${EXTEN})
exten => _.,n,Set(DID=${IF($["${EXTEN:1:2}"=""]?s:${EXTEN})})
exten => _.,n,Goto(s,1)
exten => s,1,GotoIf($["${ALLOW_SIP_ANON}"=“yes”]?from-trunk,${DID},1)
exten => s,n,Set(TIMEOUT(absolute)=15)
exten => s,n,Answer
exten => s,n,Wait(2)
exten => s,n,Playback(ss-noservice)
exten => s,n,Playtones(congestion)
exten => s,n,Congestion(5)
exten => h,1,NoOp(Hangup)
exten => i,1,NoOp(Invalid)
exten => t,1,NoOp(Timeout)

[from-internal-xfer]
; applications are now mostly all found in from-internal-additional in _custom.conf
include => from-internal-custom
include => parkedcalls
;allow phones to dial other extensions
include => ext-fax
;allow phones to access generated contexts
;
; MODIFIED (PL)
;
; Currently the include for findmefollow is being auto-generated before ext-local which is the desired behavior.
; However, I haven’t been able to do anything that I know of to force this. We need to determine if it should
; be hardcoded into here to make sure it doesn’t change with some configuration. For now I will leave it out
; until we can discuss this.
;
include => ext-local-confirm
include => findmefollow-ringallv2
include => from-internal-additional
; This causes grief with ‘#’ transfers, commenting out for the moment.
; include => bad-number
exten => s,1,Macro(hangupcall)
exten => h,1,Macro(hangupcall)

[from-internal]
include => from-internal-xfer
include => bad-number

;------------------------------------------------------------------------
; [macro-setmusic]
;------------------------------------------------------------------------
; CONTEXT: macro-setmusic
; PURPOSE: to turn off moh on routes where it is not desired
;
;------------------------------------------------------------------------
[macro-setmusic]
exten => s,1,NoOp(Setting Outbound Route MoH To: ${ARG1})
exten => s,2,SetMusicOnHold(${ARG1})
;------------------------------------------------------------------------

; ##########################################
; ## Ring Groups with Confirmation macros ##
; ##########################################
; Used by followme and ringgroups

;------------------------------------------------------------------------
; [macro-dial-confirm]
;------------------------------------------------------------------------
; This has now been incorporated into dialparties. It still only works with ringall
; and ringall-prim strategies. Have not investigated why it doesn’t work with
; hunt and memory hunt.
;
;------------------------------------------------------------------------
[macro-dial-confirm]
; This was written to make it easy to use macro-dial-confirm instead of macro-dial in generated dialplans.
; This takes the same paramaters, with an additional paramater of the ring group Number
; ARG1 is the timeout
; ARG2 is the DIAL_OPTIONS
; ARG3 is a list of xtns to call - 203-222-240-123123123#-211
; ARG4 is the ring group number

; This sets a unique value to indicate that the channel is ringing. This is used for warning slow
; users that the call has already been picked up.
;
exten => s,1,Set(DB(RG/${ARG4}/${CHANNEL})=RINGING)

; We need to keep that channel variable, because it’ll change when we do this dial, so set it to
; fallthrough to every sibling.
;
exten => s,n,Set(__UNIQCHAN=${CHANNEL})

; The calling ringgroup should have set RingGroupMethod appropriately. We need to set two
; additional parameters:
;
; USE_CONFIRMATION, RINGGROUP_INDEX
;
; Thse are passed to inform dialparties to place external calls through the [grps] context
;
exten => s,n,Set(USE_CONFIRMATION=TRUE)
exten => s,n,Set(RINGGROUP_INDEX=${ARG4})
exten => s,n,Set(ARG4=) ; otherwise it gets passed to dialparties.agi which processes it (prob bug)

exten => s,n,Macro(dial,${ARG1},${ARG2},${ARG3})

; delete the variable, if we are here, we are done trying to dial and it may have been left around
;
exten => s,n,DBDel(RG/${RINGGROUP_INDEX}/${CHANNEL})
exten => s,n,Set(USE_CONFIRMATION=)
exten => s,n,Set(RINGGROUP_INDEX=)
;------------------------------------------------------------------------

;------------------------------------------------------------------------
; [ext-local-confirm]
;------------------------------------------------------------------------
; If call confirm is being used in a ringgroup, then calls that do not require confirmation are sent
; to this extension instead of straight to the device.
;
; The sole purpose of sending them here is to make sure we run Macro(auto-confirm) if this
; extension answers the line. This takes care of clearing the database key that is used to inform
; other potential late comers that the extension has been answered by someone else.
;
; ALERT_INFO is deprecated in Asterisk 1.4 but still used throughout the FreePBX dialplan and
; usually set by dialparties.agi. This allows ineritance. Since no dialparties.agi here, set the
; header if it is set.
;
;------------------------------------------------------------------------
[ext-local-confirm]
exten => _LC-.,1,Noop(IN ext-local-confirm with - RT: ${RT}, RG_IDX: ${RG_IDX})
exten => _LC-.,n,GotoIf($[“x${ALERT_INFO}”=“x”]?godial)
exten => _LC-.,n,SIPAddHeader(Alert-Info: ${ALERT_INFO})
exten => _LC-.,n(godial),dial(${DB(DEVICE/${EXTEN:3}/dial)},${RT},M(auto-confirm^${RG_IDX})${DIAL_OPTIONS})

;------------------------------------------------------------------------
; [findmefollow-ringallv2]
;------------------------------------------------------------------------
; This context, to be included in from-internal, implements the PreRing part of findmefollow
; as well as the GroupRing part. It also communicates between the two so that if DND is set
; on the primary extension, and mastermode is enabled, then the other extensions will not ring
;
;------------------------------------------------------------------------
[findmefollow-ringallv2]
exten => _FMPR-.,1,Noop(In FMPR ${FMGRP} with ${EXTEN:5})
exten => _FMPR-.,n,Set(RingGroupMethod=)
exten => _FMPR-.,n,Set(USE_CONFIRMATION=)
exten => _FMPR-.,n,Set(RINGGROUP_INDEX=)
exten => _FMPR-.,n,Macro(simple-dial,${EXTEN:5},${FMREALPRERING})
exten => _FMPR-.,n,GotoIf($["${DIALSTATUS}" != “BUSY”]?nodnd)
exten => _FMPR-.,n,Set(DB(FM/DND/${FMGRP}/${FMUNIQUE})=DND)
exten => _FMPR-.,n(nodnd),Noop(Ending FMPR ${FMGRP} with ${EXTEN:5} and dialstatus ${DIALSTATUS})
exten => _FMPR-.,n,Hangup()

exten => _FMGL-.,1,Noop(In FMGL ${FMGRP} with ${EXTEN:5})
exten => _FMGL-.,n,GotoIf($["${DB(FM/DND/${FMGRP}/${FMUNIQUE})}" = “DND”]?dodnd)
exten => _FMGL-.,n,Wait(1)
exten => _FMGL-.,n,GotoIf($["${DB(FM/DND/${FMGRP}/${FMUNIQUE})}" = “DND”]?dodnd)
exten => _FMGL-.,n,Wait(1)
exten => _FMGL-.,n,GotoIf($["${DB(FM/DND/${FMGRP}/${FMUNIQUE})}" = “DND”]?dodnd)
exten => _FMGL-.,n,Wait(${FMPRERING})
exten => _FMGL-.,n,GotoIf($["${DB(FM/DND/${FMGRP}/${FMUNIQUE})}" = “DND”]?dodnd)
exten => _FMGL-.,n,DBDel(FM/DND/${FMGRP}/${FMUNIQUE})
exten => _FMGL-.,n(dodial),Macro(dial,${FMGRPTIME},${DIAL_OPTIONS},${EXTEN:5})
exten => _FMGL-.,n,Noop(Ending FMGL ${FMGRP} with ${EXTEN:5} and dialstatus ${DIALSTATUS})
exten => _FMGL-.,n,Hangup()
exten => _FMGL-.,n+10(dodnd),DBDel(FM/DND/${FMGRP}/${FMUNIQUE})
exten => _FMGL-.,n,GotoIf($["${FMPRIME}" = “FALSE”]?dodial)
exten => _FMGL-.,n,Noop(Got DND in FMGL ${FMGRP} with ${EXTEN:5} in ${RingGroupMethod} mode, aborting)
exten => _FMGL-.,n,Hangup()

;------------------------------------------------------------------------
; [block-cf]
;------------------------------------------------------------------------
; This context is set as a target with FORWARD_CONTEXT when Call Forwarding is set to be
; ignored in a ringgoup or other features that may take advantage of this. Server side
; CF is done in dialparties.agi but if a client device forwards a call, it will be caught
; and blocked here.
;------------------------------------------------------------------------
[block-cf]
exten => _X.,1,Noop(Blocking callforward to ${EXTEN} because CF is blocked)
exten => _X.,n,Hangup()

;------------------------------------------------------------------------

; ############################################################################
; Extension Contexts [ext]
; ############################################################################

[ext-fax]
exten => s,1,Answer
exten => s,2,Goto(in_fax,1)
exten => in_fax,1,StopPlayTones
exten => in_fax,2,GotoIf($["${FAX_RX}" = “system”]?3:analog_fax,1)
exten => in_fax,3,Macro(faxreceive)
exten => in_fax,4,Hangup
exten => analog_fax,1,GotoIf($["${FAX_RX}" = “disabled”]?4:2) ;if fax is disabled, just hang up
exten => analog_fax,2,Set(DIAL=${DB(DEVICE/${FAX_RX}/dial)});
exten => analog_fax,3,Dial(${DIAL},20,d)
exten => analog_fax,4,Hangup
;exten => out_fax,1,wait(7)
exten => out_fax,1,txfax(${TXFAX_NAME},caller)
exten => out_fax,2,Hangup
exten => h,1,system(/var/lib/asterisk/bin/fax-process.pl --to ${EMAILADDR} --from ${FAX_RX_FROM} --dest “${FROM_DID}” --subject “Fax from ${URIENCODE(${CALLERID(number)})} ${URIENCODE(${CALLERID(name)})}” --attachment fax_${URIENCODE(${CALLERID(number)})}.pdf --type application/pdf --file ${FAXFILE});
exten => h,2,Hangup()

;this is where parked calls go if they time-out. Should probably re-ring
[default]
include => ext-local
exten => s,1,Playback(vm-goodbye)
exten => s,2,Macro(hangupcall)

So who can help me with funkcion .conf files i send HIM 80 dollars or MORE

I am seerious Man.

Thanks for reply

Rudolf JurÄ