Sangoma A200 & UK BT PSTN Multiline

I am new at FreePBX/Asterisk… I have a few PBXs out that are purely VoIP with no problems, this is the first one that I have tried to setup with a PSTN line of any description…

About the system:
It is a Fujitsu TX1310 Server with a Fresh Install of FreePBX 12 and a Sangoma A200 with a UK BT PSTN Multiline,

The Problem:
The Multiline is not detected as ringing until after the caller has hung up/timed out… once the caller hangs up asterisk directs the call as expected but obviously the caller is not there, I have a Standard Single PSTN Line also available (will be used for the FAX eventually) and when that is connected to the card it works perfectly. the problem only exists when the Multiline is connected.

What I have done so far:
Connected normal phone to the lines and checked incoming and outgoing calls…all fine
Spoken to Sangoma who have said that there is nothing wrong with the card/module they have even remoted in and looked…
Spoke to BT and asked them to increase Disconnect Time to 800ms
Tried various different configurations I have found…
Wen watching the Voltage on the Sangoma card it fluctuates as is suppose but doesn’t answer, once caller hangs up the card is “connected” voltage is dropped to 7 which I am told is correct…
BT have attended to look at lines and have confirmed that there is no fault on either line

In terms of the Line:
a BT Multiline PSTN, basically two separate lines, two end connections (sockets), one number, if a caller dials in the BT Exchange will try the first Line if “busy” will go to the second if that’s “busy” will be engaged to caller.

If you need any more information please ask, I am new to all this but I can work my way around using putty and WinSCP etc…

Many many thanks… I am at a loss… kind of thinking that if there is no way of getting this done I might have to try a gateway/PSTN adapter to then SIP the call to Asterisk/FreePBX

Thanks again

I did try adding the Config Files as copy and paste from the Text File but it would not let me “new user cannot add links” I was copying and pasting from a text file

Config Files:

;--------------------------------------------------------------------------------;
;          Do NOT edit this file as it is auto-generated by FreePBX.             ;
;--------------------------------------------------------------------------------;
; For information on adding additional paramaters to this file, please visit the ;
; FreePBX.org wiki page, or ask on IRC. This file was created by the new FreePBX ;
; BMO - Big Module Object. Any similarity in naming with BMO from Adventure Time ;
; is totally deliberate.                                                         ;
;--------------------------------------------------------------------------------;
[general]

; generated by module
#include chan_dahdi_general.conf

; for user additions not provided by module
#include chan_dahdi_general_custom.conf

[channels]
language=en
busydetect=yes
busycount=6
usecallerid=no
callwaiting=yes
usecallingpres=no
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=no
immediate=no
faxdetect=no
rxgain=10.0
txgain=10.0
cidsignalling=v23
module_name=wctdm24xxp
cidstart=polarity
usedistinctiveringdetection=yes
progzone=uk
callprogress=yes
hanguponpolarityswitch=yes
answeronpolarityswitch=yes
signalling=fxs_ks
opermode_checkbox=on
opermode=uk
alawoverride_checkbox=0
alawoverride=0
fxs_honor_mode_checkbox=on
fxs_honor_mode=0

; for user additions not provided by module
#include chan_dahdi_channels_custom.conf

; include dahdi groups defined by DAHDI module of FreePBX
#include chan_dahdi_groups.conf

; include dahdi extensions defined in FreePBX
#include chan_dahdi_additional.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: http://freepbx.org/configuration_files       ;
;--------------------------------------------------------------------------------;

;*******************************************************************************
; AUTO-GENERATED AND CUSOTM USER DIALPLAN INCLUDED HERE                        *
;*******************************************************************************

;--------------------------------------------------------------------------------;
; 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.
;
; WARNING ABOUT: #include extensions_override_freepbx.conf
;
;  This include file is put first to allow the auto-generated dialplan in FreePBX
;  to be overwritten if necessary. Overriding auto-generated dialplan should be done
;  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 situations where you need to modify the existing dialplan or macros, put it
;  here.
;
#include extensions_override_freepbx.conf
#include extensions_additional.conf
#include extensions_custom.conf
;--------------------------------------------------------------------------------;



;*******************************************************************************
; DIALPLAN OPEN TO THE PUBLIC, USING INBOUND DIDS AND SIMILAR                  *
;*******************************************************************************
;

;-------------------------------------------------------------------------------
; from-digital:
;
; Context to set for PRI's and equivalent
;
[from-digital]
include => from-pstn
;-------------------------------------------------------------------------------

;-------------------------------------------------------------------------------
; from-analog:
;
; Context to set for analog DAHDi cards and equivalent
;
[from-analog]
include => from-dahdi
;-------------------------------------------------------------------------------

;-------------------------------------------------------------------------------
; from-trunk:
;
; Context is really just an aliax of from-pstn
;
[from-trunk]
include => from-pstn
;-------------------------------------------------------------------------------

;-------------------------------------------------------------------------------
; from-pstn:
;
; Entry context for calls from the outside world to hit FreePBX
[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
include => ext-did-catchall		; THIS MUST COME AFTER ext-did
;-------------------------------------------------------------------------------

;-------------------------------------------------------------------------------
; from-pstn-e164-us:
;
; The context is designed for providers who send calls in e164 format and is
; biased towards NPA calls, callerid and dialing rules. It will do the following:
;
;  DIDs in an NPA e164 format of +1NXXNXXXXXX will be converted to 10 digit DIDs
;
;  DIDs in any other format will be delivered as they are, including e164 non NPA
;  DIDs which means they will need the full format including the + in the inbound
;  route.
;
;  CallerID(number) presented in e164 NPA format will be trimmed to a 10 digit CID
;
;  CallerID(number) presented in e164 non-NPA (country code other than 1) will be
;  reformated from: +<CountryCode><Number> to 011<CountryCode><Number>
;
[from-pstn-e164-us]
exten => _+1NXXNXXXXXX/_+1NXXNXXXXXX,1,Set(CALLERID(number)=${CALLERID(number):2})
exten => _+1NXXNXXXXXX/_NXXNXXXXXX,2,Goto(from-pstn,${EXTEN:2},1)
exten => _+1NXXNXXXXXX/_+NX.,1,Set(CALLERID(number)=011${CALLERID(number):1})
exten => _+1NXXNXXXXXX/_011NX.,n,Goto(from-pstn,${EXTEN:2},1)
exten => _+1NXXNXXXXXX,1,Goto(from-pstn,${EXTEN:2},1)
exten => _[0-9+]./_+1NXXNXXXXXX,1,Set(CALLERID(number)=${CALLERID(number):2})
exten => _[0-9+]./_1NXXNXXXXXX,1,Set(CALLERID(number)=${CALLERID(number):1})
exten => _[0-9+]./_NXXNXXXXXX,n,Goto(from-pstn,${EXTEN},1)
exten => _[0-9+]./_+NX.,1,Set(CALLERID(number)=011${CALLERID(number):1})
exten => _[0-9+]./_011NX.,n,Goto(from-pstn,${EXTEN},1)
exten => _[0-9+].,1,Goto(from-pstn,${EXTEN},1)
exten => s/_+1NXXNXXXXXX,1,Set(CALLERID(number)=${CALLERID(number):2})
exten => s/_NXXNXXXXXX,n,Goto(from-pstn,${EXTEN},1)
exten => s/_+NX.,1,Set(CALLERID(number)=011${CALLERID(number):1})
exten => s/_011NX.,n,Goto(from-pstn,${EXTEN},1)
exten => s,1,Goto(from-pstn,${EXTEN},1)
;-------------------------------------------------------------------------------

;-------------------------------------------------------------------------------
; from-pstn-to-did:
;
; The context is designed for providers who send the DID in the TO: SIP header
; only. The format of this header is:
;
; To: <sip:[email protected]>
;
; So the DID must be extracted between the sip: and the @, which this does
;
[from-pstn-toheader]
exten => _.,1,Goto(from-pstn,${CUT(CUT(SIP_HEADER(To),@,1),:,2)},1)
;-------------------------------------------------------------------------------

;-------------------------------------------------------------------------------
; from-sip-external
;
; This context is the default SIP context unless otherwise changed in the SIP
; Settings module or other sip configuration locations. This context is hit by
; either anonymous SIP calls or mis-configured SIP trunks when the incoming call
; can not be matched with a SIP section.
;
[from-sip-external]
exten => _.,1,NoOp(Received incoming SIP connection from unknown peer to ${EXTEN})
exten => _.,n,Set(DID=${IF($["${EXTEN:1:2}"=""]?s:${EXTEN})})
exten => _.,n,Goto(s,1)
exten => s,1,GotoIf($["${ALLOW_SIP_ANON}"="yes"]?checklang:noanonymous)
exten => s,n(checklang),GotoIf($["${SIPLANG}"!=""]?setlanguage:from-trunk,${DID},1)
exten => s,n(setlanguage),Set(CHANNEL(language)=${SIPLANG})
exten => s,n,Goto(from-trunk,${DID},1)
exten => s,n(noanonymous),Set(TIMEOUT(absolute)=15)
exten => s,n,Log(WARNING,"Rejecting unknown SIP connection from ${CHANNEL(recvip)}")
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,Hangup
exten => i,1,Hangup
exten => t,1,Hangup
;-------------------------------------------------------------------------------



;*******************************************************************************
; INTERNAL DIALPLAN, NOT OPEN TO THE PUBLIC WORLD                              *
;*******************************************************************************
;

;-------------------------------------------------------------------------------
; from-internal:
;
; Internal dialplan that most internal phones have access to
;
[from-internal]
include => from-internal-noxfer
include => from-internal-xfer
include => bad-number ; auto-generated
;-------------------------------------------------------------------------------

;-------------------------------------------------------------------------------
; from-internal-noxfer:
;
; Place to put internal dialplan that should not be accessible during a blind
; transfer, this context will not be visible during such.
;
[from-internal-noxfer]
include => from-internal-noxfer-custom
include => from-internal-noxfer-additional ; auto-generated
;-------------------------------------------------------------------------------

;-------------------------------------------------------------------------------
; from-internal-xfer:
;
; Place to put most internal dialplan, will be visible during normal calls and
; blind transfers.
;
[from-internal-xfer]
include => from-internal-custom
include => from-internal-additional ; auto-generated
exten => s,1,Macro(hangupcall)
exten => h,1,Macro(hangupcall)
;-------------------------------------------------------------------------------



;*******************************************************************************
; INTERNAL MACROS & SPECIAL CONTEXTS                                           *
;*******************************************************************************
;

;-------------------------------------------------------------------------------
; from-did-direct:
;
; forces ext-findmefollow to take precedence over ext-local. Also exposed to
; the public side to allow an extension number to be used as an external DID
; without requiring inbound routes to be created, common in many PRI installations
; where the last 4 digits are used as the extnension and DIDs are delivered in
; 4 digit formats.
;
[from-did-direct]
include => ext-findmefollow
include => ext-local
;-------------------------------------------------------------------------------

;-------------------------------------------------------------------------------
; macro-dial:
;
; 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,Set(CHANNEL(musicclass)=${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}" != "" & ("${DIALSTATUS}" = "TORTURE" | "${DIALSTATUS}" = "DONTCALL"))  | "${DIALSTATUS}" = "ANSWER"]?${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}}${ds})
exten => s,n,GotoIf($["${DIALSTATUS}" = "ANSWER"]?ANSWER,1)
exten => s,n,Set(HuntLoop=$[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),Noop(Deleting: CALLTRACE/${CT_EXTEN} ${DB_DELETE(CALLTRACE/${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)
exten => ANSWER,1,Noop(Call successfully answered - Hanging up now)
exten => ANSWER,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)
;-------------------------------------------------------------------------------

;-------------------------------------------------------------------------------
; macro-get-vmcontext:
;
; returns the user's voicemail context
;
[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()
;-------------------------------------------------------------------------------

;-------------------------------------------------------------------------------
; macro-outisbusy:
;
; called if all trunks are busy.  This macro is overwridden by the
; Outbound Route Messages module if installed and configured to provide
; better messages or choices of tones.
;
[macro-outisbusy]
exten => s,1,Progress
exten => s,n,Playback(all-circuits-busy-now,noanswer)
exten => s,n,Playback(pls-try-call-later,noanswer)
exten => s,n,Macro(hangupcall)
;-------------------------------------------------------------------------------

;-------------------------------------------------------------------------------
; macro-dialout-trunk-predial-hook:
;
; this macro intentionally left blank so it may be safely overwritten for any custom
; requirements that an installation may have.
;
; the macro is called by macro-dialout-trunk just prior to making a Dial() attempt
; to a trunk.
;
; MACRO RETURN CODE: ${PREDIAL_HOOK_RET}
;                    if set to "BYPASS" then this trunk will be skipped
;
;
[macro-dialout-trunk-predial-hook]
exten => s,1,MacroExit()
;-------------------------------------------------------------------------------

;-------------------------------------------------------------------------------
; macro-dialout-one-predial-hook
;
; this macro intentionally left blank so it may be safely overwritten for any custom
; requirements that an installation may have.
;
; the macro is called by macro-dialout-trunk just prior to making a Dial() attempt
; to an individual extension.
;
[macro-dialout-one-predial-hook]
exten => s,1,MacroExit()
;-------------------------------------------------------------------------------

;-------------------------------------------------------------------------------
; macro-dialout-dundi-predial-hook:
;
; this macro intentionally left blank so it may be safely overwritten for any custom
; requirements that an installation may have.
;
; the macro is called by macro-dialout-dundi just prior to making a Dial() attempt
; to a trunk.
;
; MACRO RETURN CODE: ${PREDIAL_HOOK_RET}
;                    if set to "BYPASS" then this trunk will be skipped
;
;
[macro-dialout-dundi-predial-hook]
exten => s,1,MacroExit()
;-------------------------------------------------------------------------------

;-------------------------------------------------------------------------------
; macro-user-logon:
;
; Used to log a user onto an adhoc device. Most of the work is done by
; user_login_out.agi AGI script
;
[macro-user-logon]
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,AGI(user_login_out.agi,login,${CALLERID(number)},${AMPUSER})
exten => s,n,Playback(agent-loginok)

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:
;
; Used to log a user off of an adhoc device. Most of the work is done by
; user_login_out.agi AGI script
;
[macro-user-logoff]
exten => s,1,Set(DEVICETYPE=${DB(DEVICE/${CALLERID(number)}/type)})
exten => s,n,GotoIf($["${DEVICETYPE}" = "fixed"]?s-FIXED,1)
exten => s,n,AGI(user_login_out.agi,logout,${CALLERID(number)})
exten => s,n(done),Playback(agent-loggedoff)

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

;-------------------------------------------------------------------------------
; default
;
; FreePBX does not use the default context. This context is used by asterisk when
; it has no other information provided and needs to deliver a call. Hitting this means
; there has been some sort of configuration error, or a potential bug somehwere.
; This context can be reached from either internal or external sources.
;
[default]
include => ext-local
exten => s,1,Playback(vm-goodbye)
exten => s,n,Noop(ERROR: FreePBX Does not use the [default] context, confguration error)
exten => s,n,Macro(hangupcall)
;-------------------------------------------------------------------------------



;*******************************************************************************
; DEPRECATED DIALPLAN - THESE WILL BE REMOVED, NOT USED BY FREEPBX             *
;*******************************************************************************
;

;-------------------------------------------------------------------------------
; macro-fixcid: [DEPRECATED]
;
; For some reason, if I don't run setCIDname, CALLERID(name) will be blank in my AGI
;
[macro-fixcid]
exten => s,1,Set(CALLERID(name)=${CALLERID(name)})
;-------------------------------------------------------------------------------


;-------------------------------------------------------------------------------
; macro-sayXXXX: [DEPRECATED]
;
; 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,
; otherwise, fallback to the default asterisk method.
;

;-------------------------------------------------------------------------------
; macro-saytext: [DEPRECATED]
;
; 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)
;-------------------------------------------------------------------------------

;-------------------------------------------------------------------------------
; macro-sayname: [DEPRECATED]
;
; 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)
;-------------------------------------------------------------------------------

;-------------------------------------------------------------------------------
; macro-saynumber: [DEPRECATED]
;
; 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)
;-------------------------------------------------------------------------------

;-------------------------------------------------------------------------------
; macro-saydigits: [DEPRECATED]
;
; 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)
;-------------------------------------------------------------------------------

I’m in a similar position. I have a single line and a multiline PSTN, so in effect 3 lines with 2 numbers in total. I’ve got everything working on the single line, it’s been up for quite some time but now I need to get the other two working.

How does FreePBX pick it’s lines? Is it in sequence? Our single line isn’t to be used for outgoing calls as the number isn’t advertised and we tend to us that line for engineers to call in on.

Can anyone provide any pointers? I’ve got myself all confused hehe! Please bear in mind I am only a beginner!