provided by Star2Billing S.L.

Hi

i am trying to configure a calling card with a DID number with a2billing and trixbox.

the inbound call is transfer to a2billing then agi script start asking for the number to dial
but instead of dialing my trunk outbound i am getting an incoming call Loop Detected.

i do know how to change that.

could someone please tell how to fix it

thanks

i am using this context in trixbox
custom-a2billing,${EXTEN},1


here is my extensions_custom.conf

[macro-dialout-trunk-predial-hook]
exten => s,1,GotoIf($["${OUT_${DIAL_TRUNK}:4:4}" = "Local/"]?custom-freepbx-a2billing,${OUTNUM},1:2)
exten => s,2,MacroExit

[a2biling-did]
exten => _.,1,Goto(from-trunk,${CUT(CUT(SIP_HEADER(To),@,1),:,2)},1)

[a2billing]
exten => s,1,NoOp(A2Billing Start)
exten => s,n,Answer()
exten => s,n,Wait(2)
;exten => s,Goto(a2billing,${EXTEN},1)
exten => s,n,DeadAGI(a2billing.php,1,${EXTEN})
exten => s,n,Wait(2)
exten => s,n,Hangup

[custom-a2billing]
;for call through service
exten => s,1,NoOp(A2Billing Start)
exten => s,n,Answer()
exten => s,n,Wait(2)
;exten => s,Goto(a2billing,${EXTEN},1)
exten => s,n,DeadAGI(a2billing.php,1)
exten => s,n,Wait(2)
exten => s,n,Hangup



[custom-a2billing-did]
;To deliver DID and bill for them
exten => s,1,deadAGI(a2billing.php,1,did)
exten => s,2,Hangup

and here is my console log


== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
-- Executing [s@from-trunk:1] Set("SIP/callwithus-00000008", "__FROM_DID=s") in new stack
-- Executing [s@from-trunk:2] Gosub("SIP/callwithus-00000008", "app-blacklist-check,s,1") in new stack
-- Executing [s@app-blacklist-check:1] GotoIf("SIP/callwithus-00000008", "0?blacklisted") in new stack
-- Executing [s@app-blacklist-check:2] Return("SIP/callwithus-00000008", "") in new stack
-- Executing [s@from-trunk:3] ExecIf("SIP/callwithus-00000008", "0 ?Set(CALLERID(name)=33xxxx)") in new stack
-- Executing [s@from-trunk:4] Set("SIP/callwithus-00000008", "__CALLINGPRES_SV=allowed_not_screened") in new stack
-- Executing [s@from-trunk:5] Set("SIP/callwithus-00000008", "CALLERPRES()=allowed_not_screened") in new stack
-- Executing [s@from-trunk:6] Goto("SIP/callwithus-00000008", "custom-a2billing,s,1") in new stack
-- Goto (custom-a2billing,s,1)
-- Executing [s@custom-a2billing:1] NoOp("SIP/callwithus-00000008", "A2Billing Start") in new stack
-- Executing [s@custom-a2billing:2] Answer("SIP/callwithus-00000008", "") in new stack
-- Executing [s@custom-a2billing:3] Wait("SIP/callwithus-00000008", "2") in new stack
-- SIP/callwithus-00000007 answered SIP/10-00000006
-- Executing [s@custom-a2billing:4] DeadAGI("SIP/callwithus-00000008", "a2billing.php,1") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php
-- <SIP/callwithus-00000008> Playing 'prepaid-enter-pin-number.gsm' (language 'en')
-- Playing 'prepaid-auth-fail' (escape_digits=#) (sample_offset 0)
-- ast_get_srv: SRV lookup for '_sip._UDP.sip.callwithus.com' mapped to host sip.callwithus.com, port 5060
-- <SIP/callwithus-00000008> Playing 'prepaid-enter-pin-number.gsm' (language 'en')
-- <SIP/callwithus-00000008> Playing 'prepaid-enter-dest.gsm' (language 'fr')
-- Playing 'prepaid-you-have' (escape_digits=#) (sample_offset 0)
-- <SIP/callwithus-00000008> Playing 'digits/60.slin' (language 'fr')
-- <SIP/callwithus-00000008> Playing 'digits/6.slin' (language 'fr')
-- Playing 'prepaid-minutes' (escape_digits=#) (sample_offset 0)
-- AGI Script Executing Application: (DIAL) Options: (SIP/[email protected],,60,HRrL(3960000:61000:30000))
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
-- Called [email protected]
-- Got SIP response 482 "Loop Detected" back from 1x.xx.xx.xx
-- Now forwarding SIP/callwithus-00000008 to 'Local/005xxxxxxx@from-sip-external' (thanks to SIP/1x.xx.xx.xx-00000009)
-- Executing [005xxxxxx@from-sip-external:1] NoOp("Local/005xxxxxx@from-sip-external-cd50;2", "Received incoming SIP connection from unknown peer to 005xxxxxx") in new stack
-- Executing [005xxxxxxx@from-sip-external:2] Set("Local/005xxxxx@from-sip-external-cd50;2", "DID=005xxxxxx") in new stack
-- Executing [005xxxxxxx@from-sip-external:3] Goto("Local/005xxxxxx@from-sip-external-cd50;2", "s,1") in new stack
-- Goto (from-sip-external,s,1)
-- Executing [s@from-sip-external:1] GotoIf("Local/005xxxxxx@from-sip-external-cd50;2", "0?from-trunk,005xxxxxxxxx,1") in new stack
-- Executing [s@from-sip-external:2] Set("Local/005xxxxxxxx@from-sip-external-cd50;2", "TIMEOUT(absolute)=15") in new stack
Channel will hangup at 2013-03-20 12:38:53.000 AST.

What address is 1x.xx.xx.xx ?

(DIAL) Options: (SIP/[email protected],,60,HRrL

Is that the address of your a2billing server, or the address of your call provider?

Hi bucasia

What address is 1x.xx.xx.xx ?

(DIAL) Options: (SIP/[email protected],,60,HRrL

Is that the address of your a2billing server, or the address of your call provider?

1x.xx.xx.xx is my asterisk adr

thanks

I think you want it to be placing the call with your call provider?

What details do you have in the trunk set up in A2Billing. It should be your call provider details.

Hi bucasia

Yep , i changed it and now is working thank you

Py