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- Apr 26, 2011
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Dear All,
I tried to integrate FreePBX with A2Billing. I’m just trying to allow calls to be passed from FreePBX to A2Billing.
I followed many tutorial such as :
http://sysadminman.net/blog/2009/integr ... illing-621
But nothing happens and the calls are dropped.
Also I'm getting the following logs:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
localhost*CLI> -- Executing [s@from-trunk:1] Set("SIP/inbound_route-00000006", "__FROM_DID=s") in new stack
localhost*CLI> -- Executing [s@from-trunk:2] Gosub("SIP/inbound_route-00000006", "app-blacklist-check,s,1") in new stack
localhost*CLI> -- Executing [s@app-blacklist-check:1] GotoIf("SIP/inbound_route-00000006", "0?blacklisted") in new stack
localhost*CLI> -- Executing [s@app-blacklist-check:2] Set("SIP/inbound_route-00000006", "CALLED_BLACKLIST=1") in new stack
localhost*CLI> -- Executing [s@app-blacklist-check:3] Return("SIP/inbound_route-00000006", "") in new stack
localhost*CLI> -- Executing [s@from-trunk:3] ExecIf("SIP/inbound_route-00000006", "0 ?Set(CALLERID(name)=336XXXXXXXX)") in new stack
localhost*CLI> -- Executing [s@from-trunk:4] Set("SIP/inbound_route-00000006", "__CALLINGPRES_SV=allowed_not_screened") in new stack
localhost*CLI> -- Executing [s@from-trunk:5] Set("SIP/inbound_route-00000006", "CALLERPRES()=allowed_not_screened") in new stack
localhost*CLI> -- Executing [s@from-trunk:6] Goto("SIP/inbound_route-00000006", "a2billing,s,1") in new stack
localhost*CLI> -- Goto (a2billing,s,1)
localhost*CLI>
I have this in my extensions_custom.conf :
[a2billing]
exten => _X.,1,Answer
exten => _X.,n,Wait(1)
exten => _X.,n,DeadAGI(a2billing.php,1)
exten => _X.,n,Hangup
System info:
Elastix 2.4.0
freePBX 2.8.1
asterisk 1.8.21.0
Please help . Thank you ahead for your support!
Cheers.
Mickael
I tried to integrate FreePBX with A2Billing. I’m just trying to allow calls to be passed from FreePBX to A2Billing.
I followed many tutorial such as :
http://sysadminman.net/blog/2009/integr ... illing-621
But nothing happens and the calls are dropped.
Also I'm getting the following logs:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
localhost*CLI> -- Executing [s@from-trunk:1] Set("SIP/inbound_route-00000006", "__FROM_DID=s") in new stack
localhost*CLI> -- Executing [s@from-trunk:2] Gosub("SIP/inbound_route-00000006", "app-blacklist-check,s,1") in new stack
localhost*CLI> -- Executing [s@app-blacklist-check:1] GotoIf("SIP/inbound_route-00000006", "0?blacklisted") in new stack
localhost*CLI> -- Executing [s@app-blacklist-check:2] Set("SIP/inbound_route-00000006", "CALLED_BLACKLIST=1") in new stack
localhost*CLI> -- Executing [s@app-blacklist-check:3] Return("SIP/inbound_route-00000006", "") in new stack
localhost*CLI> -- Executing [s@from-trunk:3] ExecIf("SIP/inbound_route-00000006", "0 ?Set(CALLERID(name)=336XXXXXXXX)") in new stack
localhost*CLI> -- Executing [s@from-trunk:4] Set("SIP/inbound_route-00000006", "__CALLINGPRES_SV=allowed_not_screened") in new stack
localhost*CLI> -- Executing [s@from-trunk:5] Set("SIP/inbound_route-00000006", "CALLERPRES()=allowed_not_screened") in new stack
localhost*CLI> -- Executing [s@from-trunk:6] Goto("SIP/inbound_route-00000006", "a2billing,s,1") in new stack
localhost*CLI> -- Goto (a2billing,s,1)
localhost*CLI>
I have this in my extensions_custom.conf :
[a2billing]
exten => _X.,1,Answer
exten => _X.,n,Wait(1)
exten => _X.,n,DeadAGI(a2billing.php,1)
exten => _X.,n,Hangup
System info:
Elastix 2.4.0
freePBX 2.8.1
asterisk 1.8.21.0
Please help . Thank you ahead for your support!
Cheers.
Mickael