Hi, I have the following dialplan. Problem is, if the user authenticates, * starts counting as billable seconds even if i hangup the phone before the called party answers..And also as disposition.. it accepts all calls authenticated as 'answered' If i commentout the authentication line everything works as it should be. How should i use authentication that, it should accept it as aswered by default. Here is my dialplan: [CallingRule_testcall] exten = _0XXXXXXXXXX,1,Authenticate(/etc/asterisk/passwords,an) exten = _0XXXXXXXXXX,n,Set(CALLERID(num)=0312290${CALLERID(num)}) exten = _0XXXXXXXXXX,n,Macro(trunkdial-failover-0.3,${test}/${EXTEN:0},${span_1}/9${EXTEN:0},test,span_1)
And here is the debug on ast. cli Using UDPTL CoS mark 5 -- Executing [051111...@dlpn_test:1] Authenticate("SIP/8000-b51e04e0", "/etc/asterisk/passwords,an") in new stack -- <SIP/8000-b51e04e0> Playing 'agent-pass.ulaw' (language 'de') -- <SIP/8000-b51e04e0> Playing 'auth-thankyou.ulaw' (language 'de') -- Executing [0511111...@dlpn_test:2] Set("SIP/8000-b51e04e0", "CALLERID(num)=03122901111") in new stack -- Executing [0511111...@dlpn_test:3] Macro("SIP/8000-b51e04e0", "trunkdial-failover-0.4,SIP/test/051111111,DAHDI/g1/90531111111,test,span_1") in new stack -- Executing [...@macro-trunkdial-failover-0.4:1] GotoIf("SIP/8000-b51e04e0", "1?1-dial,1") in new stack -- Goto (macro-trunkdial-failover-0.4,1-dial,1) -- Executing [1-d...@macro-trunkdial-failover-0.4:1] Dial("SIP/8000-b51e04e0", "SIP/test/0511111111") in new stack == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 == Using UDPTL CoS mark 5 -- Called test/05323939196 -- SIP/Superonline-b474f218 is making progress passing it to SIP/8000-b51e04e0 -- SIP/Superonline-b474f218 is making progress passing it to SIP/8000-b51e04e0 == Spawn extension (macro-trunkdial-failover-0.4, 1-dial, 1) exited non-zero on 'SIP/8000-b51e04e0' in macro 'trunkdial-failover-0.4' == Spawn extension (DLPN_test, 0511111111, 3) exited non-zero on 'SIP/8000-b51e04e0' _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users