Calls being billed before called party answers!


Hello All,

I am using Asterisk to terminate calls to T1 trunk but as soon as the call hits the Asterisk box the call shows are connected before dialling out the Zap trunk which is to do with Answer Supervision I think.

The call comes from Customer >>> my PortaSip Server >> my Asterisk (TrixBox) >> T1 PSTN Trunk.

Can anyone tell me how to correct this problem? Here is the dial plan below:

exten => 1., 1, Dial(Zap/g0/${EXTEN})
exten => 1., 2, Hangup()




If you want to do anything other than a PBX then trix is not realy a good option, It will be answereing calls to listen for faxtone etc

Post the cli output of a call and maybe we can see whats happening


Hi Ian,

Here is the CLI info:

Connected to Asterisk 1.2.13 svn rev 47264 currently running on asterisk1 (pid = 3745)
Verbosity is at least 6
– Executing Progress(“SIP/cguanas2-b7a2d088”, “”) in new stack
– Executing Dial(“SIP/cguanas2-b7a2d088”, “ZAP/g0/5551234”) in new stack
– Called g0/5551234
– Zap/1-1 answered SIP/cguanas2-b7a2d088
– Hungup ‘Zap/1-1’
== Spawn extension (from-trunk, 17185551234, 2) exited non-zero on ‘SIP/cguanas2-b7a2d088’
– Executing Macro(“SIP/cguanas2-b7a2d088”, “hangupcall”) in new stack
– Executing ResetCDR(“SIP/cguanas2-b7a2d088”, “w”) in new stack
– Executing NoCDR(“SIP/cguanas2-b7a2d088”, “”) in new stack
– Executing GotoIf(“SIP/cguanas2-b7a2d088”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,6)
– Executing GotoIf(“SIP/cguanas2-b7a2d088”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing Wait(“SIP/cguanas2-b7a2d088”, “5”) in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on ‘SIP/cguanas2-b7a2d088’ in macro ‘hangupcall’
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on ‘SIP/cguanas2-b7a2d088’

Here is the Dial Plan

exten => _1718555xxxx,1,Dial(ZAP/g0/${EXTEN:4})
exten => _1718555xxxx,2,Hangup

When I am watching the CLI as soon as the call hits the TrixBox my phone shows that the call is connected even before Asterisk has dialled out to the PSTN!


exactly the same behviour with outgoing calls trhough a tdm04b with 4 fxo ports. the call is shown as “called g1/xxxxxxxxx” and then “zap/1-1 answered SIP/102-xxxxxx” before the call is answered by the other part!!!

how can it be???

I am using asterisknow beta5

best regards, paketecuento

Analog ports can not determine if a call is answered cause it is analog. Once you dial out via an FXO port, it will be considered as answered.

umhhhh, but what about “answeronpolarityswitch”??? if I have a polarity reverse, I know the other part has answered!!!

for me, this parameter does not work, as expected because the zap channel goes Up before this polarity reversal is done…
I’m running asterisknow beta5