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<DIV><FONT face=Arial size=2>hello, using Asterisk, is there any clever way to
provide answer supervision based upon the received audio only from the FXO
interface (from a public PSTN switch that does not have battery reversal, or
CPC).</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>I would like to use something like:</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>[toCALLOUT<BR>;script to call a particular number
regardless, but we want to terminate if call is not answered</FONT></DIV>
<DIV><FONT face=Arial size=2>;we do NOT have Battery/Polarity reversal & we
do NOT have CPC from the FXO port.</FONT></DIV>
<DIV><FONT face=Arial size=2>;Zap/3 is a FXO port connected to PSTN</FONT></DIV>
<DIV><FONT face=Arial size=2>exten => s,1,Answer<BR>exten =>
s,n,Playback(beep)<BR>exten => s,n,Playback(beep)<BR>exten =>
s,n,Playback(pls-wait-connect-call) ; nice of us don't you think?<BR>exten =>
s,n,ChanIsAvail(Zap/3)<BR>exten => s,n,Cut(theChannel=AVAILCHAN,,1)<BR>exten
=> s,n,SetAccount(007)<BR>exten => s,n,Dial(${theChannel}/w1234567)
<------------------ we need to fork a process here to monitor
timeout<BR>exten => s,n,Hangup<BR></FONT></DIV>
<DIV><FONT face=Arial size=2>* if 15-30 seconds go by and the call is not
answered, how can we determine the call progress ? Note this is not a US network
we are using. Can we utilise answer supervision based completely on received
audio (like voice speech detection) ?</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>-samudra</FONT></DIV>
<DIV><FONT face=Arial size=2> </DIV></FONT></BODY></HTML>