<DIV>
<DIV>Hi</DIV>
<DIV> </DIV>
<DIV>I am doing</DIV>
<DIV> </DIV>
<DIV>PSTN -> Asterisk -> SIP -> yate -> H323 -> Telco</DIV>
<DIV> </DIV>
<DIV>When user intiate a call from asterisk, it is pass to yate for SIP - H323 signalling, </DIV>
<DIV>and forward the calls to telco. Everything is fine there.</DIV>
<DIV> </DIV>
<DIV>My problem is, i am not getting an actual PSTN ringing tone.</DIV>
<DIV>instead i am getting a fake tone and anypart of the world i call is the same</DIV>
<DIV>ringing tone and even if the phone is busy, it keeps ringing until i hang up.</DIV>
<DIV> </DIV>
<DIV>telco claim that asterisk is not requesting for the tone</DIV>
<DIV> </DIV>
<DIV>i am reading a "180 Ringing: from SIP messages</DIV>
<DIV> </DIV>
<DIV>I believe there is something like this in the previous post but was</DIV>
<DIV>unanswered.</DIV>
<DIV> </DIV>
<DIV>I do not have a "r" in my dial command and i am not doing callprocess either</DIV>
<DIV> </DIV>
<DIV>any help is highly appreciated</DIV>
<DIV> </DIV>
<DIV>Thank You</DIV></DIV><p>
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