<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><p>
<hr size=1> <a href="http://us.rd.yahoo.com/evt=34442/*http://www.yahoo.com/r/hs">Start your day with Yahoo! - make it your home page </a>