Dear All, I would like to ask please about how to fix the problem of sending fake ring back tone by asteriskserver when trying to make a call from an extension registered on asterisk to any PSTN number...I made some comparaison between calls made through Asterisk server that generate a fake ring back tone during dialing and other sip proxies that does not generate such tone...I noticed that Asterisk server is not sending back "183 Session Progress" or "Ringing" Sip packets after "Trying" to the Softphone so I think that's why the Softphone begin sending RTP packets to the Asterisk that generate ring back tone...
Please let me know if what I said is correct and how i can fix it. Regards
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