[asterisk-users] Session Progress

[michel freiha]

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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