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