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 --------------------------------- Start your day with Yahoo! - make it your home page -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050719/2be2c3ad/attachment.htm
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 --------------------------------- Start your day with Yahoo! - make it your home page -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050719/ae07cc61/attachment.htm