Dear All
I m using brooktrout’s TR1500 telephony board to receive PSTN (PRI) calls, TR1500 generated these calls to any IP with RTP streams, how could I receive these call at my Asterisk server. Each call has different port and it uses g711 codec. Upon changing port in “bindport=” which come with RTP streams, “Error is Cseq Error, Dropping SIP message, its incomplete”
Please help to get rid of this problem
Aamir