Hi, I've recently encountered some problem with my SIP service whereby i call out to a specific number and i encounter a one way voice. If i'm the initiator, i cannot hear the other party but he can hear me. At first i thought it was a return route issue (as i'm going thru NAT) , so i switch my SIP to a public IP but i still face the same problem. Its really only that specific PSTN number that i have dialed facing this problem. The only difference that i can think of is that PSTN number is on a different route. I did a NGREP from my SIP server for the PSTN number that works (2-way voice) and the Number that doesn't work (1-way voice) . The only difference is there is an extra : NGW --> Proxy SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP Proxy --> SIP Device SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP for the PSTN number that works (the one with 2-way voice). Anyone has idea what does the Session Progress is for ? Or what problem am i facing ? Thanks a mILLION ! Regards, Sam _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers