Hi all, Today i try to use asterisk to make SIP call between two office A and B. At the office A, i use testA@myasteriskdomain.com. testA is softphone (for testing, i use sjphone) which is running in PC with IP: 192.168.4.100<http://192.168.4.100> . At the office B, i use testB@myasteriskdomain.com. testB is softphone (for testing, i use sjphone) which is running in PC with IP: 192.168.0.100<http://192.168.0.100> . Now from office A, testA can register with my server Asterisk and test B can also register with my server Asterisk.Now from testA, i make a call to test B. 1) Test A ----------send INVITE------------>Asterisk 2) Test A<-----------send Trying-------------Asterisk 3) Asterisk---------send INVITE------------->TestB 4) Asterisk<-------100 Trying-----------------TestB 5) Asterisk<-------180 Ringing---------------TestB 6) TestA<---------180Ringing------------Asterisk Now in test B, i accept the call, then 7) Asterisk<-------200 OK ---------------------TestB 8) TestA<--------200 OK------------------Asterisk 9) TestA------------------------ACK--------Asterisk------------------------------------------->TestB 10) TestA--------------------------------RTP stream--------------------------------------------TestB Here the problem begins, i talk and i hear anything. I see in my Asterisk. I see that when Asterisk receive a packet RTP from TestA, it forward immediately to IP adress of TestB, because TestB is behind a server. So IP adress of TestB is invisible from the world. Then, i can't hear anything. Can you please share your experience with me in this problem? Thank you so much. Julien -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051022/dcd8a174/attachment.htm