Hello! I'm implementing some VoIP system with SER+Asterisk. I've already implemented many features, but all of them was tested only in non-NAT environments. Now, at final-alpha stage (before starting massive beta tests) I found big (as for me) problem with NAT... Now I need a working example of ser.cfg (it will be really good) or, simply, to talk with someone, who realized such scheme... My current problem is with ua-to-ua calls between my users... I have a following call routing scheme... UA1->...->SER->Asterisk->SER->...->UA2... Description: UA1 calls to some number, ser routes this call to asterisk, asterisk resolves that number to some account (login name, that being used by users to login to ser) and makes call to sip/ser/user_name... and ser routes this call for user with such login... I have strange problems with natted UAs... Please, send me a working example of similiar schemes (ser+asterisk+nat)... or help me by a word ;-) Great thanks! -- /Scoundrel