> I have two SIP phones. one is 6002, w: > > which is behind a NAT, and another is 5009 which has public IP . > > When a call between 6002 to 5009, the 6002 cannot hear any from 5009, and > 5009 did hear from 6002. > > [6002] > type=friend > host=dynamic > nat=1 > qualify=yes > > [5009] > type=friend > host=dynamic > nat=1 > qualify=yes >
Add "canreinvite=no" to the [6002] config.. Personally I use "canreinvite=no" on all my internet based SIP UA's to avoid these type of issues, the disadvantage is that all traffic will go via the Asterisk server but if the choice is no voice or more traffic I think more traffic is acceptable.. :) Later.. -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
