Hi, I have a remote extension that goes through a router and internet to our elastix server which is NAtted behind a gateway with fixed IP public address and ports 5060-5065 and 10000-10500 redirected to elastix. The extension works fine, can dial to other extensions and also receive calls with no problems. It also dials out through a BRI trunk (ISDN) with no problems and audio works fine. What does not work is dialling out through a SIP trunk. When dialling, the call is generated and the other end can take the call but no audio either way goes through. I have tried different options to discard elements: - Remote extension with a PAP2 adapter and a Siemens Gigaset C470IP phone. They both behave exactly the same. - Different SIP trunks with different providers. They all behave the same. Of course this trunks work fine with all the internal extensions. Extension has of course canreinvite=no, qualify=yes etc. Phone port settings match those in Elastix and the gateway. Externip, NAT enabled etc. are all set in Elastix (Otherwise the remote extension would not work at all). The SIP trace does not reveal anything clear: This is driving me crazy and do not know what else to try. Any ideas? Can you help?