Hi all! I'm having a real headache with an external SIP account on an Aastra phone. I managed to connect the telephone to the PBX but still I can't get any audio on both ends. I can even monitor other lines through the programmable keypad. Additionally to that, when I call a mobile the call is connected fine but after a few seconds the it's hanged up and also if I call a land line the connection process takes around 1 minute. Not sure if everything is related but troubleshooting the audio issue, I've followed what I've read on some other posts and also on Elastix Without Tears. Here's what I've done so far: Configured port forwarding on the routers for both sides: ports 5060, 4569 and 10000-20000 from any external IP to my internal Elastix IP (Elastix side) and the Phone IP (Phone side) I also included the following lines on the sip_nat.conf file as suggested by pnaves (http://www.elastix.org/es/component/kun ... 606/#38854) and also tried to do it directly on the sip.conf. bindport=5060 bindaddr=0.0.0.0 nat=yes externhost=(IP for the router at the phone side) localnet=10.0.0.0/255.255.255.0 (Net address at the Elastix side) externrefresh=10 allowguest=yes context=from_internal rtptimeout=60 rtpholdtimeout=120 useragent=Elastix What could I be missing? Do I also need to configure IAX settings on the config files or SIP is enough? Thanks!