I am newish to elastix but have the task of testing a dedicated voice vlan and SIP trunk. I have been provided with a limited amount of information from the VSP but I have been able to get things working to a point. I am running the latest version of elastix at the time of writing this post (stable repo not beta) My Elastix server has 2 Nics in it. It was installed with one NIC and then I installed a second. eth0 is connected to my local network and the gateway is set to my local firewall. Elastix has internet access through eth0 eth1 is connected to a VSP supplied cisco layer 3 switch. A dedicated Vlan is setup between this cisco switch and the VSP through our fibre optic cable. There is no internet access on eth1. eth1 IP = 10.12.176.10 I have setup the eth0 IP and gateway according to the VSP's instructions. Their SIP media is on 10.20.61.1 behind a gateway of 10.12.176.252 I have a static route that routes like this: 10.20.61.1 gw 10.12.176.252 eth1 I can ping 10.20.61.1 I have setup incoming DDI's to some test extensions and can successfully call them from outside the phone network. I have experimented with the dial plans quite a lot and I think I have it sending the correct E.164 format to the VSP (see below) This is the error I get when calling out on the test system.(phone numbers changed naturally) == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called myVSP/6441111111 -- Got SIP response 603 "Decline" back from 10.20.61.1 -- SIP/myVSP-0000000e is busy == Everyone is busy/congested at this time (1:1/0/0) There are two things I suspect now after all my troubleshooting: sip_nat.conf <======== what would I want to put in here given my current network config. The VSP supplied layer 3 switch could be blocking required incoming ports?