Hi All, I am running Elastix 1.6 on my machine. Here is a brief description of my hardware: 1 X OpenVox A800P Card Modules & Configuration: 3 X FXO Modules ( Connected to PSTN Lines) 5 X FXS Modules ( Connected to POTS Phone Extensions) Essentially 3 PSTN lines will be aggregated to 2 Ring groups & 1 Extension will be setup for my office. I have been following installation instructions from elastix without tears. I have so far as part of testing, created the following: 2 Trunks for 2 of the PSTN lines. 2 Extensions for testing to my POTS phones. Outbound routes for each of the extensions have also be created. Everything works great, i can call out on the appropriate trunk and all, however i encountered problems when i tried calling in. On the asterisk console, i get a no "CID or DID Match" when a call comes in, i am very sure i specified the correct DID on my incoming route but yet it still indicates a DID mis-match. So i tested by setting up a catch-all inbound route and it works great. However this is not going to work with what i have in mind, so i looked through my configurations and changed the context from 'from-pstn' to 'from-zaptel' in : chan_dahdi.conf & dahdi-channels.conf, but it still did'nt work ( even with restarting the server as well as services) Really hoping someone here can get me going in the right direction, i have redone the installation twice and it is still the same. Thanks!