Hi Folks, Its my first post here and Merry Christmas. I've just spend the last few days installing freeswitch + wikipbx with the latest svn sources (as of a couple of days ago). It is working. The server is an amd64 vserver on a gentoo. Machine is a dual EM64T 3GHZ with 8GB of RAM with a public IP. Twinkle able to call in from behind a nat and hear the echo test. I then got the speex codec working .It required sofiaconfig.py to be modified on wikipbx: "codec-prefs":"speex,speex@16000k,iLBC@30i,GSM,G722,PCMU@20i,PCMA@20i", I am able to call the sip provider (vitelity) and get an 800 number IVR audio. I am unable to get an incoming DID call answered via twinkle. Firstly it takes a long time to decide to call the extension and the provider hangs up, and then I'm not sure the NAT configuration is right. Is there some dns lookup stalling the extension calling? On my Grandstream BT200 I can get it registered but it does not do the echo test at all - for some reason this works on twinkle. I would like to see SRTP/ZRTP working. I've grepped around the configs and googled but all I can find that looks relevant is in: libs/sofia-sip/libsofia-sip-ua/soa/soa.c where there is an srtp_enable check. How is this enabled in the xml configs? Opportunistic zrtp encryption would be great. Should I bother hooking up an openser + mediaproxy solution for seamless NAT? I was hoping that freeswitch was a complete solution for an internet wide pbx where there would be a mix of NAT'ed and normal clients, and secure conferences would be possible. Cheers, Ray. -- http://www.rayservers.com/