Hi, I've a SIP multi domain environment based on OpenSer, so subscribers use OpenSer as registrar and proxy. I want to use FreeSwitch (it's my first time with it) as an application server / B2BUA, but I need it to work in SIP multi-domain way, this is, if for example I use the voicemail server of FS, there will be voicemails for users called: - alice@domain1 - alice@domain2 - bob@domain1 ... This is unfeasible with Asterisk that just handles the username part of the Request URI and From/To headers. How possible is it with FS? is it designed to handle multidomain in SIP? I have other questions more related to pure SIP protocol and FS as transparent B2BUA: alice ---(leg_a)--- FS (B2BUA) ---(leg_b)--- bob - If alice adds a custom header ("Subject: hello!"), will this header appear in leg_b to bob? - If alice sends a custom in-dialog request (INFO request with no standar Content-Body), will FS forward it to bob by just changing the needed data (From_tag, To_tag, Call-ID, Via, CSeq...)? - When alice sends the initial INVITE to bob (through FS), if bob replies a "strange" response code ("497 Strange response"), will FS respect it and forward upstream to alice? - Is it possible FS to act as a SessionTimer endpoint with alice and bob after the call (both legs) has been established? I don't want FS to handle media (so I'll use "media_bypass" mode) but I need FS controling the call status by sending in-dialog requests, hopefully using SIP Session Timers (RFC 4028). Thanks a lot for all your replies, I hope to enjoy FS :) -- Iñaki Baz Castillo