I actually gave Linphone a try last week and it worked very well for me on a Ubuntu machine. (via my own Asterisk server setup for h.263/h.263+) So just to let you know it's possible. I don't remember doing anything special, except possibly disabling any other video codec options Linphone had maybe? Other then that, Asterisk does have a few quirks to get around if that's what you're using.
> Request-Line: REGISTER sip:dus.net SIP/2.0 < Status-Line: SIP/2.0 100 Trying < Status-Line: SIP/2.0 401 Unauthorized > Request-Line: REGISTER sip:dus.net SIP/2.0 > Request-Line: SUBSCRIBE sip:123456789@dus.net SIP/2.0 < Status-Line: SIP/2.0 100 Trying < Status-Line: SIP/2.0 200 OK
> Request-Line: REGISTER sip:voip.dus.net SIP/2.0 < Status-Line: SIP/2.0 100 Trying < Status-Line: SIP/2.0 401 Unauthorized > Request-Line: REGISTER sip:voip.dus.net SIP/2.0 < Status-Line: SIP/2.0 100 Trying < Status-Line: SIP/2.0 401 Unauthorized