hi! I'm working an a asterisk test project at college at the moment. right now we're experiencing two problems. calling our sipphone (optipoint400) from a firefly client leaves us with no audio (no noise...nothing at all...) [the phone is ringing however and the connection seems to be set up] other way round works just fine!! firefly2firefly (stun enabled) also works perfectly!(??)!...(testclients and gateway are in different subnets).
the other thing is rather a config issue I guess :) when receiving incoming calls from the pstn direct access isn't working (dialing asteriskpstnnoplusextension) -> asterisk voicebox always answers standard config example could be pretty useul I guess ;P thanx in advance seb -- GMX ProMail mit bestem Virenschutz http://www.gmx.net/de/go/mail +++ Empfehlung der Redaktion +++ Internet Professionell 10/04 +++ _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
