[Asterisk-Users] sip phone...direct access...

[s_fw at gmx.net]

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

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