That was exactly it! Thanks you VERY much! Mike
---- For the sip setting in sip.conf that setsup your voip provider add: canreinvite=no On 2/6/06, Michakl Gaudette <[EMAIL PROTECTED]> wrote: > > Hi, > > I've had a bit of a problem with one way audio, and it happens exactly when > I believe it shouldn't (and works perfectly when I would guess I could have > issues. > > Setup: > GrandStream GXP2000-------Linksys > Router-----------Internet------Asterisk box (hosted > somewhere, fixed IP, no NAT) ----------- VoIP provider -------PSTN > > When a call comes in from the PSTN, the call goes all the way to my desk > phone (the GXP2000) and it rings. Audio is clear, both ways. > > When a call is made from my GXP2000 phone to a PSTN phone (I use my cell and > my home phone as benchmark, they both get the same result) then I get no > audio at all. but ti does rin on the PSTN phone. > > > I've tried rerouting ALL of the relevant ports on my Linksys router directly > to my VoIP phone (5060 for SIP, 5004 for local RTP on the phone, 10000-20000 > as the Asterisk RTP ports)....Nothing works. > > What ports am I missing? Could the problem be entirely something else? > Somehow I had the feelings that calls going out (since they originate from > the device behind the NAT) would not be a problem, but calls coming in could > be. > > I really would appreciate a hint from somebody who knows better than I do > (i.e. anybody) > > Mike _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
