can u give me the configuration for the firewall?? with the same configuration i can't even talk or hear... its giving me the RTP Read Error whenever one picks up the phone.
cm ----- Original Message ----- From: "Steve" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, January 12, 2004 6:09 AM Subject: Re: [Asterisk-Users] Asterisk + BudgeTone (behind NAT) > On Saturday 10 January 2004 06:07 pm, Owen Kelso wrote: > > I'm using Asterisk on a open server (no firewall or NAT) and trying to > > communicate with a Grandstream BudgeTone 102 SIP phone which is behind > > NAT. The BudgeTone is at firmware level 1.0.4.30 and Asterisk is from CVS > > about a week ago. My problem is that I'm only getting half-duplex > > communication -- I can hear voice from the Asterisk server but the server > > does not understand any voice from me. From the console "sip debug" shows > > that the SIP part is working fine and DTMF via SIP INFO works. > > > I use OpenBSD firewalls with NAT and redirect and it works just as it's > supposed to. > > That's not even half duplex. In half duplex each side Can talk, but only one > at a time. It seems to be an error with configuring your firewall. (One > common error is to only turn on redirect. But you also need to Allow the > traffic to flow... > > -- > Steve > > __________________________________________________ > You actually need to constantly be alert > and willing to handle things, or life > will find a way to get you good! > _______________________________________________ > 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 > > _______________________________________________ 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
