On Sun, 30 Oct 2005 [EMAIL PROTECTED] wrote:
So, with this rules, the ATA receive calls, and I speak and my contrapart
listen me, but I can't hear him.
Any idea? Anybody can talk on VoIP behind NAT?
I've had the same problem myself. My pf.conf was "just" like yours, but I
had to configure my Sipura-adaptor right. In my case, this was "RTP Port
Min" and "RTP Port Max".
I don't know which ports your voip-provider uses, so that part you have to
find out yourself.
Now I haven't studied this enough to say this with 100% confidence, but
I've been playing with a bunch of Cisco IP phones (for a biz service) and
I may have a little insight here. The provider we use is setup similarly
to Vonage and other consumer services in that they use a "Session Border
Controller" between the phone and the SIP server. This box has the sole
purpose of "fixing things up that NAT breaks".
Anyhow, the Cisco phones also have that parameter adjustable, and it seems
to me that simply setting it on the phone somehow does insure that both
ends will use that port range (I assume it gets sent in some SIP control
messages). So setting them on the Sipura will probably have the desired
effect.
That said, at home I have both Vonage (w/ATA-186 adapter) and a Cisco
phone talking through PF with no special rules. :)
Charles
Also, remember to try forwarding the needed ports to your adaptor-ip. Also
set, if possible, the official i-net ip on your adaptor. On my Sipura this
is called "External IP".
Other than that, i'm out of ideas for your problem...
