On 10/12/10 8:33 AM, Scott Richesson wrote:
Hello,
We are using sipx 4.2.1 and have sipxbridge working fine for a couple of
outbound trunks. Our SIPX server and phones are behind a NAT.
I have been playing around with SIP URI dialing, and I am not sure how the
media relaying/NAT busting is supposed to work.
I am using 4.2.0, and don't have any problems with this. The phones and
lines 'proxy' through sipx.
does it work if you set your domain.com as the proxy for the line?
and/or phone?
a good test is sip:[email protected]
You have a G.711 audio test then an echo test.
--
Michael Scheidell, CTO
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d: 561-948-2259
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