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<DIV><FONT face=Arial size=2>Hey Guys,</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>I am having issues with One way Audio for outgoing
phone calls from my SIP phones. It works fine for Incoming, but outgoing
audio is not working. Also, Outgoing works fine if I put my SIP phone on
an Internet IP address, but if it's NAT'd then I get the 1 way audio on outgoing
calls.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>I am using the default nathelper module config, but
I have hacked it a bit and maybe my changes are causing the problem? I had
to use t_relay_to_tcp for our PSTN gateway and so I had to change the routing
around a little.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Prior to today I was using SerMediaProxy (AG
Projects), but I switched to PortaOnes rtpproxy just to make sure that there
wasn't a version incompatibility problem.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>I have also attached my current ser.cfg
file.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Anyone have any suggestions? Or should I
change the routing in my ser.cfg file? or would that make a
difference?</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Any help would be greatly appreciated.
Thanks!</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Darren Nay - <A
href="mailto:dnay@libertyisp.com">dnay@libertyisp.com</A></FONT></DIV>
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