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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Hi,<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'> I have asterisk 1.4.16 behind a NAT-FW which is using a
hosted SIP trunk for PSTN calling. Asterisk is configured to support nat with
nat=yes in sip.conf. Now the hosted PSTN Gateway supports symmetric RTP and
early media using 183 Session Progress. So If I call a PSTN number which has
IVR message played before the call is connected (via 183), those media RTP
packets do not reach the asterisk inside till asterisk sends out media packet
to the PSTN gateway. I have used rtpkeepalive option and set it to 1 sec. But
it seems that I drop rtp voice packets in the initial instructions played by
the IVR.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>How do I make sure that asterisk sends RTP packets (null rtp)
to the PSTN gateway just after receiving the media details in 183 SDP to open
the firewall port from inside?<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Regards,<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Mayur <o:p></o:p></span></font></p>
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