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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>This belongs in the asterisk-users list.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Seems like a NAT issue, make sure nat=yes
in your sip.conf and take a look at voip-info.org<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>__</span></font><font color=navy><span
style='color:navy'><o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Justin</span></font><o:p></o:p></p>
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<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'>
asterisk-gui-bounces@lists.digium.com
[mailto:asterisk-gui-bounces@lists.digium.com] <b><span style='font-weight:
bold'>On Behalf Of </span></b>Eric Swanson<br>
<b><span style='font-weight:bold'>Sent:</span></b> Friday, March 30, 2007 5:25
AM<br>
<b><span style='font-weight:bold'>To:</span></b> asterisk-gui@lists.digium.com<br>
<b><span style='font-weight:bold'>Subject:</span></b> [asterisk-gui] asterisk
1.4.2 and outgoing sip providers</span></font><o:p></o:p></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
<p class=MsoNormal><span class=postbody><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>I am a newbie to asterisk. </span></font></span><font
size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'><br>
<br>
<span class=postbody>I have an asterisk server that is behind a cisco pix
firewall. I have opened up ports udp / tcp 5060 and rtp 10000 - 20000 ports. </span><br>
<br>
<span class=postbody>When I make a phone call out through the sip provider, I
can not hear the other person, but the other person can hear me. Do I need to
setup a stun entry for the asterisk server since it is behind the firewall? </span><br>
<br>
<span class=postbody>How can I fix this issue? </span><br>
<br>
<span class=postbody>Thanks </span><br>
<br>
<span class=postbody>Eric</span><o:p></o:p></span></font></p>
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