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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'>Two reccomendations:<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 style='margin-left:36.0pt;text-indent:-18.0pt;mso-list:l0 level1 lfo1'><![if !supportLists]><font
size=2 color=navy face=Arial><span style='font-size:10.0pt;font-family:Arial;
color:navy'><span style='mso-list:Ignore'>1)<font size=1 face="Times New Roman"><span
style='font:7.0pt "Times New Roman"'> </span></font></span></span></font><![endif]><font
size=2 color=navy face=Arial><span style='font-size:10.0pt;font-family:Arial;
color:navy'>Enable nat for the SIP channels of the phones in SIP.conf.<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'>Or<o:p></o:p></span></font></p>
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10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal style='margin-left:36.0pt;text-indent:-18.0pt;mso-list:l0 level1 lfo1'><![if !supportLists]><font
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color:navy'><span style='mso-list:Ignore'>2)<font size=1 face="Times New Roman"><span
style='font:7.0pt "Times New Roman"'> </span></font></span></span></font><![endif]><font
size=2 color=navy face=Arial><span style='font-size:10.0pt;font-family:Arial;
color:navy'>If all the remote phones are in the same location, an IPSEC tunnel
between the remote router, and your Asterisk machine.<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'>Jason.<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><b><font size=2 face=Tahoma><span lang=EN-US
style='font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</span></font></b><font
size=2 face=Tahoma><span lang=EN-US style='font-size:10.0pt;font-family:Tahoma'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b><span style='font-weight:
bold'>On Behalf Of </span></b>Zeeshan Zakaria<br>
<b><span style='font-weight:bold'>Sent:</span></b> Saturday, 23 June 2007 1:37
AM<br>
<b><span style='font-weight:bold'>To:</span></b> Asterisk Users Mailing List -
Non-Commercial Discussion<br>
<b><span style='font-weight:bold'>Subject:</span></b> [asterisk-users] Audio
going one way for a few seconds during thecall</span></font><span lang=EN-US><o:p></o:p></span></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><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>Hi,<br>
<br>
This question was posted earlier, but there was no satisfactory answer to it.
Afterwards I tried everything but to no avail.<br>
<br>
The problem of audio going one way during the call for a few seconds is still
there. <br>
<br>
Its Asterisk 1.2.18 hosted Dell server with no NAT.<br>
Phones connect remotely through a hi-speed Internet connection, they are behind
NAT on a D-Link router, UDP ports 5060, 10001-20000 are forwarded to LAN,*,
which means they are forwarded to all the IPs. <br>
<br>
How can I fix this problem.<br clear=all>
<br>
-- <br>
Zeeshan A Zakaria <o:p></o:p></span></font></p>
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