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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'>Well, as the user stated on the original
message, the asterisk server is behind a NAT and the client is also behind a
NAT….</span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>if you make it work just by opening ports,
let me know..I have never been able to get it to work, that’s why I don’t
use sip, just plain iax2 for everything… </span></font><font size=2
color=navy face=Wingdings><span style='font-size:10.0pt;font-family:Wingdings;
color:navy'>J</span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'> </span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Manny</span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'> </span></font></p>
<p class=MsoNormal style='margin-left:.5in'><font size=2 face=Tahoma><span
style='font-size:10.0pt;font-family:Tahoma'>-----Original Message-----<br>
<b><span style='font-weight:bold'>From:</span></b>
asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]
<b><span style='font-weight:bold'>On Behalf Of </span></b>Bharath<br>
<b><span style='font-weight:bold'>Sent:</span></b> </span></font><font size=2 face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'>Wednesday,
November 23, 2005</span></font><font size=2 face=Tahoma><span
style='font-size:10.0pt;font-family:Tahoma'> </span></font><font
size=2 face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'>10:08 AM</span></font><font
size=2 face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'><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> Re: [Asterisk-Users] SIP
Extension behind NAT,Asterisk on a public domain</span></font></p>
<p class=MsoNormal style='margin-right:0in;margin-bottom:12.0pt;margin-left:
.5in'><font size=3 face="Times New Roman"><span style='font-size:12.0pt'>Thanks
Michael,<br>
I think thats is the problem, I have opened only ports 5060-5082, I need to
open 10000-20000 as well. I will try that and post the result when i get back
home.<br>
Thanks</span></font></p>
<div>
<p class=MsoNormal style='margin-left:.5in'><span class=gmailquote><font
size=3 face="Times New Roman"><span style='font-size:12.0pt'>On </span></font></span><span class=gmailquote>11/23/05</span><span class=gmailquote>, <b><span style='font-weight:
bold'>Michael West</span></b> <<a href="mailto:mwest@westmarkinc.com">mwest@westmarkinc.com</a>>
wrote:</span></p>
<p class=MsoNormal style='margin-left:.5in'><font size=2 color=blue face=Arial><span
style='font-size:10.0pt;font-family:Arial;color:blue'>I'm pasting something
from another user on this list from </span></font><font
size=2 color=blue face=Arial><span style='font-size:10.0pt;font-family:Arial;
color:blue'>14/11/05</span></font></p>
<p style='margin-left:.5in'><font size=2 face="Times New Roman"><span
style='font-size:10.0pt'>I would recommend that you do a little research on
google, voip- <a href="http://info.org" target="_blank"
onclick="return top.js.OpenExtLink(window,event,this)">info.org</a>, and the
list archives.</span></font></p>
<p style='margin-left:.5in'><font size=2 face="Times New Roman"><span
style='font-size:10.0pt'>To connect to an Asterisk box that sits behind NAT,
you need to forward ports 5060 and 10000-20000 too the asterisk box, and you
need to configure the externip, localnet, and nat variables in sip.conf. </span></font></p>
<p style='margin-left:.5in'><font size=2 face="Times New Roman"><span
style='font-size:10.0pt'>audio problems are almost always due to the RTP stream
(ports 10000-20000) not being forwarded properly, either due to the port
forwarding setup or the sip.conf settings.</span></font></p>
<p style='margin-left:.5in'><font size=2 face="Times New Roman"><span
style='font-size:10.0pt'>Tom</span></font></p>
<p style='margin-left:.5in'><font size=2 face="Times New Roman"><span
style='font-size:10.0pt'>----------------------------------------------------------</span></font></p>
<p style='margin-left:.5in'><font size=2 face="Times New Roman"><span
style='font-size:10.0pt'>Tom Rymes</span></font></p>
<p style='margin-left:.5in'><font size=2 face="Times New Roman"><span
style='font-size:10.0pt'>Cascade Link Systems</span></font></p>
<p style='margin-left:.5in'><u><font size=2 color=blue face="Times New Roman"><span
style='font-size:10.0pt;color:blue'>www.cascadelinksystems.com</span></font></u></p>
<p style='margin-left:.5in'><font size=2 face="Times New Roman"><span
style='font-size:10.0pt'>(603) 375-1414</span></font></p>
<p class=MsoNormal style='margin-left:.5in'><font size=3 face="Times New Roman"><span
style='font-size:12.0pt'> </span></font></p>
<div class=MsoNormal align=center style='margin-left:.5in;text-align:center'><font
size=3 face="Times New Roman"><span style='font-size:12.0pt'>
<hr size=2 width="100%" align=center>
</span></font></div>
<p class=MsoNormal style='margin-right:0in;margin-bottom:12.0pt;margin-left:
.5in'><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'> <a
href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank"
onclick="return top.js.OpenExtLink(window,event,this)">asterisk-users-bounces@lists.digium.com</a>
[mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank"
onclick="return top.js.OpenExtLink(window,event,this)">asterisk-users-bounces@lists.digium.com</a>]
<b><span style='font-weight:bold'>On Behalf Of </span></b>Bharath Khambadkone<br>
<b><span style='font-weight:bold'>Sent:</span></b> </span></font><font size=2 face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'>Wednesday,
November 23, 2005</span></font><font size=2 face=Tahoma><span
style='font-size:10.0pt;font-family:Tahoma'> </span></font><font
size=2 face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'>9:29 AM</span></font><font
size=2 face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'><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> Re: [Asterisk-Users] SIP
Extension behind NAT,Asterisk on a public domain</span></font></p>
<p class=MsoNormal style='margin-right:0in;margin-bottom:12.0pt;margin-left:
.5in'><font size=3 face="Times New Roman"><span style='font-size:12.0pt'>By
default AMP had NAT=yes in sip.conf, I read in some posts to change it to one,
i was just trying my luck if that works. I have tried NAT=yes, The Phone gets
registered, I can also make & recieve calls but as soon as the call is
picked I dont hear anything at both ends. Does this have anything to do with
codecs?<br>
<br>
Thanks</span></font></p>
<div>
<p class=MsoNormal style='margin-left:.5in'><span class=gmailquote><font
size=3 face="Times New Roman"><span style='font-size:12.0pt'>On </span></font></span><span class=gmailquote>11/22/05</span><span class=gmailquote>, <b><span style='font-weight:
bold'>C F</span></b> <<a href="mailto:shmaltz@gmail.com" target="_blank"
onclick="return top.js.OpenExtLink(window,event,this)">shmaltz@gmail.com</a>>
wrote:</span> </p>
<p class=MsoNormal style='margin-left:.5in'><font size=3 face="Times New Roman"><span
style='font-size:12.0pt'>On </span></font>11/22/05, Bharath Khambadkone <<a href="mailto:bkalthod@gmail.com" target="_blank"
onclick="return top.js.OpenExtLink(window,event,this)">bkalthod@gmail.com</a>>
wrote:<br>
> Hello All,<br>
> I'm fairly new to asterisk. I have read about the problems
about NAT, But<br>
> can't seem to find a solution. <br>
> My Asterisk is on a public domain, there is no NAT or firewall
in front of<br>
<br>
<br>
If no nat then why do you have nat=1 in sip.conf?<br>
<br>
<br>
> the asteris box. I have sucessfully connected iax2 softphones & was
able to <br>
> recieve & make calls. In the same locations where I have the iax2
extensions<br>
> working I have set up a a SIP softphone & a SIP ATA (Sipura2002). Both
teh<br>
> sip phones are able to register. I can also make & recieve calls but
cannot <br>
> hear anything after the call is answered at both ends. I'm not sure what
is<br>
> causing this problem. By the way I'm using SME server 7(centos
4.2) with<br>
> A@H installed.<br>
><br>
> my Sip.conf :<br>
> [2008] ;(Sipura2002)<br>
> username=2008<br>
> type=friend<br>
> secret=2008<br>
> record_out=Adhoc<br>
> record_in=Adhoc<br>
> qualify=no<br>
> port=5060<br>
> nat=1<br>
> mailbox=2008@device <br>
> host=dynamic<br>
> dtmfmode=rfc2833<br>
> context=from-internal<br>
> canreinvite=no<br>
> callerid=device <2008><br>
><br>
><br>
> [2009] ;X-Lite Soft Phone<br>
> username=2009<br>
> type=friend <br>
> secret=2009<br>
> record_out=Adhoc<br>
> record_in=Adhoc<br>
> qualify=no<br>
> port=5060<br>
> nat=1<br>
> mailbox=2009@device<br>
> host=dynamic<br>
> dtmfmode=rfc2833<br>
> context=from-internal <br>
> canreinvite=no<br>
> callerid=device <2009><br>
><br>
> Thanks in advance..<br>
</p>
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