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>
<br>
Thanks<br><br><div><span class="gmail_quote">On 11/23/05, <b class="gmail_sendername">Michael West</b> <<a href="mailto:mwest@westmarkinc.com">mwest@westmarkinc.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div align="left" dir="ltr"><span><font color="#0000ff" face="Arial" size="2">I'm pasting something from another user on this list from
14/11/05</font></span></div>
<div align="left" dir="ltr"><span><font color="#0000ff" face="Arial" size="2"></font></span> </div>
<div align="left" dir="ltr"><span>
<p><font size="2">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.</font></p>
<p><font size="2">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. </font></p>
<p><font size="2">audio problems are almost always due to the RTP stream
(ports<span> </span></font><font size="2">10000-20000)
not being forwarded properly, either due to the port forwarding setup or the
sip.conf settings.</font></p>
<p><font size="2">Tom</font></p>
<p><font size="2">----------------------------------------------------------</font></p>
<p><font size="2">Tom Rymes</font></p>
<p><font size="2">Cascade Link Systems</font></p>
<p><a><u><font color="#0000ff" size="2">www.cascadelinksystems.com</font></u></a></p><font size="2">
<p>(603) 375-1414</p></font></span></div><br>
<div align="left" dir="ltr" lang="en-us">
<hr>
<font face="Tahoma" size="2"><b>From:</b> <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>On Behalf Of </b>Bharath
Khambadkone<br><b>Sent:</b> Wednesday, November 23, 2005 9:29 AM<br><b>To:</b>
Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> Re:
[Asterisk-Users] SIP Extension behind NAT,Asterisk on a public
domain<br></font><br></div>
<div></div>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<br><br>
<div><span class="gmail_quote">On 11/22/05, <b class="gmail_sendername">C F</b>
<<a href="mailto:shmaltz@gmail.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">shmaltz@gmail.com</a>> wrote:</span>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">On
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>><br>><br>><br>><br>><br>>
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