Manny,<br>
Sorry if my post caused any confusion. I'm talking about 2 different locations of the server & client.<br>
My Asterisk server is located at my office and is not behind a NAT or firewall. It is directly connected to my Cable modem. <br>
I'm using a Sipura2002 ATA at home. This ATA is connected to the
asterisk server which is located at my office. The ATA at my home is
behind a NAT. The ATA sucessfully registers and can also make &
recieve calls only the voice is blocked. <br>
The external ports 10000-20000 were not opened on my Asterisk
box. Only port 5060-5082 were opened. I guess thats the reason I was
not able to hear any voice. Will try that this evening and post my
results.<br>
<br>
Thanks<br>
<br>
<br><div><span class="gmail_quote">On 11/23/05, <b class="gmail_sendername">Manny A. Wise</b> <<a href="mailto:mannywise@gmail.com">mannywise@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;">
<div>
<p><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; 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><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; 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 color="navy" face="Wingdings" size="2"><span style="font-size: 10pt; font-family: Wingdings; color: navy;">J</span></font></p>
<p><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;"> </span></font></p>
<p><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;">Manny</span></font></p>
<p><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;"> </span></font></p>
<p style="margin-left: 0.5in;"><font face="Tahoma" size="2"><span style="font-size: 10pt; font-family: Tahoma;"><span class="q">-----Original Message-----<br>
<b><span style="font-weight: bold;">From:</span></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><span style="font-weight: bold;">On Behalf Of </span></b>Bharath<br></span>
<b><span style="font-weight: bold;">Sent:</span></b> </span></font><font face="Tahoma" size="2"><span style="font-size: 10pt; font-family: Tahoma;">Wednesday,
November 23, 2005</span></font><font face="Tahoma" size="2"><span style="font-size: 10pt; font-family: Tahoma;"> </span></font><font face="Tahoma" size="2"><span style="font-size: 10pt; font-family: Tahoma;">10:08 AM</span>
</font></p><div><span class="e" id="q_107bde282083c8e2_3"><font face="Tahoma" size="2"><span style="font-size: 10pt; 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></span></div><p></p><div><span class="e" id="q_107bde282083c8e2_5">
<p style="margin-right: 0in; margin-bottom: 12pt; margin-left: 0.5in;"><font face="Times New Roman" size="3"><span style="font-size: 12pt;">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 style="margin-left: 0.5in;"><span><font face="Times New Roman" size="3"><span style="font-size: 12pt;">On </span></font></span><span>11/23/05</span><span>, <b><span style="font-weight: bold;">Michael West</span></b> <
<a href="mailto:mwest@westmarkinc.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">mwest@westmarkinc.com</a>>
wrote:</span></p>
<p style="margin-left: 0.5in;"><font color="blue" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: blue;">I'm pasting something
from another user on this list from </span></font><font color="blue" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: blue;">14/11/05</span></font></p>
<p style="margin-left: 0.5in;"><font face="Times New Roman" size="2"><span style="font-size: 10pt;">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: 0.5in;"><font face="Times New Roman" size="2"><span style="font-size: 10pt;">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: 0.5in;"><font face="Times New Roman" size="2"><span style="font-size: 10pt;">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: 0.5in;"><font face="Times New Roman" size="2"><span style="font-size: 10pt;">Tom</span></font></p>
<p style="margin-left: 0.5in;"><font face="Times New Roman" size="2"><span style="font-size: 10pt;">----------------------------------------------------------</span></font></p>
<p style="margin-left: 0.5in;"><font face="Times New Roman" size="2"><span style="font-size: 10pt;">Tom Rymes</span></font></p>
<p style="margin-left: 0.5in;"><font face="Times New Roman" size="2"><span style="font-size: 10pt;">Cascade Link Systems</span></font></p>
<p style="margin-left: 0.5in;"><u><font color="blue" face="Times New Roman" size="2"><span style="font-size: 10pt; color: blue;"><a href="http://www.cascadelinksystems.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
www.cascadelinksystems.com</a></span></font></u></p>
<p style="margin-left: 0.5in;"><font face="Times New Roman" size="2"><span style="font-size: 10pt;">(603) 375-1414</span></font></p>
<p style="margin-left: 0.5in;"><font face="Times New Roman" size="3"><span style="font-size: 12pt;"> </span></font></p>
<div style="margin-left: 0.5in; text-align: center;" align="center"><font face="Times New Roman" size="3"><span style="font-size: 12pt;">
<hr align="center" size="2" width="100%">
</span></font></div>
<p style="margin-right: 0in; margin-bottom: 12pt; margin-left: 0.5in;"><b><font face="Tahoma" size="2"><span style="font-size: 10pt; font-family: Tahoma; font-weight: bold;">From:</span></font></b><font face="Tahoma" size="2">
<span style="font-size: 10pt; 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 face="Tahoma" size="2"><span style="font-size: 10pt; font-family: Tahoma;">Wednesday,
November 23, 2005</span></font><font face="Tahoma" size="2"><span style="font-size: 10pt; font-family: Tahoma;"> </span></font><font face="Tahoma" size="2"><span style="font-size: 10pt; font-family: Tahoma;">9:29 AM</span>
</font><font face="Tahoma" size="2"><span style="font-size: 10pt; 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 style="margin-right: 0in; margin-bottom: 12pt; margin-left: 0.5in;"><font face="Times New Roman" size="3"><span style="font-size: 12pt;">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 style="margin-left: 0.5in;"><span><font face="Times New Roman" size="3"><span style="font-size: 12pt;">On </span></font></span><span>11/22/05</span><span>, <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 style="margin-left: 0.5in;"><font face="Times New Roman" size="3"><span style="font-size: 12pt;">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>
</div>
</div>
<p style="margin-left: 0.5in;"><font face="Times New Roman" size="3"><span style="font-size: 12pt;"> </span></font></p>
</span></div></div>
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