<br><br><div><span class="gmail_quote">On 6/5/07, <b class="gmail_sendername">Tom Rymes</b> <<a href="mailto:trymes@cascadelinksystems.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">trymes@cascadelinksystems.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 Jun 5, 2007, at 9:46 AM, Cosmin Prund wrote:<br><br>>> -----Original Message-----<br>>> From: <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-" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
asterisk-users-</a><br>>> <a href="mailto:bounces@lists.digium.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">bounces@lists.digium.com</a>] On Behalf Of Henry Cobb<br>>> Sent: Tuesday, June 05, 2007 4:30 PM
<br>>> To: Asterisk Users Mailing List - Non-Commercial Discussion
<br>>> Subject: Re: [asterisk-users] NAT<br>>><br>>> On 6/5/07, Iban Lopetegi Zinkunegi <<a href="mailto:lopetaz@hotmail.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">lopetaz@hotmail.com
</a>> wrote:<br>>>> Hi All!!<br>>>>
<br>>>> I have my asterisk working in my house (working with mandriva 2007<br>>> and<br>>>> asterisk 1.4 svn). Iīve looking on the net but i couldnīt find the<br>>> way of<br>>>> making work my asterisk in a real enviroment. Seems that the problem
<br>>> of NAT<br>>>> is a big problem. How can I sort out this, a mean crossing the NAT<br>>> and<br>>>> having asterisk connected?<br>>><br>>> If you want to receive calls and not just place them and you have a
<br>>> broadband connection with a dynamic IP then your server must register<br>>> with the VoIP provider and I suggest using IAX with the proper UDP<br>>> port assigned to your Atrisk server.<br>>>
<br>>> -HJC<br><br>> NAT is not that big of a problem, not anymore.<br>> Do a "NAT" search on <a href="http://www.voip-info.org" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
http://www.voip-info.org</a> - it'll get you<br>> started (got me started at least)
<br>><br>> --<br>> Cosmin Prund<br><br>Specifically, you need to set the following in sip.conf (if applicable)<br><br>nat=<br>localnet=<br>externip=<br>externhost=<br><br>You also need to configure your router to forward port 5060 and ports
<br>10000-20000 to your asterisk server.</blockquote><div><br>You make it sound very easy :-)<br><br>I've got a host connected to the internet with eth1 and to an internal LAN with eth0. The host runs asterisk. The internal LAN contains a number of SIP phones.
<br><br>eth0 = <a href="http://192.168.254.254" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">192.168.254.254</a> (network <a href="http://192.168.254.0/24" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
192.168.254.0/24</a>)<br>
eth1 = internet IP-address<br><br>I've set "externhost" to the dyndns name I've registered. When I do a lookup, this name returns the same IP-address as the one on eth1.<br><br>I've got a DID, and when I dial that number from my cell, the phones ring in my home. When I pick up the phone, audio only goes one way (from my home phone (behind the NAT) to the DID) audio the other way (from the DID to my home phone behind the NAT) is missing, due to NAT.
<br><br>It figured because my asterisk server tells the DID to send the audio to the IP-address of my SIP phone on the internal network (<a href="http://192.168.254.105">192.168.254.105</a>). I fired up wireshark and captured the packets.
<br><br>What I want to accomplish:<br>- calls within the LAN are re-invited (RTP goes from endpoint to endpoint)<br>- asterisk detects when a call is going beyond the local LAN (over the NAT), and then stays in the middle.
<br><br>I'm wondering if this is hard to do and how I'm supposed to configure this.<br><br>regards,<br><br>Stefan</div></div>