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Hi Gordon,<br><br>I did everything that you suggested, however, the symptoms remain.<br><br>I set the rtp.conf to use ports 10000 to 20000<br><br>I assured that my router was forwarding these ports. However, the Media Description Section of the SIP/SD packet (captured with ethereal) reads:<br><br>Media Description, name and address (m): audio 50486 RTP/AVP 0 8 101<br><br>50486 is the destination port of all RTP packets sent from the client. These are filtered out by my server NAT's firewall. It seems that Asterisk is not using rtp.conf<br><br>I did some searching and found the following link. This is right around the time that I downloaded. Could this be the trouble?<br><br>http://lists.digium.com/pipermail/asterisk-bugs/2007-July/001213.html<br><br>> Date: Sun, 19 Aug 2007 11:08:57 +0100<br>> From: gordon+asterisk@drogon.net<br>> To: asterisk-users@lists.digium.com<br>> Subject: Re: [asterisk-users] Asterisk and Client NAT<br>> <br>> On Sun, 19 Aug 2007, G B wrote:<br>> <br>> ><br>> > Hi,<br>> ><br>> ><br>> > I realize that this is amongst the worst configurations, but I have been <br>> > made to believe that it can work... eventually. However, currently SIP <br>> > call set up seems to go fine, but no media is transferred in either <br>> > direction. For example, the following is output on the asterisk CLI <br>> > despite no voice being heard. -- Executing [101@john:1] <br>> > Playback('SIP/john-081da978', 'hello-world') in new stack<br>> <br>> *sigh* The old NAT & SIP issue - again... )-:<br>> <br>> There is a lot of the VoIP WiKi on it. Eg:<br>> http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions<br>> <br>> However, assuming the asterisk and client are behind different NAT <br>> firewalls, do this:<br>> <br>> 1. Tell the client to use a stun server and don't fiddle with the client's <br>> firewall (other than to make sure it's not actually firewalling 5060 and <br>> 10000-20000)<br>> <br>> If you're stuck for a stun server, use stun1.drogon.net:3478<br>> <br>> 2. Port forward 5060-5069 and 10000-20000 on the firewall that fronts the <br>> asterisk box to the asterisk box.<br>> <br>> 3. Tell asterisk it's behind a NAT firewall.<br>> <br>> > 1. sip.conf<br>> > [global]<br>> > nat=yes<br>> > canreinvite=no<br>> <br>> This isn't enough. You also need to tell it the IP address of the external <br>> firewall, and your local network address.<br>> <br>> nat=yes<br>> localnet=192.168.2.0/24<br>> externip=1.2.3.4<br>> <br>> Where 1.2.3.4 is the external IP address - the one the client is pointing <br>> to. This needs to be a static IP address (or at least not change for the <br>> duration of your use) the client can be behind a dynamic IP address.<br>> <br>> you might need a bit more in the client definition - eg:<br>> <br>> [100]<br>> context=internal<br>> type=friend<br>> secret=very<br>> qualify=yes<br>> nat=yes<br>> host=dynamic<br>> canreinvite=no<br>> dtmfmode=rfc2833<br>> mailbox=100<br>> callerid=Joe Bloggs <100><br>> callgroup=1<br>> pickupgroup=1<br>> subscribecontext=BLF<br>> <br>> And that's it.<br>> <br>> Gordon<br>> <br>> _______________________________________________<br>> --Bandwidth and Colocation Provided by http://www.api-digital.com--<br>> <br>> asterisk-users mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> http://lists.digium.com/mailman/listinfo/asterisk-users<br><br /><hr />Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! <a href='http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx&mkt=en-us' target='_new'>Try it!</a></body>
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