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Thank you very much for your prompt replies. Perhaps I will consider moving to a 1.2 version of Asterisk. <br><br>> Date: Sun, 19 Aug 2007 12:08:36 +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 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 <br>> > 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. <br>> > These are filtered out by my server NAT's firewall. It seems that <br>> > Asterisk is not using rtp.conf<br>> ><br>> > I did some searching and found the following link. This is right around <br>> > 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>> I know what when I do that on my systems, it "just works". Even with <br>> xlite.<br>> <br>> I've never fiddled with rtp.conf. Mine is as it came with the default <br>> installation.<br>> <br>> rtpstart=10000<br>> rtpend=20000<br>> <br>> However, I'm running asterisk version 1.2.X, so there might be some other <br>> issues with 1.4.<br>> <br>> This is the scenario that 99% of my installations work under for people <br>> with phones not on the office LAN, and so-far so good (for me!)<br>> <br>> Gordon<br>> <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>> > _________________________________________________________________<br>> > Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy!<br>> > http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx&mkt=en-us<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 />Connect to the next generation of MSN Messenger <a href='http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=wlmailtagline' target='_new'>Get it now! </a></body>
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