On Sun, 19 Aug 2007, G B wrote: > > Hi Gordon, > > I did everything that you suggested, however, the symptoms remain. > > I set the rtp.conf to use ports 10000 to 20000 > > I assured that my router was forwarding these ports. However, the Media > Description Section of the SIP/SD packet (captured with ethereal) reads: > > Media Description, name and address (m): audio 50486 RTP/AVP 0 8 101 > > 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 > > I did some searching and found the following link. This is right around > the time that I downloaded. Could this be the trouble? > > http://lists.digium.com/pipermail/asterisk-bugs/2007-July/001213.html
I know what when I do that on my systems, it "just works". Even with xlite. I've never fiddled with rtp.conf. Mine is as it came with the default installation. rtpstart=10000 rtpend=20000 However, I'm running asterisk version 1.2.X, so there might be some other issues with 1.4. This is the scenario that 99% of my installations work under for people with phones not on the office LAN, and so-far so good (for me!) Gordon > >> Date: Sun, 19 Aug 2007 11:08:57 +0100 >> From: [EMAIL PROTECTED] >> To: [email protected] >> Subject: Re: [asterisk-users] Asterisk and Client NAT >> >> On Sun, 19 Aug 2007, G B wrote: >> >>> >>> Hi, >>> >>> >>> I realize that this is amongst the worst configurations, but I have been >>> made to believe that it can work... eventually. However, currently SIP >>> call set up seems to go fine, but no media is transferred in either >>> direction. For example, the following is output on the asterisk CLI >>> despite no voice being heard. -- Executing [EMAIL PROTECTED]:1] >>> Playback('SIP/john-081da978', 'hello-world') in new stack >> >> *sigh* The old NAT & SIP issue - again... )-: >> >> There is a lot of the VoIP WiKi on it. Eg: >> http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions >> >> However, assuming the asterisk and client are behind different NAT >> firewalls, do this: >> >> 1. Tell the client to use a stun server and don't fiddle with the client's >> firewall (other than to make sure it's not actually firewalling 5060 and >> 10000-20000) >> >> If you're stuck for a stun server, use stun1.drogon.net:3478 >> >> 2. Port forward 5060-5069 and 10000-20000 on the firewall that fronts the >> asterisk box to the asterisk box. >> >> 3. Tell asterisk it's behind a NAT firewall. >> >>> 1. sip.conf >>> [global] >>> nat=yes >>> canreinvite=no >> >> This isn't enough. You also need to tell it the IP address of the external >> firewall, and your local network address. >> >> nat=yes >> localnet=192.168.2.0/24 >> externip=1.2.3.4 >> >> Where 1.2.3.4 is the external IP address - the one the client is pointing >> to. This needs to be a static IP address (or at least not change for the >> duration of your use) the client can be behind a dynamic IP address. >> >> you might need a bit more in the client definition - eg: >> >> [100] >> context=internal >> type=friend >> secret=very >> qualify=yes >> nat=yes >> host=dynamic >> canreinvite=no >> dtmfmode=rfc2833 >> mailbox=100 >> callerid=Joe Bloggs <100> >> callgroup=1 >> pickupgroup=1 >> subscribecontext=BLF >> >> And that's it. >> >> Gordon >> >> _______________________________________________ >> --Bandwidth and Colocation Provided by http://www.api-digital.com-- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > _________________________________________________________________ > Invite your mail contacts to join your friends list with Windows Live Spaces. > It's easy! > http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx&mkt=en-us _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
