Nicholas, Sorry I don't know, but are your calls working okay ?
Depending on the verbosity level being set, I see warning msgs all the time, that I ignore. Frequently, an upgrade to the next release of the same major version also eliminates the warning msgs. If you are really concerned, I would find an unused machine, install Linux & Asterisk 1.6.x on it, try out your calls and see if the warnings still appear. If there are no warnings of this kind, it is an issue specific to a module in that 1.4.x release and likely to go away. Good luck ! -- On Tue, Jan 5, 2010, Nicholas Blasgen wrote: > Asterisk 1.4.29 or so. > > access-list _dmz_acl extended permit udp 10.129.42.0 255.255.255.0 any range > 10000 20000 > access-list _dmz_acl extended permit udp 10.129.42.0 255.255.255.0 any eq > 5060 > > But yes, all your feedback worked. I didn't need to port-forward any > incoming ports, only 5060/10000-20000 for outgoing UDP. The only issue I'm > now having is: > > <--- SIP read from 66.227.100.20:5060 ---> > SIP/2.0 200 OK > Via: SIP/2.0/UDP 209.34.93.68:5060;branch=z9hG4bK3eb38bde;rport=51566 > .... > Warning: 392 66.227.100.20:5060 "Noisy feedback tells: pid=9611 > req_src_ip=209.34.93.68 req_src_port=51566 in_uri=sip:sip.jnctn.net > out_uri=sip:sip.jnctn.net via_cnt==1" > > 209.34.93.68 is my IP, 209.34.93.68 is Junction Networks (for this > example). I also get it from my backbone providers as well so it's likely > something to do with that 51566 req_src_port thing. Any idea what this is > an how to configure it to a restricted range of IP addresses? > > Nicholas Blasgen > Partner / Network Operations > Refractive Dialer LLC > (724) 252-7436 > > > On Sun, Jan 3, 2010 at 8:29 PM, Max McGraw wrote: >> >> Nicholas, >> >> you haven't specified which version, which does make >> a lot of difference. >> >> 1.6.x can easily traverse NAT. If you are only making >> outbound calls, you shouldn't need to forward 5060. >> >> Unless you have a special NAT that is blocking >> outbound connections, the SIP.conf settings below >> should work whether your provider uses SIP >> registrations or not. My codec related settings may >> not be applicable to your installation : >> >> ; ------------------------------------- >> [general] >> dtmfmode=rfc2833 >> relaxdtmf=yess >> bandwidth=high >> disallow=all >> allow=ulaw >> ; >> ; NAT stuff >> ; >> localnet=192.168.x.0/255.255.255.0 >> externip=a.b.c.d:5060 >> nat=yes >> ; >> ; Media stuff >> ; >> canreinvite=no >> ; >> ; >> [your-voip-provider-para] >> ; >> context=default >> type=friend >> ; >> ; your provider's outbound gateway >> ; >> host=w.x.y.z >> ; >> dtmfmode=rfc2833 >> relaxdtmf=yess >> disallow=all >> allow=ulaw >> ; >> ; ------------------------------------- >> >> >> On Sun, Jan 3, 2010, Nicholas Blasgen wrote: >> >> > I'm trying to move my Asterisk deployments under a Virtual IP address >> > and >> > now remember why I dislike this. My primary Asterisk system is now >> > behind a >> > firewall in private address space. My question is what ports are needed >> > to >> > be opened just for the purpose of placing outgoing calls. I would have >> > assumed none, but I can't even get replies on registration from any of >> > my 3 >> > VoIP providers. I tried defining the External IP and some other stuff, >> > but >> > I assume it's fully an issue with the firewall. Do I really need 5060 >> > port >> > forwarded just to register with remote hosts? >> > >> > Nicholas Blasgen >> > Partner / Network Operations >> > Refractive Dialer LLC >> > (724) 252-7436 >> > >> > __________________________________ _______________________________________________ -- 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
