Re: [asterisk-users] SIP/RTP Nat problem, can't solute it.
Still i cannot resolve this issue, please anyone can help me with this? Thanks in advance -- _ Facundo Agustin Barrera -- www.openlabs.com.ar "Let the penguins do the work" - Buenos Aires - Argentina _ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP/RTP Nat problem, can't solute it.
Thanks for the answers , tried canreinvite=no , but still cannot listen any soung from the outside, any other idea?? Thanks in advance -- _ Facundo Agustin Barrera -- www.openlabs.com.ar "Let the penguins do the work" - Buenos Aires - Argentina _ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP/RTP Nat problem, can't solute it.
Isn't that what externhost=sip.server.com.ar > my server name on the internet localnet=192.168.5.0/255.255.0.0 > my LAN is supposed to do? Bob... Rudolf Ladyzhenskii wrote: NAT changes address of the packet, but does not go inside of the SIP packet itself. And SIP packet contains address as well. If you look at debug output, you will see that SIP packets have remote host local address in them, not the public IP as one would expect. At least this is the problem I have. Basically one needs some software to "NAT" the addresses inside of SIP packets. STUN server is one alternative. I am about to put one in. Rudolf On 1/7/07, C F <[EMAIL PROTECTED]> wrote: Change To canreinvite=no On 1/6/07, Facundo Barrera - GMail <[EMAIL PROTECTED]> wrote: > Dear list: > I have the typical one way audio problem, as far as i know > it's a nating problem, my hosts inside my lan can call to outside > internet hosts, but can't listen a thing, i read a lot about sip and > rtp and protocols and the problem it seems to be with NAT, this is the > config i put on my sip.conf file about nat: > > externhost=sip.server.com.ar > my server name on the internet > localnet=192.168.5.0/255.255.0.0 > my LAN > nat=yes > canreinvite=yes > > And this are the ports i opened on my firewall script > > iptables -A INPUT -p udp -m udp --dport 8766:35000 -j ACCEPT > iptables -A INPUT -p udp -m udp --dport 5004:5082 -j ACCEPT > > > But still can't hear a thing from an outside call, any hel will be > appreciate > > Thanks a lot > > -- > _ >Facundo Agustin Barrera > -- > www.openlabs.com.ar > "Let the penguins do the work" > - >Buenos Aires - Argentina > _ > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP/RTP Nat problem, can't solute it.
NAT changes address of the packet, but does not go inside of the SIP packet itself. And SIP packet contains address as well. If you look at debug output, you will see that SIP packets have remote host local address in them, not the public IP as one would expect. At least this is the problem I have. Basically one needs some software to "NAT" the addresses inside of SIP packets. STUN server is one alternative. I am about to put one in. Rudolf On 1/7/07, C F <[EMAIL PROTECTED]> wrote: Change To canreinvite=no On 1/6/07, Facundo Barrera - GMail <[EMAIL PROTECTED]> wrote: > Dear list: > I have the typical one way audio problem, as far as i know > it's a nating problem, my hosts inside my lan can call to outside > internet hosts, but can't listen a thing, i read a lot about sip and > rtp and protocols and the problem it seems to be with NAT, this is the > config i put on my sip.conf file about nat: > > externhost=sip.server.com.ar > my server name on the internet > localnet=192.168.5.0/255.255.0.0 > my LAN > nat=yes > canreinvite=yes > > And this are the ports i opened on my firewall script > > iptables -A INPUT -p udp -m udp --dport 8766:35000 -j ACCEPT > iptables -A INPUT -p udp -m udp --dport 5004:5082 -j ACCEPT > > > But still can't hear a thing from an outside call, any hel will be > appreciate > > Thanks a lot > > -- > _ >Facundo Agustin Barrera > -- > www.openlabs.com.ar > "Let the penguins do the work" > - >Buenos Aires - Argentina > _ > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP/RTP Nat problem, can't solute it.
Change To canreinvite=no On 1/6/07, Facundo Barrera - GMail <[EMAIL PROTECTED]> wrote: Dear list: I have the typical one way audio problem, as far as i know it's a nating problem, my hosts inside my lan can call to outside internet hosts, but can't listen a thing, i read a lot about sip and rtp and protocols and the problem it seems to be with NAT, this is the config i put on my sip.conf file about nat: externhost=sip.server.com.ar > my server name on the internet localnet=192.168.5.0/255.255.0.0 > my LAN nat=yes canreinvite=yes And this are the ports i opened on my firewall script iptables -A INPUT -p udp -m udp --dport 8766:35000 -j ACCEPT iptables -A INPUT -p udp -m udp --dport 5004:5082 -j ACCEPT But still can't hear a thing from an outside call, any hel will be appreciate Thanks a lot -- _ Facundo Agustin Barrera -- www.openlabs.com.ar "Let the penguins do the work" - Buenos Aires - Argentina _ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP/RTP Nat problem, can't solute it.
Dear list: I have the typical one way audio problem, as far as i know it's a nating problem, my hosts inside my lan can call to outside internet hosts, but can't listen a thing, i read a lot about sip and rtp and protocols and the problem it seems to be with NAT, this is the config i put on my sip.conf file about nat: externhost=sip.server.com.ar > my server name on the internet localnet=192.168.5.0/255.255.0.0 > my LAN nat=yes canreinvite=yes And this are the ports i opened on my firewall script iptables -A INPUT -p udp -m udp --dport 8766:35000 -j ACCEPT iptables -A INPUT -p udp -m udp --dport 5004:5082 -j ACCEPT But still can't hear a thing from an outside call, any hel will be appreciate Thanks a lot -- _ Facundo Agustin Barrera -- www.openlabs.com.ar "Let the penguins do the work" - Buenos Aires - Argentina _ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
