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
[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
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
[asterisk-users] Maybe a NAT problem
Hi list: This is my first post and first off all i want to wish a good year for everone! well my problem is; i already installed asterisk on a server and created a channel and a couple of extensions, all seems to work just fine, y can make calls and receive them, i'm using the x-lite client that also works very good, this is the topology of the net (LAN - some clients) || Internal interface-private IP(server Running Asterisk)external interface-public IP ||-INTERNET Well i configure * to bind all address, so it's service listen on the two interfaces, when i make a call from a client inside my LAN to a client on the INTERNET, the person receives the call and listen me perfectly, but i can't listen any audio from him, i read about the issue and it seems to be a problem of nating, keep in mind that this server is masquerading all my LAN ips, so i can share my internet conenction, so when i receive a call form the outside world in fact x-lite shows me that the call originate from my inside interface IP of the server, but this is the strange thing the packets that originate the call from the outside world arrive just fine but when i answer the call i can't hear any audio at all. Any ideas how to solute this? hope not receive too much flames of this common issue 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
Re: [asterisk-users] Maybe a NAT problem
007/1/4, Bob Chiodini [EMAIL PROTECTED]: Facundo Barrera - GMail wrote: Hi list: This is my first post and first off all i want to wish a good year for everone! well my problem is; i already installed asterisk on a server and created a channel and a couple of extensions, all seems to work just fine, y can make calls and receive them, i'm using the x-lite client that also works very good, this is the topology of the net (LAN - some clients) || Internal interface-private IP(server Running Asterisk)external interface-public IP ||-INTERNET Well i configure * to bind all address, so it's service listen on the two interfaces, when i make a call from a client inside my LAN to a client on the INTERNET, the person receives the call and listen me perfectly, but i can't listen any audio from him, i read about the issue and it seems to be a problem of nating, keep in mind that this server is masquerading all my LAN ips, so i can share my internet conenction, so when i receive a call form the outside world in fact x-lite shows me that the call originate from my inside interface IP of the server, but this is the strange thing the packets that originate the call from the outside world arrive just fine but when i answer the call i can't hear any audio at all. Any ideas how to solute this? hope not receive too much flames of this common issue Thanks a lot In your SIP configs specify that the extensions are natted: nat=yes externhost=External IP address localnet=Local IP subnet/local subnet mask These are global settings. It might also be helpful to set canreinvite=no for each extension. There are probably firewall tricks you can do as well, but its early and I'm a couple cups of coffee shy. Bob... ___ --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 Thanks for the answer, will try that, but keep in mind that my server don't have an static public address, i use a dynamic DNS to resolve my sip domain. 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
Re: [asterisk-users] Maybe a NAT problem
2007/1/4, Carlos Rojas [EMAIL PROTECTED]: Hello Do no forget the rtp ports 1 to 2 Regards On 1/4/07, Facundo Barrera - GMail [EMAIL PROTECTED] wrote: 007/1/4, Bob Chiodini [EMAIL PROTECTED] : Facundo Barrera - GMail wrote: Hi list: This is my first post and first off all i want to wish a good year for everone! well my problem is; i already installed asterisk on a server and created a channel and a couple of extensions, all seems to work just fine, y can make calls and receive them, i'm using the x-lite client that also works very good, this is the topology of the net (LAN - some clients) || Internal interface-private IP(server Running Asterisk)external interface-public IP ||-INTERNET Well i configure * to bind all address, so it's service listen on the two interfaces, when i make a call from a client inside my LAN to a client on the INTERNET, the person receives the call and listen me perfectly, but i can't listen any audio from him, i read about the issue and it seems to be a problem of nating, keep in mind that this server is masquerading all my LAN ips, so i can share my internet conenction, so when i receive a call form the outside world in fact x-lite shows me that the call originate from my inside interface IP of the server, but this is the strange thing the packets that originate the call from the outside world arrive just fine but when i answer the call i can't hear any audio at all. Any ideas how to solute this? hope not receive too much flames of this common issue Thanks a lot In your SIP configs specify that the extensions are natted: nat=yes externhost=External IP address localnet=Local IP subnet/local subnet mask These are global settings. It might also be helpful to set canreinvite=no for each extension. There are probably firewall tricks you can do as well, but its early and I'm a couple cups of coffee shy. Bob... ___ --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 Thanks for the answer, will try that, but keep in mind that my server don't have an static public address, i use a dynamic DNS to resolve my sip domain. 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 Ok, what about my dynamic IP address when setting NAT?? Thanks. FB -- _ 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
