[asterisk-users] NAT issues
Hello, this isn't an Asterisk specific problem but I don't know who else to ask for help. This is my setup, it oftens finds double NAT situations: [Asterisk box] - [Firewall IPCop] -INTERNET- [Random Router] - [Softphone] In certain situations, when two or more client softphones use the port 5060 at the same time and try to register, the UDP translation state of the port fails to assure the connection and drops both phones. If I change the client ports to random ones, they register, they can make calls and everything. It just happens if there is port clashing. I am not sure how to tackle this situation as enforcing random ports to the softphones is not viable for the setup. Is this a problem with the IPCop Firewall? I tried flushing the conntrack tables yet this situation kept happening. It gets to the point that nobody can use the 5060 port after a while (when everyone is trying to register). Thank you, Perssy Llamosas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NAT Issues?
Hi All I am having various problems that I am convinced is NAT related. I have a Vega box on public IP talking to an Asterisk box on a public IP address. Calls from the Asterisk to the Vega and back are fine. I have 2 VoIP phones in a NAT network registered to the Asterisk box. The problems I am having appear to be intermitent and adding quality to the phone config I can see that they are constantly changing from reachable to not. Incoming calls come from Vega to Asterisk fine and then dials the extension they should end up at: Executing Dial(SIP/xx.xx.xx.xx-0816dbf8, SIP/111|20|tr) in new stack -- Called 111 Sometimes the call will go and the extension will dial immediately but more often than not it will just sit and not do anything or go straight to voicemail. Another problem is when you make an outgoing call from the phones they are passed to the Asterisk and then to the Vega, when the person answers the Vega and Asterisk shows the call as connected but the VoIP phone continues to ring. Also when a call does make it all the way incoming and outgoing when you hang up the VoIP phone still thinks its connected! All the above happens when trying to call VoIP phone to VoIP phone as well! Any advice would be most appreciated, email me off list if you wish. Many Thanks Scott Pinhorne ___ --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] NAT issues
Hi there I have got a really strange issue and my problem is not that it is not working, but why it is working. I have Asterisk set up on a public IP, but the clients are behind a Port Restricted NAT with no support for UPnP. My clients dial into a meetme conference. When I don't specify nat=yes in the sip.conf file, then it works?? But not sure why it works because I cannot find any reference to the IP of the NAT in the SIP messages. I have not put in any nat support in my custom built client either. The reason that this is a problem is I a not sure if it will work on other LANs, and also find it hard to debug if it is working when my research tells me that it should not be working. I tried putting in nat=yes in the sip.conf file, and asterisk then rewrites the sip message with the IP of the Nat and the external port. It still works, but only if there is a constant flow of rtp traffic. If there is a break in the traffic, then the connection is lost. However, this problem may be to do with the fact that pinging is disabled on our network, but not sure. I am really stuck here. I have read that dealing with NATs can be a big problem, but it seems to work better when I dont put in any NAT support. Am I missing something here? Does anyone have any ideas or advice? Many thanks Steven ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NAT issues
Steven Langley wrote: I tried putting in nat=yes in the sip.conf file, and asterisk then rewrites the sip message with the IP of the Nat and the external port. It still works, but only if there is a constant flow of rtp traffic. If there is a break in the traffic, then the connection is lost. However, this problem may be to do with the fact that pinging is disabled on our network, but not sure. If you are the same person I spoke with on IRC then you forgot to mention that the SIP clients use VAD and that it cannot be disabled. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
