No, I have been doing this for months… Each device has its own IP and some ISP’s allow this, some don’t.
I have 3 leased IP’s right now, 2 for routers and 1 for my Vonage adapter.
Vonage works fine, I Can talk and surf at the same time, I can download and have no issues while talking on the phone. Saying that Vonage is direct to the modem, no firewalls or NAT issues to worry about.
PBX should not be seeing a NAT issue either if the ports are forwarded to the server so I am lost, I am missing something or something is screwed up somewhere…
Even if the ports are forwarded, you are still using NAT, because your PBX has a non-routable IP (192.168.0.4). If you ping out to the internet from your PBX and ping from a workstation on the same network, both pings will appear to come from the same public IP (set by the PPPoE negotiation), but there are two different devices sending pings and your router will send replies to your appropriate network device.
Just out of curiousity, do you have DynDNS configured on both your PBX router and your network router? Also, are the PBX router LAN and network router LAN the same network?
It just seems that if you are working for a period of time like 2 days and then not working until a reboot of everything, it seems like some sort of timeout or lease time is expiring…DynDNS, PPPoE session, etc…
No, DynDNS is updated from the router that has the PBX on it, the other router is used for my wireless, other computers and devices.
Yes, all of the computers, notebooks, routers, phones, devices are all on the same network range I.E. within the 192.168.0.0 range.
Each router as I have said has its own IP supplied by my ISP and all I do is set the default gateway on the device that needs to access the individual router…
For instance and I will use fake IP’s for security… well the last digits will be fake anyway but I will show you what is happening.
Router 1 = 192.168.0.35 DNS router1.home.local Internet IP=184.13.139.***
Router 2 = 192.168.0.36 DNS router2.home.local Internet IP=184.13.146.**
As you can see from above, internet ip is different for each device.
PBX = 192.168.0.4 - to 192.168.0.35 - to 184.13.139.*** onto the internet.
I was intending on putting the PBX and phones on another network later when I can iron out these bugs.
One thing I was wondering is if my Vonage adapter is messing with the setup? it is Voip as we know and I assume it uses SIP and the same port? but being on another IP address you would think it should not affect anything… unless the Modem is getting freaked out by multiple hits on the same port?
I am a newb myself on freepbx but I found that using wireshark and monitoring all traffic to my PBX helped tremendously. I was able to watch all the SIP requests and find out where they were going to/coming from.
Also, when your system fails, do you still get call initiation, etc.? In other words, it still places and receives calls, just no audio, correct? That tells me that your SIP configuration is still correct.
If I understand VoIP correctly, the SIP protocol actually makes the connection between two VoIP devices, then SIP “gets out of the way” and the two devices exchange audio between themselves. IF your SIP is still working (placing/receiving calls), sounds like your UDP port forwarding is still working. Must be something else is getting lost.
Probably not much help but I hope it gives you something else to work with.
John
Yes, the calls can be placed and received, just no inbound audio. Its very odd it only happens sometimes.
I have had it happen on power failures where equipment comes up before the server and so on, I am in the process of replacing a faulty UPS which I hope will rectify that problem but I might try the packet sniffer to see what is going on, but I think the PBX handles the call from start to finish… its all very confusing for this semi-newb too.
I’ve seen several references in my searches that no audio (especially one-way) is usually related to NAT issues. Notice that several other posts in this thread have said something about NAT as well. You might want to research that a bit more.
Again, just a stab in the dark. Sometimes any little tidbit of a suggestion points me in the right direction.
John
I have done so many things related to NAT, but can’t figure it out why it will only do it once in a while and not all the time. NAT issues normally would appear all the time and not when it feels like it.
I am open to ideas but can’t find the solution, trust me I have done what I know I can do and have followed every thing possible on this forum and many others to no end.
I am stuck…
Someone else mentioned comparing when it works to when it doesn’t. That would be my next step. Using whatever packet capture tool you’re familiar with, capture all the traffic to and from your pbx while placing and receiving a few calls that work. Look especially for NON-SIP traffic since you know that the call progress is functioning. Then, when it breaks, do the same monitoring and look for what has changed. That should help you narrow it down.
John
Very good idea, I will give that a try! Thanks.
I am such a newb at wireshark, could you inform me how you set up to remote view the network traffic? I have Winpcap but that is not for linux.
Do you have a managed switch? You can run Wireshark on your Windows box and use span or port mirroring on the managed switch to look at the traffic.
Or if you have a spare dual-nic linux box you can turn it into a bridge pretty easily.
Just put it inline and sniff away.
I did a yum install wireshark and I have it running on the server 
So you are exporting the pcap file to a windows machine or you running x-windows on your phone system?
Both seem like a pain.
The easiest way - and what we did - was to put a simple HUB (not switch) between the wireshark PC (windows) and the pbx box and the rest of the network.
A switch will not forward traffic destined for a device on one port to any other ports, hence the need for the hub which simply rebroadcasts anything it receives.
Then, in wireshark, under the capture options, set a filter to only the IP for your pbx. There are examples in the drop down already. Just use one of them and modify for your IP.
Look for your SIP requests in your capture. Your voice/media packets should come real close after those.
HTH,
John
The hub option is another way of doing it.
If you are using managed PoE switches the proper way to monitor a port is to use the switches span or equivalent option.
I don’t believe I have a managed switch, most of my equipment is not commercial but I know you can buy switches and routers that are managed and not commercial.
Anyway, the PBX has been up System Uptime: 2 days, 15 hours, 11 minutes and it is still working with 2 way audio, saying that I have been watching the IP and it has not changed in a few days but I am unsure of the lease times on the IP. I will continue to see if that is causing the issue or I could go into my router and renew the lease and see what happens after that?
As for the wireshark, I run it on the server and then use WINSCP to view the data on my windows machine. It is not that much of a hassle but would love to remote in with wireshark to log information as I go… as of now I have to command line the operation but as I said, its not big deal.
One interesting thing I have seen in the log is the following…
I am not sure if the above would be any cause for alarm as it looks like it is just trying to send a report back from the server to the phone so maybe this port on the phone is blocked?
192.168.0.80 is my Cisco 7960 Phone and 192.168.0.4 is the PBX server, now the phone has been converted over to a SIP phone just for informational purposes.
I did manage to capture a working call yesterday so I am waiting now for the one way audio if it ever comes again… hope it does not but we will see.
I would like to just thank everyone that has been helping out up to this point, I very much appreciate it, without guys like you there would be a lot less people using the system as your all very knowledgeable and helpful.