[Freeswitch-users] FreeSWITCH -- 30 Second call drop

Paul pasha at prosperity4ever.com
Fri Aug 2 01:13:02 MSD 2013



On Thu, 1 Aug, 2013 at 1:47 PM, E.J.C. Lindner <office at fordior.net> 
wrote:
> Hi Paul,
> 
> I saw you emails regarding 30 seconds call drop. I have exactly the
> same issue when I want to connect my client over OpenVPN to my FS
> server, which is also the vpn server at the same time (server is
> on a public IP). 
> 
> All my calls are dropped after 32 seconds... very annoying and I can't
> find the solution. So I'm constantly switching config for standard vs 
> vpn 
> for my client and FS (the standard config is just connecting to the
> public ip which is unencrypted).
> 
> Did you find a solution in the mean time? Please inform me, cause it's
> really killing me. :=|
> 
> I hope you can inform me on a short term; really appreciated!
> 
> -- 
> Yours sincerely, 
> E.J.C. Lindner
> 
Yes I found a solution to my problem.

My case differs a little bit from yours in that my openvpn server is on 
the router, not on the PBX, in your case it's probably even a little 
simpler.

So here is what I can advise you,

1. Ensure NAT is working properly over openVPN, so to test that all you 
need to do is have 2 phones on the same PBX able to call each other 
internally (extension 100 to extension 101 for instance).

If that works then your VPN NAT is probably good.

2. Ensure that your PBX is reporting the correct external address to 
your vendor upstream (this is where it was messing me up).

For example, for me my PBX was 10.0.0.40 (itnernal address) which the 
router was NATing to 24.38.231.4 (just a sample external address) and 
the same on the way back, when a request would come in for 24.38.231.4 
it would translate it to 10.0.0.40.

Because my PBX was not aware of 24.38.231.4 as it's external address, 
it was only listening on 10.0.0.40, it wasn't responding to the SIP ACK 
packets correctly, and the call was being perceived as finished and 
therefore hung up.

To fix it in FreeSwitch I had to edit external.xml and insert the 
proper IP address for sip_ext_ip and rtp_ext_ip (this is from memory)

I hope this helps you out cuz it cost me hours of frustration :)

Paul
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