[Freeswitch-users] OT SIP Debugging ideas

[ik]

Hello,

I have a bit of off topic question.

I have a PBX with soft-phones and everything seems to go well. The PBX have
several gateways, and all seems to work.

But when an extension is connected using a VPN (different IP range then the
PBX), an incoming call is having one side RTP, but only for some gateways,
that is, there are gateways that you are able to hear the call.
When the VPN based extension dial to a gateway, you always have RTP.

I used tcpdump, and was able to see when there is an RTP arriving back from
the extension, and when there are none, but I could not figure out any
reason for it. IT does not act as NAT issue, and there is no firewall.

My question is, how to debug this mess, what should I be looking for,
because I'm out of ideas.

Thank you for any help
Ido
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