I’m using both Xten Lite and Eyebeam, both behind NAT. My * server is sitting in a datacenter sans firewall. I’m experiencing two problems:
- Dialing into my voicemail using either softphone works just fine. SIP appears to be doing it’s job. I get connected to the VM prompt, but 4-6 seconds later the RTP stream just stops. Watching the CLI, I can see that the dialplan continues though. The same problem occurs when a PSTN call is bridged to my softphone. I can hear the person for only a few seconds before the sound dies; the call is still connected however.
Plugging my PC directly into the router, bypassing the NAt/firewall (WatchGuard SOHO6), things work perfectly.
- I installed Xten Lite on another PC on the same LAN as my first PC. Dialing extension-to-extension causes the call to be routed and setup fine. However, again, RTP is broken. Only this time I don’t get any sound at all, except for a couple of miliseconds worth every 30-60 seconds or so.
I’ve enabled/disabled reinvites, tried multiple codecs, and made sure that “nat=yes” in my sip.conf. I’m about out of ideas. Can anyone suggest anything? I have a Cisco 7960 sitting on my desk, but I’m still working with Cisco to try an get a SIP image.