[OpenSIPS-Users] OpenSIPS 3.1 & NAT issues

[Mark Allen]

Hi all - I've been banging my head against this but not succeeding.

Our setup...

UAC           192.168.x.x
  |
Router        5.x.x.x
  |
(internet)
  |
Firewall      46.x.x.x maps
  |           directly to
OpenSIPS      192.168.x.x      Mid-registrar
  |
Asterisk      192.168.x.x


Current situation:
- UAC can register on Asterisk via OpenSIPS
- UAC can call destination registered on Asterisk on local n/w to Asterisk
box
- Destination extension rings and can pick up call
- There is no audio either way & call drops after about 30 secs (Asterisk
kills call with "Requested channel not available" because not RTP traffic
is reaching destination)

I have tried passing audio through Mediaproxy on OpenSIPS box but with no
success. Using Wireshark I can see RTP traffic initiated at both ends, but
it doesn't reach the other end either way.

Is there some definitive guide to setting this up correctly or are there
specific steps that I need to follow?
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