[OpenSIPS-Users] OpenSIPS 3.1 & NAT issues
mark at allenclan.co.uk
Wed Jan 13 16:08:27 EST 2021
Hi all - I've been banging my head against this but not succeeding.
Firewall 46.x.x.x maps
| directly to
OpenSIPS 192.168.x.x Mid-registrar
- UAC can register on Asterisk via OpenSIPS
- UAC can call destination registered on Asterisk on local n/w to Asterisk
- 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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