Dear all,
i'm having scenario where opensips 1.6.3 and asterisk is on the same host.
Opensips binds on port 5060 and asterisk on 5061.
Opensips handles user registration, nat traversal and "redirect" callls to
Asterisk PBX (configured as pstn gateway in address table and there is no
407 proxy
Dear all,
i'm having scenario where opensips 1.6.3 and asterisk is on the same host
behind nat and in DMZ. Opensips binds on port 5060 and asterisk on 5061.
Rtp proxy for nat traversal also running on same machine started and
opensips do not report any errors. ( started like rtpproxy -l
pub
Hi,
You might be in this case:
# ACK without matching transaction ->
# ignore and discard
exit;
I suggest you to put a log there and print the callid and if that
message appear search in the log for the corresponding messages -
INVITE, 200OK and ACK and send them here
When using opensips behind NAT, the config needs to be adjusted to
properly format:
- via headers,
- record-route headers,
- contact header,
- IP addresses inside SDP.
If at least one of the above headers is not properly formated, the SIP
signaling will not work.
Check your SIP traffic and mak
