Hi Doug,
Please post a SIP capture of the call in order to understand the
scenario - the capture should be made from opensips server and it should
include both inbound and outbound traffic (for opensips).
The idea is that I need to check if the record_route was correctly done
at INVITE time, i
Hi All,
I'm having an issue with loose routing and call setups.
My call flow looks like the following:
192.168.112.110 (ATA) -> 192.168.110.1:5060 (OpenSIPS) -->
192.168.10.1:5080 (Sippy b2bua) -> 192.168.10.50:5060 (TDM Gateway -
Audiocodes)
Now, OpenSIPS and Sippy B2bua are on
