Hi all, I have a running OpenSIPS installation that I'm using for testing purposes.
The fact is that I'm forwarding requests from a voip provider to a jain slee server and everything is working fine (INVITEs, ACKs, RTP flow,...), except for the BYEs generated from the server side. They reach the OpenSIPs proxy and are not forwarded to the voip provider in order to finish the call. I'm not sure if I have to manually setup a route for this to happen, or if this behaviour is only available by using the B2BUA approach in OpenSIPS. Thanks a lot! David
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