I am having some problems with asterisk IP PBX ( FreePBX).
I am having two clients and I want that all of the SIP and RTP traffic should go through proxy. SIP and RTP Traffic is going through the proxy machine currently but I changed the proxy implementation such that instead of routing the media or RTP traffic through asterisk , proxy automatically forwards it to the other client. If A and B represents client, P represents proxy so in case of SIP traffic the flow is
A <--> P <--> PBX <--> P <--> B
while in the case of RTP traffic the flow is
A <--> P <--> B
but now the problem that occurs is the Asterisk automatically issues a hangup request ( bye message) after some time.
How can I solve this problem?
