Currently i am facing dead voice issues in my asterisk server.
I have checked RTP debug and it says Sent RTP P2P packet to x.x.x.x:xxxxx
When i turn on mix monitor that p2p rtp issue vanishes(no voice issues), i have googled about the issue,
I know the bridging types in asterisk,
but my question is how can i change my bridge type, more specific how can i restrict asterisk not to send p2p rtp stream.
Note: t flag in dial command did not worked, i am not thinking of patching asterisk server(that could be the last choice).
I am using out of band signaling SIP INFO.
in simple how can i disable all briging optimizations in asterisk.
PS:I am writing this question from general to support forum