Is there any way that before dialling to 2nd party I can detect if RTP packet exchange is possible or not? and disconnect the call if exchange is not possible.
On calls I see reason for one way voice is that Asterisk is not getting RTP packets from SIP user. So I want such calls not to be processed.
I tried BACKGROUNDDETECT, but even when RTP packets are coming but silence in audio then it does not redirect to talk extension.
Or is there any solution to avoid one way audio in case when SIP softphone is not sending RTP packets.