Hello, I need help, I’m going crazy! 
I have a freepbx hosted on amazon AWS.
I use a trunk to a terminator service to call from inside the PBX to the world.
The phone inside my PBX are obviously under NAT, the terminator has public ip address.
This is my problem:
For example mister one inside the PBX call a mobile number, mister two.
The conversation starts, if mister one need help to manage the call, pushes a web button that launch a call to mister three and after the answer the command ChanSpy(SIP/XXX,w) is executed, mister three can listen all but can talk (wishper) only with mister one.
All is working good except the fact that the mister three voice reach mister one only if mister one and/or two are talking.
I try to whisper mister two and is all perfect no needed to have mister one and/or mister talking. The voice from mister three always reach mister two. The different between mister one and mister two is that mister one in under NAT…
I’ve tried different ways to manage this
- Asterisk call files
- ARI SnoopChannel
- Directing calling dialplan
but the result is always the same.
Searching on the web I’ve found this:
Audiohooks require constant media flow for whispering
https://issues.asterisk.org/jira/browse/ASTERISK-24397
I’m the only one with this problem?
Is a bug?
A NAT problem?
A problem concerned to the AWS hosting that cannot manage the UDP packet timing?
Please help! 