Hi, I have two problems,
The first is that internal calls have no sound and subsequently drop.
The second is that calls coming in from the trunk have sound both ways but subsequently drop within a minute or two.
I have pastebinned the sip debug but from what I have seen and my little understanding of it, firstly my introduction with further detail of my tests is at the top, and it seems sensible to approach the debug from the bottom as I suspect I’ve added in much more than I need to. I’m just not sure what to cut out.
The calling party has sent the INVITE to a private use address, but has given a public contact address and its source IP is a public one. You have answered without making an outgoing call. Asterisk has responded to the public IP address and has supplied its public IP address. Either that response didn’t get through, or there was no route from the calling party to Asterisk’s public address, for the ACK.
Your NAT configuration seems to be messed up on the calling party and maybe more generally.
Thanks for the message.
Where do I even start at trying to resolve the issue?
By understanding your network topology.
Many thanks for shining a supportive light on the issue
There is basically something wrong with the way your network is configured, or how Asterisk is configured to work with your network, but unless you know how the network is configured, you are not going to be able to provide enough information to work out how to cope with that configuration. There is no single answer to such symptoms.