I just noticed something.
Looks like the SIP response from the softphone is still coming in on 5060.
That needs resolving. What about some other softphone set to use 5061 ?
I fixed that.
Still have the issue.
Incoming call (Failed)
https://pastebin.freepbx.org/view/ad1b8e0b
Outgoing Call (Success)
So now we have to understand how your incoming calls are routed.
I inherited this system, but from the looks of inbound routes there isn’t much going on.
Is there somewhere else I should be looking to find route related info?
If you don’t have ‘much going on’ for inbound DID routing then where do they go by default ?
There is a wiki , linked to at the top of this page, that might help you.
There are time conditions that determine which IVR is active (open/closed)
IVRs have direct dial enabled
If I direct dial my extension my soft phone rings, but disconnects on answer.
What about if it hits one of you 80ish hard phones ?
If I direct dial a desk phone from the IVR the call completes with audio both directions.
Then by a logical process of elimination you need to fix your soft phones, because your PBX is non denominational. Again , try another softphone maybe?
Tried Zoiper and MicroSIP
They call out, but go straight to voicemail on incoming.
Here’s the zoiper straight to voicemail log.
Best I can see is that your TDM call to your endpoint 146 finds it un-available
Show endpoints has it, and when I disconnect Zoiper it’s gone.
Endpoint: 146/146 Not in use 0 of inf
InAuth: 146-auth/146
Aor: 146 3
Contact: 146/sip:[email protected]:54794;rinstance 6e5f5f8594 Avail 22.843
I’m starting to lean towards ghosts. I think there’s ghosts in this machine. 
Calls can’t be completed to endpoints that are not available, no ghosts are ever involved.
Since 3CX is normally robust and reliable, I assume that the malformed response is either caused by the networking environment (IMO unlikely), or by something in Asterisk’s INVITE that it is not prepared to handle.
Although I see nothing ‘wrong’ with the INVITE, please try to simplify it:
- Don’t offer optional encryption. Ensure that the extension settings have Media Encryption set to None and Allow Non-Encrypted Media (Opportunistic SRTP) set to No.
- Don’t offer a long list of Codecs. Set Disallowed Codecs to all and Allowed Codecs to
ulaw&alaw
If you still have trouble, paste a new log.
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For Zoiper and MicroSIP, did they show as registered? Did Asterisk see them as registered?
You are a miracle worker.
I knew it was something simple, and you nailed it.
Set media encryption, Opportunistic SRTP and Codecs, and all is well.
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