I setup a time condition and the “Destination non-matches” I set to just terminate the call (tried both hangup and busy). However when the call gets routed there, the caller never hears a phone ring, nothing seems to ring on the PBX, and no “active calls” show asterisk info.
What I do see is a TON of “Active SIP Channel(s)” in the asterisk info Summary report (upwards to 20+). And in the Channels section I see under Chan_Sip Channel(s) all of those channels listed. All the Peer IP addresses are from my SIP Trunking providers 2 IP’s. User/ANR is the same number on each. Call ID shows multiple pairs of values. And the Peer is the same for each as well.
If I simply change the destination to a voicemail, IVR, queue, or anything else it works fine.
This is on a fully updated FreePBX 13 installation, running Asterisk 13.
Anyone have any insight to why this might be happening? Does this sound like a bug I should report?