Incoming calls not seen at all after upgrade from 12 to 13

We’ve been using FreePBX via RasPBX for about two years now and it’s been working pretty well. We’ve been running version 12 for probably a year now. Yesterday I upgraded from 12 to 13 using the web GUI. Everything seemed to go well, except for the fact that all users lost their voice mail boxes. This happened last time I upgraded as well. I set up everyone’s voice mail again, and it all seems fine, except for the fact that incoming calls aren’t working. When attempting to make an incoming call, the caller will hear dead air for about 15 seconds, then the call will end. I have once heard a busy tone, but have never heard a ringback since this issue started. Calls are not being forwarded to the global failover number, as they are when the PBX is shut down. All my extensions are registering correctly as far as I can tell. Outgoing calls work fine, extension-to-extension calls work fine. I am using SIPSTATION with two trunks. I have tried deleting and re-adding the trunks to no effect. FreePBX shows that both trunks are registered, and the firewall test in the SIPSTATION module passes. I have run the command tcpdump -i eth0 -n -s 0 port 5060 -vvv |grep "INVITE sip and can see the invite messages coming in when an incoming call is placed. Looking at the asterisk logs, even with sip debugging turned on, and core debugging and verbosity set to 5, I don’t see any output at all when an incoming call is placed. Based on past experience, I would really expect to see some log messages, even if the call was failing fairly early. I have turned off fail2ban, and am not running any firewall software like iptables on this device. The firewall configuration for the network hasn’t changed in months. The command netstat -an | grep 5060 returns
udp 0 0*, so it looks like asterisk has the port open. Should it be listening on TCP port 5060 as well? It looks to me like the invite packets are getting to the device, but not to asterisk itself, or else asterisk is not handling them at all for some reason. I have tried adjusting the inbound routes to send the calls to different places, but that hasn’t made any difference either. I am pretty much out of ideas at this point and would appreciate any help that anyone can offer. I do have a backup from the built-in backup/restore module from before the upgrade, but I would prefer not to have to roll back to that if I can avoid it.

I’ve just noticed something else. When the SIP invite comes in, it goes to both port 5060 and 5061, but lists 5061 as the target for both ports (e.g. INVITE sip:[email protected]:5061 SIP/2.0). It doesn’t look like asterisk is listening at all on 5061, so that’s kind of strange. The SIP registration going out to the SIPSTATION trunk also does not mention port 5061 at all. If I change the SIP bind port to 5061 and restart, then all the invites show up with 5060 as the target. I’m not sure what to make of that, but it seems very odd to me.

Well, it seems to have started working overnight. I’m not sure why, but I guess I’ll take it.