Line not registered every so often

Good morning,

I have a pretty strange issue and I can’t seem to find a permanent fix for it. I have a deployment (Asterisk 13.22.0 / FreePBX that spans across 4 sites. Each branch points back to the main office over a site-to-site VPN. The server is on-premises at the main office.

Every so often, phones at one of the branches will stop registering. The VPN is still up, I can reach the server from the branch, and I can see the phone’s web interface from the main office. The phones, however, just refuse to connect to the phone server. Interestingly, if I change the IP address of a phone that isn’t working, it starts working again after rebooting. The phones will work just fine until the same thing happens (it might be a few days, it might be a few months). I’m guessing it has something to do with a failed registration blacklisting the IP, but I don’t see the block happening anywhere. I do have all of the subnets for my locations set to trusted and whitelisted in intrusion detection. Responsive Firewall is on and nothing shows up under Blocked Hosts.

The phones in question are below with models and firmware:
Polycom VVX310 (5.5.1)
Polycom VVX400 (5.5.1)
Polycom VVX410 (5.5.1)
Yealink W52P (

Any ideas or recommendations for next troubleshooting steps?


Responsive Firewall shouldn’t be included in this mix. If you are doing everything through VPNs, there are no unknown IP addresses, which is what RF is all about. Unless you have people calling in from roaming cell phones or Internet Cafes, I’d turn RF off.

In answer to the original question - look through your logs and see what the phone is doing. Many phones, especially if connecting through a Chan-SIP port, will retry connecting a few times and then give up. If you are seeing network errors causing the phones to become unreachable, Chan-SIP can and will just ignore them.

Thanks, Dave. I just turned RF off.

Should my end goal be to migrate to pjsip?

I’ll be poring over logs now.

Yes, it should. There are only a couple of scenarios where Chan-SIP is still relevant. PJ-SIP should be considered the primary method for SIP connectivity and Chan-SIP can be used as a last resort.

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I converted the extension to pjsip yesterday. I’ll give it time and then start converting the rest of my extensions to pjsip.

Thanks again, Dave.

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