Just after ideas on this one. I have ONE site out of a good number that sometimes just falls on it’s face.
The customer will call saying the “phones are down”
A look in my PRTG will show no such thing
Log into the console, “sip show peers” and all trunks are up
Any attempt to dial out fails. Out I get the message saying all circuits are busy. A check on channels shows this to be a lie.
Dialling in, you get diverted by the trunk provider to Mobile (Failover configuration)
In the provider’s UI (VOIPFone) the trunks are shown as down.
This seems to be triggered by a connectivity issue at the site, its as if the fact the trunk has failed isn’t registered by Asterisk/FreePBX
“sip reload” in the console and all is well again.
I don’t want to go down the route of leaving a script running to restart everything just before they get to work in the morning but this one has me stumped. I have exactly the same setups elsewhere and no issues.
And please don’t start jumping up and down about PJSIP
If State shows other than ‘Registered’, note that it’s common for router/firewalls to have ‘poisoned’ NAT associations that are kept alive by aggressive retries. You may be able to fix this by having Asterisk delay a long time (longer than the NAT timeout in the router) before retrying a registration failure. You may need to also set a long qualify frequency, in case these are contributing to keeping the bad association alive. Or, you may have settings that caused Asterisk to give up after too many registration failures.
If the registry State shows Registered, see what sip debug shows for the attempts. There may be a SIP ALG in the router that is somehow messing up the registration process.
Most providers don’t require registration to make calls; I don’t know whether this is the case for Voipfone. If so, sip debug may show useful info on a failed outbound attempt.