After updating FreePBX from 2.9.0.11 to 2.9.0.12 the inbound calls were not routing to my ringgroup / extensions. The calls were coming from Sipura 3102 and no settings were changed other than upgrading the framework.
I Tailed the log in /var/log/asterisk/full I found the following errors when an incoming call was placed:
WARNING[2402] chan_sip.c: username mismatch, have
NOTICE[2402] chan_sip.c: Failed to authenticate device
This meant the Sipura was forwarding the calls correctly and FreePBX was receiving the calls too. The biggest clue was pstn between the brackets because the name matched the inbound route I had setup for the landline.
The second notice line was just information about the call itself so 123456 was the caller ID, the IP address (192.168.0.1) was the FreePBX server address. Not sure what tag was but that changed every so often.
Log in to the Cisco Linksys SPA management webpage as admin and go to the advanced view. Go to Voice > PSTN Line
tab and change User ID under Subscriber Information to the name of the inbound route on FreePBX. In my example above this would be “pstn”. This must match exactly the name of the inbound route name.
If the above is incorrect then the digest value will be populated with the SPA name. For example if the User ID was changed to Danny then the the new error message would be something like this:
WARNING[2402] chan_sip.c: username mismatch, have
Not sure why it had stopped working and looking back at my previous post on how to setup the Sipura and FreePBX it doesn’t look like I set the User ID for the PSTN line. Maybe a change from the old to the new version of FreePBX