I have an asterisk 16.30.1 version and when i make calls to it, i see it received on the asterisk server side ( I use sngrep to track the call flow). But, nothing happens after it. i have my dialplan to expect a prefix and dial out to the B-side. Prefix coming in from the A-side is matching the one expected in the dial plan.
But, i don’t see any activity and call simply times out and drops. Any suggestions?
There are two SIP channel drivers. Which one is in use? Does it show up in the respective SIP logging (sip set debug on/pjsip set logger on)? If it does not, then either the configuration is incorrect for listening or a firewall is blocking it (sngrep shows the packet BEFORE any firewall rules are applied).