Call not going to failover in queue when the calls terminated from PSTN (ZAP) but it works when it comes(dials to queue ) from SIP extensions. Please help me out.Charles
Please provide a call trace showing the error as we have nothing here to work with. We don’t even know what version of things you are using asterisk 1.2.x, 1.4.x, 1.6.x? Version of FreePBX?
It normally takes a bit of useful information for us to be of some help.