Hi Guys,
I am new to asterisk and i have installed a FreePBX. i have a main Pilot number for the SIP trunk and outgoing and incoming calls are working fine,then i added a secondary DID as a second inbound route, every time i try to call in to this DID the calls terminates to the pilot number, i did some troubleshooting and i found out that the server looks at the INVITE sip:Pilot Number@192.168.66.242:5060 SIP/2.0 instead of the To: “” <sip:Secondary DID@Host, so it always route the call to the pilot number instead of the secondary DID
i have been trying to edit the extensions.conf and extensions_custom.conf but no success, any help would be appreciated.