I have a Meta Switch Trunking to a Freepbx Server ( Current Asterisk Version: 15.4.0, FreePBX 18.104.22.168)
I am using freepbx as a Voicemail box. I have it working using an old script that a guy setup years ago on a old Trixbox setup. However getting it to pass stutter dial tone is extremely difficult for me.
I am extremely new to Asterisk so does anyone know where i can start to try to figure out if the Asterisk is actually passing any MWI to the MetaSwitch?
I have the extensions setup as virtual extensions with the Name being the actual phone number.
Here is my SIP.conf