I have a Meta Switch Trunking to a Freepbx Server ( Current Asterisk Version: 15.4.0, FreePBX 14.0.5.1)
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
[SIP_IN]
dtmfmode=auto
context=from-trunk-sip-SIP_OUT
[SIP_OUT]
type=peer
username=**********
register=false
realm=voicemail
dtmfmode=rfc2833
canreinvite=no
context=default