i’m using freepbx with Asterisk Version: 16.4.1
I have Sip trunks with service provider, I have issue to calling some numbers,
the reply of SIP trunk service provider " Your PBX is calling some VOLTE user, so when INVITE is sent from PBX with SDP param (a=fmtp:101 0-16), then PBX received response 183 Session Progress with SDP parameter a=fmtp:101 0-15 . Now as per standard RFC 4733 this SDP parameter must have values from 0 to 15 only. But PBX is sending values from 0 to 16"…
in trunk I’m already using dtmf mode = RFC 4733
How i can change in Invite message a=fmtp:101 0-16 to a=fmtp:101 0-15 to fit SP requirements ?