This should be a relatively easy answer, I hope.
I have an above listed Asterisk flavor by Fonality/Trixbox, and while I've setup and configured a new SIP Trunk providers lines via their recommended config, we are experiencing a call reject situation, which they believe to be based upon the type of CSeq request we are providing.
Request URI: sip:firstname.lastname@example.org
From: "4224" <sip:email@example.com>;tag=18e2d6971b470beco0
CSeq: 101 INVITE
Contact: "4224" <sip:firstname.lastname@example.org:5060>
SDP IP: 18.104.22.168
SDP Port: 16552
Request URI: sip:xxx.xxx.xxx.xxx@VAxxx.xxx.xxx.xxx.net
From: "Al Fisher" <sip:email@example.com>;tag=as3b570f9a
CSeq: 102 INVITE
SDP IP: xxx.xxx.xxx.xxx
SDP Port: 10534
Any thoughts on how to control that? Remember, this flavor is the hosted hybrid PBX, so config line alterations are often overwritten when the system performs a refresh, so configuration through the GUI is preferable from a stability/management standpoint.