Hi All
I need a little help, I need to try and preserve options in a call coming from a provider.
For this set up, I have a trunk coming in from a provider that is filling in a lot of P- fields in the SIP Invite packet for us to capture, however when FreePBX/Asterisk sends the call on to the system that needs the call, its stripping this out - presumably because the connecting call to the next system is a new call, not a forward.
Is there a way to preserve this data in the process? The call comes in and is sent on using an Incoming Rule with the new Trunk set as the destination.
The external trunk is using chan_sip and the internal on is on pjsip if sip driver matters.
Many Thanks