Preserve P- options in sip packet

[SilverHiro]

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

[SilverHiro]

Best I’ve been able to do is put the detail needed into the From Header. Looks like nothing else wants to carry over into the call.

[SilverHiro]

Cracked it, inbound was on sip so I used sip_header to capture what I needed into variables, then I took a copy of the standard ext-trunk and changed it to have a handler where I could add the headers on the outbound leg