I need to update the From and To uris before dialing the second leg , i’m using the pre-dial subroutines to update the headers before dialing , and when i use pjsip_header (add) its added successfully , but it leaves the original from in the sip header ,
when i use pjsip_header(update) it’s saying no header with that name to edit ,
so how can i fix the multiple To , From bug .
I think it’s fair to say Asterisk tries to abstract away from low-level details of particular transports like SIP, analog lines, ISDN or whatever. If you want to do low-level SIP-specific stuff, then maybe look at alternative SIP-specific comms engines.