We are testing with a new sip trunk provider and decided to test with asterisk 13. Everything seems to be working as should be except for call forwarding. I’m trying out PJSIP which has been challenging to say the least but the provider requires us to add a diversion header as they only allow their DID’s. I’ve been trying (and trying) to get PJSIP_HEADER to work doing the following:
When I look at the SIP messages I see a diversion header but it’s not this above that I’m trying to change. It’s the original 4 digit extension and SIP endpoint the call comes in on but seems to continue to be passed during the call forward.
Is there a reason not to use a diversion header? Prior versions of asterisk used sipaddheader so I’m assuming the same for PJSIP is to use the PJSIP_HEADER?
The reason not to add a diversion header is that one will be added automatically. Add header functions are basically there for non-standard or unsupported headers. Diversion headers weren’t supported in early versions of Asterisk.
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