[Asterisk-Users] Re: Re: RFC 3262 PRACK (Olle E. Johansson)

[Olle E Johansson]

Trond Andersen wrote:
> Headers like Supported or Require are not passed on with SIP methods
> like INVITE of 183 Session Progress.  In my dialplan I am able to read
> and add headers, but I think it only works to add one header in each
> message?

You can add several headers, but I advise you not to add standard SIP
headers. Stuff like X-asterisk-accountcode works miracles between 
Asterisk servers.

Now I see what you mean. The get header function only reads headers from
the first INVITE, not any other message. You can't go that deep into
signalling from the dial plan.

Adding headers only add headers on the first outbound INVITE (and the 
second INVITE if there's authentication required).

> Any tips on how I should move forward to make sure all headers are
> transmittet end-to-end between my endpoints through Asterisk?
Asterisk never ever forwards SIP messages between end points. So there's 
no way you can be assurred that headers are transmitted end to end.
You can grab headers on the incoming INVITE and add them to the
outbound SIP call.

That's the difference between a SIP proxy and a B2BUA :-)
We can only send supported and required headers that we actually support 
or require, not what the phone that issued an incoming call supported or 
required.

I hope I understood you and that I was able to provide you with a
decent answer...

/Olle