[asterisk-users] Configure Asterisk to preserve SIP header?

[Scott McNab]

Hello.

Is it possible to configure Asterisk to preserve specific SIP INVITE headers
when setting up a call?

Specifically, I have a custom SIP client that sends an additional header in
the INVITE request when originating a call. This is to request that the call
is auto-answered by the destination phone. i.e.

    Call-Info: <sip:192.168.100.50>;answer-after=0

If I use wireshark to sniff the packets, I can see that this header is
present in the original INVITE request from my SIP phone to Asterisk,
however, in the INVITE message that Asterisk sends to the recipient phone,
this header has been stripped.

Is it possible to configure Asterisk so that it forwards this SIP header
intact?

I know that it is possible to set up a dialplan to insert this header for
specific extensions, but I really would like to be able to generate this
header using my client!

Any ideas would be greatly appreciated!
Thanks
Scott
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