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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