Thanks for the reply.
Not what I'm looking for though.
I register an asterisk with with, say with username 1001. I receive a call from user 2 and want to terminate what 1002 sends with the gateway (not a gateway per se, but a registered user).
But all asterisk gets is a call to extension "s" in the INVITE, even though the "To" header is OK.... how do I set it correctly??
Via: SIP/2.0/UDP 1.2.3.4;rport;branch=z9hG4bKgmBy3gy8BSKZB
Max-Forwards: 69
Call-ID: 3393725d-e8d7-122f-6395-0013725ca38a
CSeq: 25549284 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 482
X-FS-Support: update_display
Remote-Party-ID: "David Villasmil" <
sip:1002@1.2.3.4>;party=calling;screen=yes;privacy=off