I have a provider that in order to set outbound CID they want me to make sure that the From Header in the sip invite matches the caller ID while the contact header matches the registration info. For example. My phone number with my provider is 2125551212 which is also my username. I want caller ID to show up as 7185551212. The provider wants that the header should be like this:
=====begin SIP Headers===== Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to y.y.y.y:5060: INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK7d87735f;rport From: "7185551212" <sip:[email protected]>;tag=as56382d68 To: <sip:[email protected]> Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "7185551212" <sip:[email protected]>;privacy=off;screen=yes Authorization: Digest username="2125551212", realm="sip10.xchangetele.com", algorithm=MD5, uri="sip:[email protected]", nonce="a718c5ddb60d", response="7beb31c547f1252dcb9bf10b593b64a5", opaque="", qop=auth, cnonce="514eefb9", nc=00000001 Date: Tue, 11 Aug 2009 23:51:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY P-Asserted-Identity: <sip:[email protected]> Content-Type: application/sdp Content-Length: 220 ====End Sip Headers===== Is that possible? If yes how? TIA _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
