On Wed, Mar 26, 2014 at 1:14 PM, Mickael MONSIEUR <mickael.monsi...@gmail.com> wrote: > Hello, > > When I get a SIP INVITE as follows: > > INVITE sip:s@10.1.0.191:5060 SIP/2.0 > Max-Forwards: 69 > From: "0475XXXXXX" <sip:1053...@sip.domain.com>;tag=as7df9ab18 > To: <sip:02XXXXXX@IP:5060> > Contact: <sip:1053212@IP:5060> > Call-ID: 344d42bd16975a54141d11f635bdf...@sip.domain.com > CSeq: 102 INVITE > Date: Wed, 26 Mar 2014 15:06:01 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH > Supported: replaces, timer > Content-Type: application/sdp > Content-Length: 252 > > Asterisk considers that the extension is 's'. (The Register) > How to make the extension number that is shown in the 'To' ??
What version of Asterisk are you using? It would help to show how you are performing the dial in dialplan or otherwise. If you are dialing a user/peer present in sip.conf or a database then show that configuration as well. Based on that someone could make a suggestion. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com & http://asterisk.org -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
