Dear List It looks like the common way to to sip signaling over a trunk is:
In the Request URI, return the 'Register' Contact. In the To: Header, send the destination number. Unfortunately, asterisk with pjsip (i did not try chan_sip) does expect the dialed extension as request uri and does ignore what it is getting in the To: header. I could not find any hint in the documentation of this can be changed. I found instructions for a work-around: http://www.kempgen.net/voip/sip-request-uri-vs-to-header-routing.html In the meantime: Is there a way to tell the asterisk with pjsip to use the To: header to address an extension? Kind regards -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________ -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
