On Fri, Dec 22, 2017, at 9:54 AM, Benoit Panizzon wrote: > 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?
Both chan_sip and chan_pjsip use the request URI, there's no configuration option currently to change it. Most people end up just doing the parsing in the dialplan. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _____________________________________________________________________ -- 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
