On Mon, Nov 27, 2017, at 05:42 AM, Benoit Panizzon wrote: > Ok, answering myself: > > Asterisk 13.14.1~dfsg-2+deb9u2 > > Apparently suffers the pjsip transfer bug described @ > > https://reviewboard.asterisk.org/r/4316/diff/ > > Specifying the full URI: > > Transfer(PJSIP/sip:${DESTEXTEN}@trunk) does resolve the URI parsing > problem and is sending back the 302 message (which does not containg a > Diversion header, Jay promising, testing that next), but as described in > the Bug the Contact header is being messed up.
PJSIP requires a full SIP URI that can be used by the remote endpoint to be provided. The code does not look up an endpoint and try to construct a SIP URI for you. -- 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
