On Tue, May 30, 2017, at 01:07 PM, Christopher van de Sande wrote: > Hi first post, so hope I'm not violating protocol. > > Been using Asterisk as home phone and hobby use for nearly 10 years. I > love this project. > > Anyway, would someone mind verifying my pjsip.conf ? This seems to work > well for 14.3.1 but I get no rtp into my natted Linphone when I upgrade > to 14.4.1. Other than that the phone registers properly on 14.4.1. > > I can provide a pjsip log as well, but for now I'll start with this. > > Asterisk is behind a shorewall firewall on a private natted network. It > has a single interface eth0.
The actual SIP traffic (pjsip set logger on) would also be useful to see exactly what is being exchanged and where it is going. -- 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
