On Thu, Apr 7, 2016 at 11:04 AM, Jan Blom <jan.b...@peopleinteractive.se> wrote: > <snip> > Is this supposed to work? Any suggestions for workarounds?
I believe so. That sounds odd. Hard to know without seeing the packet trace of the call. Which SIP channel driver are you using? I think you are safe to go ahead and file an issue report. Please include the sip.conf/pjsip.conf plus a packet capture and Asterisk debug log (be sure to get the DEBUG channel turned on in logger.conf) with correlating SIP trace. -- 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
