Joshua Colp <jcolp <at> digium.com> writes: > > Hi James, > > I've trimmed the thread down, well, completely. ^_^ > > From looking at your information and reading the code it looks as > though there is a case where this may occur if certain NAT options are > enabled. This is certainly a bug as the code should just not execute > when WebSocket is involved. For an immediate fix you can set nat=no in > the entry in sip.conf. This should change the result and would also > explain why this has not been seen by others. >
Hi Joshua, I'm still getting the same result. Here is what I have in my sip.conf: [general] context=default ; Default context for incoming calls srvlookup=yes port=5060 bindaddr=0.0.0.0 ;pedantic=no rtcachefriends=yes dtmfmode=auto disallow=all allow=g729 allow=ulaw ; Allow codecs in order of preference allow=ilbc ;allow=silk8 allow=gsm ;allow=silk16 ;allow=silk24 ;nat=force_rport nat=no externip=example.org localnet=10.168.151.65/255.255.254.0 qualify=yes [3001] username=3001 secret=xxxxx host=dynamic type=friend context=test transport=ws nat=no [3002] username=3002 secret=xxxxx host=dynamic type=friend context=test transport=ws nat=no -- _____________________________________________________________________ -- 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
