Hi, I have the same setup as you.
I didn't bother mapping any ports. I just enabled nat and keepalive. Here is a screenshot of the config on the phones. For some reason, on some phones I had to turn the NAT Mapping Enabled to Off otherwise call transfer didn't work. http://www.postimage.org/image.php?v=Tsz5WIJ The sip.conf setup for this phone is:- [winsor_202] type=friend context=winsor_phones host=dynamic secret=passwordhere nat=yes disallow=all allow=gsm allow=ulaw canreinvite=yes vmexte...@winsor mailbo...@winsor pickupgroup=1 callgroup=1 Hope that helps. Dan -- _____________________________________________________________________ -- 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
