Hi, I have in my sip.conf
disallow=all allow=alaw in order to avoid any codec problems disturbing reinvite. And of course I have: canreinvite=yes In extensions.conf there is only one Dial command. It has no qualifiers like t or T. Just Dial(SIP/[EMAIL PROTECTED]) Anyway, asterisk does not try to reinvite. asterisk tells -- Attempting native bridge of SIP/01234567 ... but in the debug output there no reinvite. Using tcpdump I can see, that the audio data are going via the asterisk box in the middle, not direct between the endpoints. Is there anything else, which can prevent a reinvite? dtmp-settings? nat-settings? Thanks for any hints! Roger. _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
