Try: canreinvite=no
in your sip user definition. Julian J. M. On Apr 8, 2005 4:23 PM, Marlène Beray <[EMAIL PROTECTED]> wrote: > When I call from an IP Phone registered to the Asterisk server, the > connection is established and I can hear what the other person says but this > other person does not hear me. In fact, the Asterisk sends an Invite message > to the VoIP operator which replies; the connection is established. However, > the Asterisk sends another Invite to the firewall of the VoIP operator which > drops the message. As a consequence, the messages from the other person > reach the IP Phone but the messages sent by th IP Phone are dropped by the > firewall. _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
