Are the SIP phones behind a NAT/Router? is * behind a NAT or Firewall? can we see your sip.conf file? Regards, Uriel
-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Kevin Sent: Tuesday, September 30, 2003 8:05 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Grandstream Phone Issue When I dial with my Grandstream 101 telephone to another sip phone or Zap FXS, the call rings, but no audio is passed. Eventually the call gets disconnected. The same thing happens if I dial the Grandstream. Any Suggestions? _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
