[Asterisk-Users] Grandstream Phone Issue

[Uriel Carrasquilla]

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: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Kevin
Sent: Tuesday, September 30, 2003 8:05 PM
To: asterisk-users at lists.digium.com
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
Asterisk-Users at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users