Thanks for the help, Sorry about not originally providing that information. It's a 10. Local area network no Nat involved. I am using the default setting of the Grandstream and the following sip.conf
[gstream] type=friend username=gstream secret=test host=dynamic defaultip=192.168.0.7 context=internal canreinvite=yes dtmfmode=rfc2833 -----Original Message----- From: Brian West [mailto:[EMAIL PROTECTED] Sent: Tuesday, September 30, 2003 8:24 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Grandstream Phone Issue Any nat involved? and what codec's are you trying? On Tue, 30 Sep 2003, Kevin wrote: > 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 _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
