Kevin: try without the password to start with (remove it from the Budgetone). reinvite=no. Make sure you have disallow all codecs and allow ulaw/alaw. In the Budgetone make sure you have the correct IP address to your Asterisk. I would not use the defaultip. In the Budgetone make sure DTMF via INFO. Uriel
-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Kevin Sent: Tuesday, September 30, 2003 9:42 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Grandstream Phone Issue 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 _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
