On Wednesday 14 January 2004 01:52 pm, Mike Machado wrote: > On Wed, 2004-01-14 at 08:45, SW wrote: > > Hi, > > > > In my experience with GS phones, you need STUN support to make it work > > properly (behind NAT), otherwise you would need lot of trial end error to > > figure out how to do port forwarding. If you have STUN you wouldn't need > > to touch the Netgear (except for firewalls). > >
You don't need stun to work with Grandstream. My * is behind NAT and so is the GS of course. Two ports are open and redirected in the F/W, udp 4569 and 5036. I make and receive internal and external calls over both PSTN and the Internet. GS is configured: Software V 1.0.4.30 Static IP SIP Server is Asterisk's IP SIP user ID is the extension of GS Authenticate ID as user ID No pw Name is Steve Codex are: PCMU, PCMA, PCMU, G729, G726, G728 - note no 723 723 Rate is 6.3 Silence Suppression is Yes Voice Frames are 2 IP SoQ is 48 VLAN 0 SIP User is NOT phone number Dial Plan 202 SIP register YEs Clear Reg oin reboot NO Expiration 60 Early Dial No Use # as Dial Key is Yes SIP port 5060 RTP 5004 Random port is No NAT traversal is NO keel alive is 20 TFTP server is 130.94.123.253 Voice mail ID is 78202 DTMF is in-audio Payload is 101 - this may need to be changed NTP time.nist.gov Now all my features used to work a few months ago. I then stopped using * and came back a week ago. Updated CVS and now Hold is not working unless I press #(!?) But I can call, receive, transfer and have a working V/M. -- Steve __________________________________________________ You actually need to constantly be alert and willing to handle things, or life will find a way to get you good! _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users