[Asterisk-Users] grandstream asterisk configuration

[Steve]

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

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