[Asterisk-Users] Grandstream Phone Issue
Kevin
Asterisk at gtcus.com
Tue Sep 30 18:42:17 MST 2003
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:brian at bkw.org]
Sent: Tuesday, September 30, 2003 8:24 PM
To: asterisk-users at lists.digium.com
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
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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