[Asterisk-Users] Grandstream Phone Issue

[Uriel Carrasquilla]

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: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Kevin
Sent: Tuesday, September 30, 2003 9:42 PM
To: asterisk-users at lists.digium.com
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: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?
>
>
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> Asterisk-Users at lists.digium.com
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