Andreas van dem Helge wrote:
> I've had the opposite problem. Press mute while the call is still
> ringing and it will say "MUTE" on the display but the microphone is
> not muted. It was very embarrassing to discover this bug.
>
Have you reported this to grandstream? (It's a new one to me, but the
I've had the opposite problem. Press mute while the call is still
ringing and it will say "MUTE" on the display but the microphone is
not muted. It was very embarrassing to discover this bug.
On Wed, Feb 13, 2008 at 2:03 AM, Thomas Kenyon
<[EMAIL PROTECTED]> wrote:
> Lutgring, Sam wrote:
> I take
Lutgring, Sam wrote:
> I am using about 70 Grandstream GXP2000 phones on 1.1.5.15 code with *
> 1.4.16.2 and have not experienced any of these issues. The one thing
> that I would suggest is make sure that you are using RFC2833 for you
> DTMF Mode. I was originally using INFO and ran into some st
Can you test this whole thing with a phone?
On Feb 11, 2008 4:52 PM, Peder @ NetworkOblivion
<[EMAIL PROTECTED]> wrote:
> I have 20-30 GXP2000's connected to * over a T1 line. Neither end is
> NAT'd and there is plenty of bandwidth available over the line. The
> GXP's are 1.1.5.15, which is the
I want to make sure that 192.168.1.10 is the IP of the asterisk machine.
Anyways without looking at the phone's config I'd recommend you switch
to a newer Asterisk 1.2.x release at the very minimum (I suppose it
would be a "best practice" to recommend 1.4, however) all you will
need is a very few
calls.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Thomas
Kenyon
Sent: Tuesday, February 12, 2008 4:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Grandstream GXP2000 Loses Connectivity
Peder
Peder @ NetworkOblivion wrote:
> I hate to reply to my own message, but I have some more info from
> debugging. A Grandstream tries to register and uses a nonce and it is
> accepted by *. The next time it tries to register, it uses the same
> none and * says "SIP/2.0 401 Unauthorized". The Gr
One thing to keep in mind is that the Grandstream's firmware is
notoriously buggy and unreliable. I've got one GXP2000 here that is on
the 1.1.5.15 firmware, and I wouldn't even consider upgrading other
phones to them. Unfortunately, the quality of the Grandstream firmware
is appalling and doesn'
I hate to reply to my own message, but I have some more info from
debugging. A Grandstream tries to register and uses a nonce and it is
accepted by *. The next time it tries to register, it uses the same
none and * says "SIP/2.0 401 Unauthorized". The Grandtream says "ok,
here try this new n
I did post most of that. Point to point T1, no firewalls and no nat,
cisco routers, bandwidth is monitored at 30 second intervals and never
exceeds 50%, almost always 25% or less. The key is that I get messages
from * like "failed to register" and it is from the IP of the phone, so
it is like
Yes but network issues are still possible. What sort of network
connections are you using? What sort of routers/firewalls/other
network gear? Are you certain of the reliability of the T1? Also you
did not post what Asterisk version is in use. Please also post the
relevant sip.conf and configuratio
I have 20-30 GXP2000's connected to * over a T1 line. Neither end is
NAT'd and there is plenty of bandwidth available over the line. The
GXP's are 1.1.5.15, which is the latest. I have a problem where the
phones keep dropping off of * and I get a "failed to register" message
in the log of *.
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