[asterisk-biz] Grandstream GXP-2000, Asterisk 1.2.13 stability problems

Fri Jan 26 12:42:45 MST 2007

I'd recommend Mikrotik for your QoS.

www.visualware.com has a good VoIP test.

-----
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com

----- Original Message ----- 
From: "James Hawks" <james.hawks at customerfunding.com>
To: <chadd at codemunchers.com>; "'Commercial and Business-Oriented Asterisk 
Discussion'" <asterisk-biz at lists.digium.com>
Sent: Friday, January 26, 2007 10:11 AM
Subject: RE: [asterisk-biz] Grandstream GXP-2000,Asterisk 1.2.13 stability 
problems

We implemented a 100 person call center using asterisk and Grandstream
phones. We had countless problems with our implementation which included
dropped calls, static, and several phone problems. Most of our problems were
solved by switching to Polycom phones from Grandstream. I have seen lots of
people happy with Grandstream but it just did not work for us. After
changing phones our system ran smooth with very little issues. One thing to
try is a different phone. Also make sure you have no IRQ conflict issues but
with FC5 you shouldn't. It looks like you use no Zaptel hardware and are
straight VOIP. 768 kb bandwidth might not be enough for both data and voice
if you are using the same internet connection for both. The last thing to
consider is QoS and the latency of you DSL connection. There are websites
that will check your latency for VOIP applications. A good QoS
implementation can solve the sharing of data and voice problems.

James Hawks

-----Original Message-----
From: asterisk-biz-bounces at lists.digium.com
[mailto:asterisk-biz-bounces at lists.digium.com] On Behalf Of
chadd at codemunchers.com
Sent: Thursday, January 25, 2007 8:58 PM
To: asterisk-biz at lists.digium.com
Subject: [asterisk-biz] Grandstream GXP-2000,Asterisk 1.2.13 stability
problems

Hello everyone,

I was directed to this mailing list by a good friend of mine in our local
LUG. I

am going to try and provide as much information as possible and a full
problem

description. If you have had any problems similar to what I have here I
would

appreciate any feedback you can provide.

About a month ago we did an installation of an asterisk server in a small
office

building. We deployed 8 Grandstream GXP-2000 phones, with firmware version

1.1.1.14. The phones have been configured to get addresses from the local
DHCP

server. The phones are configured to use the ulaw codec.

The asterisk server is an AMD Athlon 3.2 Ghz machine with 1GB of system
memory,

running Fedora Core 5 and asterisk 1.2.13. I did not install libpri or
zaptel

drivers. Just compiled asterisk from source and installed.

The phones connect to a D-Link DSS-16+ rack mounted switch. The asterisk
server

also plugs into the switch along with a Westel 611C modem provided by
Verizon.

The internet connection is a Verizon DSL line, with 3 Mb down and 768 Kbps

upstream. The PC's use a seperate modem to get onto the internet through a

differant DSL line. We are not using a spliiter, we have two seperate lines

coming in from the street.

Here are links to all the pertinant config files. If you require differant
files

please let me know, I am more than happy to upload them.

http://codemunchers.com/PUB/JCA/features.conf

http://codemunchers.com/PUB/JCA/sip.conf

http://codemunchers.com/PUB/JCA/voicemail.conf

http://codemunchers.com/PUB/JCA/extensions.conf

http://codemunchers.com/PUB/JCA/iax.conf

Now for the fun part, the list of problems. Again, any help would be welcome

=============================================

-A fairly common problem is calls get dropped completely when the duration

reaches about 10 - 15 minutes. Calls do not always get dropped, kind of a
random

thing.

-Time lag on calls, sometimes takes up to a second for my voice to go from
my

cell phone to the phone in the office.

-Once in awhile a customer will be unable to hear us, but we can hear them
clear

as day.

-Choppiness, sounds like caller is underwater.

-Yesterday I got a call that the Welcome menu mysteriousely started playing

during a conference call. In extensions.conf the main menu is located in the

context incoming-main.

========================================================

We found that restarting the asterisk server sometimes temporarily fixes
some

issues. But I am still new at using asterisk, I have been using the
GNU/Linux

operating system for years now, so don't be afraid to ask some technical

questions if they will assist you in helping me.

If there is any additional information you need please feel free to ask. Any

insight, war stories, best practices you can send my way are more than
welcome. I

have a few ideas I would like to try, but I am not sure if they would do any
good.

here are a few ideas I have had so far;

-Install a second NIC in the asterisk server, connectiong the modem directly
to

that interface. Then connecting the other interface to the switch that
connects

all the phones. Currently, asterisk has to go through the switch to get to
the

internet.

-Assign static addresses to the phones, was thinking this would help with
routing?

------------------------------------------------------

Also, if you can point me in the direction of a good troubleshooting guide
that

would be additionally helpful.

Thank You,

Chadd J.

chadd at codemunchers.com

(909)944-4009

Rancho Cucamonga, CA, USA

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