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

[chadd at codemunchers.com]

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