Updates (Sticky)

December 27 2008

I tried upgrading to the latest GUI:

ipkg remove asterisk14-gui
ipkg install asterisk-gui

but had some problems with the CDR reader. While the
interface seems a little nicer, i would stay with
asterisk14-gui for now.

August 27 2008

I reinstalled asterisk on the slug today using
my instructions. It all seemed to work OK!

I did make a couple of minor changes: I added some
code under the Platform step to automatically start
asterisk on reboot; I also updated the version numbers
of the asterisk software for the ipkg install step.

FWD is no linger free. It can still be used
as an example of SIP trunking.

May 13 2008

I added a bit of an explanation to Step 3 on how
to allow all calls to go out. The original dialplan
blocks long distance calls by default (other than
to toll free numbers).

Friday, December 28, 2007

Introduction

Using two very inexpensive Linksys devices and some sort of USB storage device you can create a full featured home PBX using Asterisk, a Linux based open source PBX system.

We'll need to get an analog terminal adaptor called the Linksys SPA3102:

The SPA3102 acts as a gateway from the PSTN to your home VoIP system.

Next we'll need to get the Linksys NSLU2 which is a home NAS device:

This little box can be flashed with a version of Linux called Unslung to create a small server that will run Asterisk, an open souce PBX system. Finally, you'll need either a USB pen or external drive.

Our intention here is to create a system that can use the Asterisk GUI to set up our users, configure voicemail and add VoIP trunks. The GUI can also be used to monitor call logs and system performance. The GUI cannot be used to configure the SPA3102.

At the end of the process you should be able to dial out using a SIP phone and receive calls that will got to a SIP extension. If no one answers the extension, the call will go to voicemail.

We'll also be able to block unwanted callers using blacklists, phone numbers that can be added directly from the phone.

There are currently 7 setup sections on the blog. Follow them in order and you should have a functional home PBX. I will not bore you with how many hours it took me to figure this out! Steps 3-7 should also work with an Asterisk Now implementation as those steps are hardware independent.

If you see some glaring errors I would very much appreciate comments. I'm new at this home PBX stuff, although I have quite a bit of UNIX/Linux experience and some knowledge of the Alcatel OmniPCX 4400.

Disclaimer!!!!

These instructions are extremely bare bones, assume you have basic Linux knowledge, and are probably incorrect in many places!!!!

16 comments:

Unknown said...

Hello, I've tried settin up the asterik14 and asterik14-gui everything works perfect up to adding a user in the gui. After saving the page will not load and upon re-entering the gui user tab error msg 'page loading..loading' any help appreciated thanks I appreciate your work so far, this is the furthest I've gotten with this application

nslu2.voip@gmail.com said...

Try the following:

1) Empty the browser's cache, close the broswer and re-launch.

2) Try using a different browser.

3) Check the permissions of the files is /opt/etc/asterisk and in /opt/var/lib/asterisk/static-http. They should all be owned by root with the group root.

4) Manually add a user in users.conf and see if the GUI can detect it. If so, change one setting (say the voicemail password) and see if the browser will save it.

5) Make sure you've got the latest asterisk-gui ipkg installed:

asterisk14-gui - 0.0.0svn-r2036-3

ipkg update if it's an SVN version lower.

Let me know!

Unknown said...

Thanks, you were correct the new browser solved my issue here.

nslu2.voip@gmail.com said...

Great!

Roland said...

Hi, I'm a newbie and I tried your step by step guide, but I'm getting a "No Matching Peer found" error. Can you post a working sample file of all the configs I need for the NSLU2 and the SPA3102? If I can get the sample to work I can then work on connecting to Gizmo or FWD....

nslu2.voip@gmail.com said...

I might be able to help you if you can be more specific. Where is the "No matching peer found" coming from? If it's the SPA I beleive you need to look at your network settings again.

I do believe all of the config files are included in the posts. It's really just a cut and paste job for the Asterisk part.

The SPA config does require going through the post carefully and following the steps.

Let me know how it works out.

PS

I don't read the blog very often so apologies for the delay in responding!

Anonymous said...

In Step 4, are you sure it's /etc/opt/init.d/S99asterisk and not /opt/etc/init.d/S99asterisk ?

nslu2.voip@gmail.com said...

Hi Roland,

Right you are! It's been corrected

andrew said...

your blog was very helpful! thanks!

Digital Voip Converter said...

Hey frnd,
What is VoIP? Cud you tell me about Digital VoIp Converter?

emery said...

Hi! I hope that you could get around with it. So far I am not having problems with my VOIP in the office. VOIP offers a lot of advantages in our work so I hope that it will work for you too.
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it will work for you too.

small business phone systems said...

nice blog... Thanks emery.. I also faced the similar problem and get out of it now.

Unknown said...

I am so glad to see this post.

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HGH Los Angeles said...

Greetings! Very helpful advice in this particular post! It's the little changes that produce the most important changes. Many thanks for sharing!

Au Telecom said...

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