# Internal LAN VoIP

## RBH

The office is looking for a VoIP solution for use on the LAN. We've essentially got Skype down as a last resort due to its funny supernode behaviour, but the ideal would be a service that's wholly inside our network. No need to get out, no need for anything to come in.

Any suggestions?

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## nevynxxx

All the major Switch makers produce kit that will do this.

Isn't this what asterisk does?

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## RBH

I was under the impression that Asterisk acted as a gateway to a PSTN?

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## BigBaaadBob

 *RBH wrote:*   

> I was under the impression that Asterisk acted as a gateway to a PSTN?

 

Asterisk is a swiss-arm-knife of PBX functionality.  It can act as a gateway, but it need not.  I think it is almost exactly what you want, except it can be a little bit of a challenge to config.  Things like Asterisk@home try to mitigate this problem.

nevynxxx:  What do you mean:  *Quote:*   

> All the major Switch makers produce kit that will do this. 

   Can you point me to an example?

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## RBH

 *BigBaaadBob wrote:*   

> I think it is almost exactly what you want, except it can be a little bit of a challenge to config.  Things like Asterisk@home try to mitigate this problem.

 

To clarify: can I set up asterisk@home, take a piece of software like Ekiga (I'm very new to non-mainstream VoIP) and connect two clients to the server, and have them talk to each other?

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## BigBaaadBob

 *RBH wrote:*   

> To clarify: can I set up asterisk@home, take a piece of software like Ekiga (I'm very new to non-mainstream VoIP) and connect two clients to the server, and have them talk to each other?

 

Yes.

Good reference is http://www.voip-info.org/wiki-Asterisk

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## nevynxxx

 *BigBaaadBob wrote:*   

> nevynxxx:  What do you mean:  *Quote:*   All the major Switch makers produce kit that will do this.    Can you point me to an example?

 

All of the major switch vendors make switches/routers that are capable of being a stand alone VOIP system.

Start here for cisco ( Also this(pdf) for a breakdown ). 3Com do similar....

Or have I got the wrong end of the stick?

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## BigBaaadBob

 *nevynxxx wrote:*   

> Or have I got the wrong end of the stick?

 

Oh, I understand: yes you've got the right end of the stick.

Currently the market is flodded with VOIP equipment and services vendors.  Some (like Cisco's "LinksysOne" brand) are targeting the "pro-sumer" or microenterprise, some the SMB, and others (Nortel) the large enterprise/government space.

Also, there is a healthy ecosystem of tiny folks who manufacture Asterisk-based solutions.

And, there is another route for getting this into your business: many IT outsourcing folks (even the one-man-shop ones) are implementing Asterisk for their customers as part of their IT offerings.  Actually, that is a good way to see if your IT folks have a clue: ask them about Asterisk.

And, of course, there is good old Gentoo and portage asterisk: that works fine for me!   :Smile: 

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## SeeksTheMoon

I am interested in using asterisk as a VoIP server in my LAN but I didn't find a howto for that, every howto focuses on some real telephone/isdn/mailbox/fax-interface-stuff which I don't want/need.

The asterisk book only tells me about VoIP protocols and codecs, but I already know about this stuff and then, when it comes to configuration for VoIP, it ends...

So how do I have to configure asterisk vor LAN-VoIP-only?

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## BigBaaadBob

 *SeeksTheMoon wrote:*   

> So how do I have to configure asterisk vor LAN-VoIP-only?

 

Unless I misunderstand your issue, any of the references you mention (and the Wiki I point to above) would work.  You simply ignore  all the stuff about trunks (IAX, SIP, or otherwise), and do the dialplan and SIP setup (assuming you will use SIP devices or softphones) and voicemail and other stuff just the same.

Trunking is a relatively minor part of Asterisk config so maybe you are confused by all the VOIP stuff, but you'll need to understand at least some of that to get a working config.

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## Ezhdeha

What I would do in your situation would be to setup an Asterisk server on one of the machines in the office (ideally you would want to get a stand-alone machine for this task, but installing Asterisk from portage also worked first time for me).

Personally, if it's your first time, just get Trixbox http://www.trixbox.org and run it up on an old machine, configure it with a static IP address, then go into your web browser and start configuring extensions.

If you want to have a physical handset on each desk in the office then I highly recommend the Linksys SPA-942, or if you want a budget model go the budgetone 200. For softphones just go to http://www.voip-info.org and search for softphones. I recommend Ekiga for linux and X-lite for windows.

However, assuming that this is an office and that you already have a phone on your desk, why would you not go for the complete VoIP replacement ? You can make that Asterisk box sing and dance in ways your current phone system could ever imagine. Be it either as a PSTN gateway or a VoIP over broadband system.

Best of luck, I'll give you whatever help I can[/url]

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## flybynite

This setup is so simple it gets left out sometimes.  

1. emerge asterisk

2. edit sip.conf to add your software phone.

Enjoy the asterisk demo functions.  You can make some calls to asterisk and hear a demo  or connect to the digium demo over the net.

Add another softphone and then you can call between softphones after changing extensions.conf

Check this users experience out.  Just ignore the install and voip provider info.

http://arstechnica.com/guides/tweaks/voip.ars/2

 *Quote:*   

> Out of the box, Asterisk is configured to provide a demonstration environment. So I started up Asterisk, by running the command asterisk vvvvc. This will start Aterisk and provide a console with debugging turned on.  The only thing this Asterisk demonstration environment was missing was a softphone and an actual account on the system. So I dug in and edited the sip.conf file since I would be using a SIP softphone. With the setup I had planned, all of the phones in my house would be a different extension. So you can use your imagination when setting up asterisk. Once you work your way through all the examples inside of sip.conf, adding a new extension is pretty simple
> 
> 

 

Here is part of a howto that shows detailed info how to setup a softphone and get a dial plan to connect them.  Just ignore the proxy and the fwd part  :Smile: 

http://www.automated.it/guidetoasterisk.htm#_Toc49248766

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