VOIP solution for home :-)

Message

mbjr · Post by **mbjr** » Sat Aug 26, 2006 1:16 pm

Hi Guys,

Can you suggest a farily working way for a home/family voip solution?
The idea is to have a local server/client installation where family members can connect to and make calls to each other w/o involving a 3rd party service provider.

Any good stuff out there?

Thanks in advance,

mw007 · Post by **mw007** » Sat Aug 26, 2006 5:27 pm

When I ran my own person VOIP service, I used ventrilo. The only problem is that it doesn't have a linux client.

TeamSpeak is also good I hear. I've tried it, but couldn't get it to work with my mic.

KlausU · Post by **KlausU** » Sat Aug 26, 2006 8:55 pm

Asterisk will do a great job if you give it some time...

mbjr · Post by **mbjr** » Sun Aug 27, 2006 6:14 am

Thanks for your replies.

The only problem is that it doesn't have a linux client.

Yeah, that strikes the option :-/

Klaus, I'm looking into asterisk, thanks for the tip.

Cheers,

John5788 · Post by **John5788** » Sun Aug 27, 2006 6:54 am

skype under wine works perfectly

linuxtuxhellsinki · Post by **linuxtuxhellsinki** » Sun Aug 27, 2006 12:50 pm

John5788 wrote:skype under wine works perfectly

So you mean that Linux version doesn't work

(why ?)

etrek · Post by **etrek** » Sun Aug 27, 2006 3:46 pm

before getting overwhelmed by things Asterisk you might want to try TrixBox (the successor to Asterisk@Home). I hear it's easier to setup and administrate..

http://www.trixbox.org/

Cheers,

E.

mbjr · Post by **mbjr** » Mon Aug 28, 2006 9:37 pm

Skype is a no-go, as I can't run my own free skype server in-house

Trixbox might be one -> any ebuilds available?

Thanks,

etrek · Post by **etrek** » Tue Aug 29, 2006 12:42 am

I think you have to download the ISO and install it on a separate box (or vm but not sure about the speed/timing). I was looking at it from an ease of use kind of a thing.

If you want it on your Gentoo box then you have to emerge Asterisk (which is masked) and work out the configuration manually. I've done several installations and it appears to work fine - I would recommend installing Asterisk in 32bit Linux (or compatible libs) as it seems more stable.

Here's where you can get a free copy (available under creative commons license) of "Asterisk: Future of Telephony" which gives an excellent introduction to things Asterisk.
http://www.asteriskdocs.org/modules/tin ... .php?id=11

Good resource for general asterisk info including commands etc:
http://voip-info.org/wiki/
http://www.asteriskdocs.org

There is a web management/config front end called freepbx (formerly ampportal) written in php - I haven't used it yet and there doesn't appear to be an ebuild but you can download it here:

http://sourceforge.net/projects/amportal/

If you need an Internet registry provider service, you can sign up for a free FreeWorldDialup http://www.freeworlddialup.com/ internet phone #. A fair number of Asterisk examples I've seen use FWD so it's something you might want to do. Note: There is an extra step after you register with FWD to get IAX2 (the default protocol used by Asterisk) working, you'll have to check your setup options.

To call your server or to make calls from your server you'll at least need a softphone. A decent IAX2 softphone to use (available in Windows too) is Kiax - there is an ebuild but like Asterisk it's masked:

http://www.kiax.org/

***** Recommend you stick with IAX2 instead of SIP if at all possible. In my experience it is much easier to route through firewalls and NAT (requiring only 2 ports) using IAX2.

Unfortunately there are not alot of hard IP phones (currently in the US anyway) that support IAX - you really have to look.

Of course you can always check out Digium's (the creator of Asterisk) website at http://www.digium.com

Sorry to be so long winded - hope you find something in this that helps!

Cheers,

E.

TheWhiteKnight · Post by **TheWhiteKnight** » Tue Aug 29, 2006 12:50 am

How do these perform? Lag? Easy of configuration? How much b/w do they take up?

pent0z · Post by **pent0z** » Tue Aug 29, 2006 11:51 am

freepbx is a great piece of software, and has a *lot* of powerful applications, such as followme, ring groups, queues, low cost routing, time conditions, etc. you can use voip trunks or telephony hardware (as quad-port analog cards or quad-port isdn cards, or simply an inexpensive HCF-PCI isdn card - 15 -)

we have a lot of pbx boxes powered by gentoo-hardened and freepbx plus a lot of our personalizations/plugins and we are very happy...
you can get it at www.freepbx.org

(i suggest freepbx because asterisk is not easy to configure manually)

install is not hard, you can follow the setup instructions for CentOS (search "freepbx install centos" on google or www.voip-info.org) except that you can't (obviouvsly) use yum to install the dependances
you'll need php, mysql, pear, asterisk, asterisk-addons, asterisk-sounds, lame, etc and some use-flags)
in a home/soho environment, we tested freepbx in a virtual machine running on vmware-server on a Pentium 4 HT machine without problems (of course without telephony hardware, only SIP phones/softphones)

you can buy voip phones for 60-80 . the linksys SPA941 is not expensive and is suitable for home/soho. (or use softphones as xten - available on windows and linux too)

etrek · Post by **etrek** » Tue Aug 29, 2006 12:05 pm

Those are tricky questions and I may not have the best answers sorry!

AFAIK For the bandwidth it depends on your hardware and what standard analog signal compression or companding algorithms you are using (uLaw/aLaw/Speex etc) to communicate. The good news is for the most part (unless you want video) you are only streaming simple mono audio which doesn't require a lot of throughput.

My installation experience has been with a basic US cable broadband connection (something aorund 10-20Mbits down 2-4 up I think) and a spotty T1 line. The asterisk servers I've setup are only handling analog connections with occasiional routes across the internet for testing - like to and from FWD for example. I've used several different IP Phones including Yuxin (IAX2) and Aastra (SIP) on the internal LAN and those work fine. Externally (over the internet) I've tested the KIAX and IAXTEL softphones and Yuxin IP phones and they work as expected. For slower speeds your biggest issue might be echo (caused by the latency in the transmission) and/or poor quality (lots of static).

I am testing the Server to Server stuff now transferring incoming analog calls to extensions on another server via the internet. If you have a relatively fast (DSL speeds or greater) and reliable connection then I think you will have no trouble depending on what else you are doing.

In terms of resources on the machine itself, you'll have to look that up over at Digium as I can't remember what they are. I believe you don't need too much though as asterisk (in simple form) has been installed on simple wireless router devices like a Linksys WRT54g.

Lastly the basic configuration is complicated which is why I initially recommended TrixBox (which is a complete Asterisk setup with easy management tools included). There are several config files you need to be familiar with and the extension language (with commands etc) is quite extensive. The book I recommended in my last post helped me to understand the basics and I went on from there. My biggest issue now is to try and figure out what features are practical to use and where to go from here (also what numbering scheme to use for extensions and commands etc).

"Asterisk: Future of Telephony" in PDF form can be downlowded from here http://www.asteriskdocs.org/modules/tin ... .php?id=11

John5788 · Post by **John5788** » Tue Aug 29, 2006 7:53 pm

linuxtuxhellsinki wrote:
John5788 wrote:skype under wine works perfectly
So you mean that Linux version doesn't work (why ?)

linux version is older and somewhat crappier. windows version is up to date and is better. at least thats what i remembered

Fog_Watch · Post by **Fog_Watch** » Mon Oct 30, 2006 9:47 am

Hello PentOz,

The line you wrote,

pent0z wrote:you'll need php, mysql, pear, asterisk, asterisk-addons, asterisk-sounds, lame, etc and some use-flags)

is about the most comprehensive FreePBX installation documentation for Gentoo I've found. Its that last bit though (I'll have to warn you that I'm about to get a bit facetious here), that I'm not sure about, etc and some use-flags. Any elaboration would be appreciated. I'll summarise.

Otherwise I'll have to wade through that CentOS document.

Regards

Fog_Watch.

undrwater · Post by **undrwater** » Wed Dec 27, 2006 2:28 am

Quick addition:

Code: Select all

USE="posix"

when emerging php. The

Code: Select all

$./install_amp

requires it.

I have an asterisk@home box that I have yet to do any serious work on. I'll have to update it to trixbox.

I'd still like to get my gentoo server configured so I'm not running 2 machines 24/7.

EDIT:
I got freePBX working on my gentoo box. The CentOS instruction link mentioned above worked pretty nicely, though you have to change some settings that are Gentoo specific like the webroot directory (/var/www/localhost/htdocs). I also had to add apache to my asterisk group in order for freePBX to write the config files. After install you can tweak /etc/amportal.conf to your liking or if you missed something during the install script.

Let me know if there is anything I can do to help out.

donjames · Post by **donjames** » Mon Jan 25, 2010 5:46 pm

HI,

I am currently using Trixbox (http://www.trixbox.org) to run an Asterisk system.

http://ipcomms.net provides my DID service. Their tech support is excellent. Once you get your Trixbox computer up and running, they will do the configuration remotely.

My email address is donaldbjames@suddenlinkmail.com .

Let me know if I can help.

Sincerely,

donjames

Don James

donjames · Post by **donjames** » Sat Nov 05, 2011 8:00 pm

Hi,

I have just switched to http://www.voipstreet.com for my DID service. They have excellent tech support and will configure once you get it installed and on line.

The outbound charge is $.019 per minute and the inbound is free. There is no flat-rate charge per month. I pay only for the outbound calls that I make.

I am still using Trixbox (http://www.trixbox.org) as my asterisk box.

Regards,

Don James

eccerr0r · Post by **eccerr0r** » Sun Nov 06, 2011 3:25 pm

Could try media-sound/mumble

http://mumble.sourceforge.net/

I haven't tried it yet myself but it's been advertised as a Ventrillo/Teamspeak alternative. Except it's GPL.

Oops. Not only this is a necro, seems like I replied to spam...

Hypnos · Post by **Hypnos** » Wed Nov 16, 2011 5:48 am

donjames,

Is your setup a drop-in replacement for Skype, i.e. only a client is requires? Or do I have to run a SIP/IAX2 server locally?

If the latter, can I run it on my laptop from any location?

Also, do your DID providers support encryption?

Finally, which client do you use? SFLphone and Yate are available in portage (Yate also provides a server).

Cheers,

Hypnos