Full VoIP video tutorial out if you want unlimited GV calling.

I thought you might be interested in a complete step by step tutorial on how to get your own FreePBX/Asterisk server up and running with Google Voice I just uploaded. I worked pretty hard on them.

Nobody else has done it with PBXes/Google Voice/CSipSimple/Android and recent versions of PBX In a Flash until now. Some people on these forums asked me to create the tutorial because they are struggling and although there is good information out there, it's spread across multiple sites, and many don't have the time to research it all. All videos in the series are viewable in 720p, 1080p, and 1200p.

I was inspired to make them because I see a lot of frugal people struggling with setting it up despite all the tutorials here. I think many people learn better visually and aurally, so I hope many will benefit from this.

Feel free to check it out.
Have a nice weekend.

FreePBX VoIP Tutorial
12-part YouTube playlist: http://www.youtube.com/watch?v=u9DzN1Pu6-Q&list=PLE_de-PBwrTSUMm-Y48aiOOHt_YyT69t0

I can't believe how good my battery life is now on top of how good the calls sound. Screenshot: http://i.imgur.com/DmR4KAn.png

Updates and additional notes

5/22/2014: GV still works and new versions of Incredible PBX
Since Google Voice is still working, you might as well take advantage of it and set up a PBX server for free unlimited calling in the US and Canada. You can even use softphones like Jitsi or Zoiper to connect to your PBX server and make calls through your PC instead of your Android device if you like.
FreePBX/IncrediblePBX has been on a bumpy ride lately. They went from using CentOS to some other Linux distribution called "Scientific Linux"
or something and now it seems to be back to CentOS--or some version of it. In my tutorials I recommend installing the .ova file of PIAF 2.0.6.4.3, but now it's up to 3.6.5.64. I just installed it and it's virtually identical. http://sourceforge.net/projects/pbxinaflash/files/PIAF-3.6.5-VirtualMachine/

5/22/2014: QoS
So it turns out there isn't a single video tutorial on YouTube that shows you how to set up Quality of Service (prioritized bandwidth in your router), so I think I may slap one together. We're at the point where QoS is almost a necessity, because when you're running a VoIP server and someone else on your network starts uploading to YouTube or watching Netflix, your VoIP calls are going to become jittery and your VoIP will be rendered useless for that time. Setting up QoS completely mitigates the hogging of bandwidth by any one service, port, protocol, or device--depending on how you set it up. I envision a tutorial that will take multiple real-world scenarios into consideration such as Xbox Live, Netflix, YouTube uploading/download, Usenet, Torrents, and big file transfers and show you how to configure it so that you maintain a 20ms ping to Google or your online game while your roommate is torrenting his rear end off.

12/8/2013 Update about XMPP coming to an end on May 15, 2014 and effectively rendering these tutorials almost useless.
Read more on page 11 of this thread because I posted some links.

6/10/2013: Verizon and the NSA
In light of the Verizon security issues with people getting very angry about the collection of metadata, I want to remind you that the Android app I recommend in my video tutorials, CSipSimple, supports ZRTP encryption. Assuming both parties have it enabled on whatever calling software they're using whether it be CSipSimple or Jitsi, you will be able to establish an actual secure call.
More info here: http://www.dslreports.com/forum/r28352878-U.S-VoIP-Privacy-Concerns
and here: https://guardianproject.info/2012/02/22/free-sip-providers-with-zrtp-support/
and here: https://jitsi.org/
Secure VoIP FAQ: http://wiki.ictd.asia/Secure_VoIP_Discussion_and_Tips


5/29/2013: Google Hangouts
Do NOT upgrade Google Talk on your device to Google Hangouts. It will screw up your ability to make calls.
Source and more info: http://www.dslreports.com/forum/r28309379-General-Google-HangOuts-vs.-XMPP

5/10/2013: Correction in the tutorials
When adding an extension in FreePBX, set "nat" to YES from the default. I made an annotation of this but it seems many people have annotations disabled.

Part 11 is up: Setting up multiple Google Voice phones

Well May 15th came and went and Google Voice is still working thus my tutorial is still useful. Not bad!

Since no QoS (quality of service) videos exist on YouTube for Tomato firmware, I think that's going to be what I do next. People deserve to be able to set up a meaningful and practical environment on their network in a short amount of time without having to earn a PhD first.

In my videos, I recommend and install PIAF-2.0.6.4.3-VirtualMachine from 4/25/2013. I just wanted to let people know that it's safe to install the new version "PIAF-2.0.6.4.4-VirtualMachine" from 6/11/2013 here: http://sourceforge.net/projects/pbxinaflash/files/PIAF-2.0.6.4.4-VirtualMachine/

I was having trouble receiving calls when I did a bunch of module updates in FreePBX + updates in Webmin (http://yourserver:9001) and was messing around with time server sync which was probably a mistake, so I started completely from scratch today with the new version.

There are a few differences in the updated version. For one, it's only 32-bit, not 64-bit. Second, it's PIAF Green and not Incredible PBX so you'll lose some features such as faxing I think. I haven't tried it, but apparently you can run a script and get all of the additional features IncrediblePBX has by reading this link: http://nerdvittles.com/?p=5844 (scroll down to "Adding Incredible PBX 11 and Incredible Fax")

After installing fresh, I followed my own video exactly and was up and running in no time using the same exact method so the information in the video is still valid.

I still feel strongly about how well this method has worked for me and I still want others to get into it. The only thing I do on my own server that I haven't made a video about yet is change from the default 5060 port to something else which is a little tricky to do. I started to get a lot of what looked like scan/hack attempts in my CDR Reports log and so I've blocked the default port 5060. I've not had any further hack attempts with the new port.

It's still fantastic being able to make free VoIP calls in HD quality! Since I've made my tutorial, I've found that Google Voice actually natively supports the G.722 codec instead of just PCMU like I thought. I would like more input from those who have gotten theirs set up and to contribute any additional useful information so we can all have a smoother experience with VoIP.

luisenriquereyes said:

HowDoIVoip, Last night I followed all your video tutorials and this morning went through them all to see if I missed anything.

The one thing that has not worked is incoming calls when my phone is connected to Wifi.

The call comes in but when I pick up the call the info status just says CONNECTING. The caller eventually goes to voice mail.

When not connected to Wifi the phone receives and make calls with no problems.

On Wifi outgoing calls work perfect.

Picking up on Wifi is the only problem I am having.

Any ideas what it might be?

Sent from my SGH-T889 using Tapatalk 2

Click to expand...

Click to collapse

It's hard to just guess without being able to see the screen/settings. One thing you could try is log into PuTTY, type asterisk -r and then type xmpp set debug on

you will see all of the google voice activity in the window in real-time. This is different than asterisk -rvvv that I talk about in video #11.

Make sure the ports I mention in video 5 are set correctly. Those all have to do with NAT/Routing. Hopefully your server machine isn't blocking ports if it has a software firewall. You could try adjusting the Static IP to Dynamic IP in Asterisk SIP Settings in FreePBX.

If nothing works, then I guess it's time to install a screen-sharing program like Skype or Jitsi so I can find the problem in 10 minutes. It's hard to guess without actually being able to see everything going on. That's why I hate text so much and prefer voice/video for technical support and tutorials.