Can I install it on this server?

[WM: Not recommended. We’ll have a new one soon. 🙂 ]

[WM: There was a hiccup at DreamHost. Should be squared away now. Sorry.]

I can bring up the FreePBX GUI while on my home network.. but when it comes to accessing the interface "externally" (when away from home), or knowing what to fill in for the "SIP Proxy" field – I am left clueless.

I know my actual public IP address (70.xx.xxx.xx) but, simply typing it in my internet browser, or using it in the SIP Proxy field will not do anything.

[WM: Visit the forums. That’s what they’re for.]

Is this a noob mistake? Am I missing a step or two along the way?

[WM: We’d recommend sticking with numbers found on standard telephones.]

[WM: 1 number = 1 call. Nothing technically precludes more than one number using IPkall.]

Within GV:
I set up GV with the Gizmo account
Set the Gizmo account to receive GV calls

In Gizmo
Forwarded all calls to gv-ringback@my.machine.ip.address.

That’s all I did differently. Everything else worked like a charm and your instructions were perfect! I’ll be posting my alternate setup on my blog.

[WM: Try removing the FreeNum Trunk sequence entry in the the Default Outbound route. If that doesn’t fix it, head over to the forums. Be sure to indicate whether you’re using SIPgate or IPkall for your DID.]

Log into your server as root and issue the following commands:

cd /root/pygooglevoice
yum install mercurial
rm -r pygooglevoice
hg clone https://pygooglevoice.googlecode.com/hg/ pygooglevoice
cd pygooglevoice
rm /usr/bin/gvoice
python setup.py install

[WM: The default setup for the Orgasmatron V provides Google Voice dialing as the default outbound route for all 10-digit and 11-digit numbers that are called. The GV prefix works as well.]

I just wanted to report that there is 1-2 second(s) delay when using IPKALL-GV-Asterisk setup, kinda similar to steve (above) reported

Great installer!! I’ve used heaps of your wisdom already!

One quick question though:

I came from TrixBox, and when creating an inbound route I was able to select the destination to go straight to someones voice mail. Am I missing a FreePBX module somewhere? Your help would be greatly appreciated. -JD

[WM: Create an inbound route for the DID, and one of the destination options will be to send the calls to voicemail. Just choose the desired voicemail extension.]

i’m a PBX newbie but just a couple tips for those who make assumptions and are careless readers like myself– heres where i screwed up:

1) Under section labeled "These next three steps are important" there are 3 steps listed. Note that steps 1 and 2 are NOT identical although my careless readings had me convinced they were.
Worse, I had rationalized why they were the same — I assumed that after updating the modules, you must need to update them again due to newer modules’ dependancies. This did in fact seem to be true but regardless, go and download all the modules.

2) For those installing into VMWare / and/or headless. I got in via SSH, but then something happened and i could no longer get in, nor could I get in to FreePBX web admin page. Somehow IPTABLES/FAIL2BAN got turned on. Probably by installing all the modules. Apparently these 2 scripts are somewhat aggressive and locked me out of even SSH and even web access (possiblt after just a failed auth). I turned them off from the shell or so I thought. Perhaps a reboot brought them back. But my IP was specifically once again excluded in a IPCHAIN. Don’t know why it got added, I’ll assume for now that that is FAIL2BANs doing. I see why this is turned on for most people who are out on the internet, but for home behind the firewall use you probably want this off, at least while testing.

3) Despite Ward’s clear instructions on tuning the vmware kernel, I failed to select the new kernel on reboot resulting in the same choppy audio I had previously, so be sure to pick the new kernel from the booup list as he said

Other notes – I’m getting about 2-3 second delay on audio using a softphone with the GV method and Sipgate. ie: I talk or they talk, and we hear each other 2-3 seconds later. Like a highly improved version of talking to someone on the moon. Don’t know if this type of latency is par for the course as of yet, but the GV/Sipgate integration does work, but only seems useful for experimentation at this point.