Ah I see! anyone can contribute to the wiki. I just assumed it was maintained by the FreePBX team.
Anyone can, you usually just have to mention it to us first and then we add you, anything is fine as wellā¦ unless you add that someone has to install make samplesā¦ 
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LOL, no. The whole reason I wanted to try the guide for Debian was to see if I could follow it while using Nafs Asterisk, and NOT have to use āmake samplesā
That is really the only thing left to fix on this Ubuntu guide. I am hoping if following the Debian guide works that I can figure out why it does not install cleanly with the Ubuntu guide.
Thanks, will try Debian again soon as I get a little time, probably tomorrow.
Iām sure eventually @xrobau and I will get some time to test as well. However, as I said before, we do the install of FreePBX monthly in a docker container and havent have to do make samples.
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If you are able to do this and if you use Debian Stretch, please keep track of any differences from the guide for FreePBX on Debian 8.8 Jessie. As others have said that guide needs updating anyway, but I think you may have an easier time making it all work in Debian Stretch than in Ubuntu for a couple of reasons.
First of all I believe there are only minor changes needed from the Debian 8.8 instructions, some of which have been mentioned here and on DSLreports.
Second, if you know how to read shell scripts you could always look at RonRās install script for the Raspberry Pi, which installs Asterisk and FreePBX to Raspbian stretch. Iām not suggesting you blindly copy that because there are probably significant differences between Debian and Raspbian due to the ARM processor in the Raspberry Pi, but if you get stuck somewhere along the line using the Debian 8.8 instructions, you could maybe try to figure out what RonR did at that point and do the equivalent in Debian.
Hope you are successful!
FWIW, Iām pretty sure thereās no longer a need to compile Asterisk from source - all distros package it, and keep it reasonably well up to date.
except when using Nafās version of asterisk, which is not yet part of the official Asterisk code. (for the new GVsip)
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@2devnull I was getting that exact same error today after doing a git pull for Nafās latest changes. Can you check your logs and see if that error is still showing up in your logs? I was also noticing that my GV trunks were registering every 10 minutes. Appreciate any Response, that way I can provide more information to Naf.
I still just canāt wrap my head around the extent people are going to for a service that has no guarantees and could die the next week because the solution so far is pretty much a āhackā.
Everyone got excited when tm1000 said this might become a module in FreePBX and now are sad because Sangoma made the right choice and decided to not make this a module in FreePBX. For all sorts of reasons.
But letās step back a bit and Sangoma did release a GVSIP module for FreePBX. You now have to keep in mind that SNG7 is now the sole supported distro/release of FreePBX. While 13 is getting security and bug fixes this would not qualify as either of those. So it wouldnāt (shouldnāt) be back ported to Distro 6 (FreePBX 13) releases. That would also mean nothing below 13 would get updated either.
So again, if if a module came out tomorrow for GVSIP on FreePBX all the people still sitting on older versions of FreePBX (and there are plenty) would still be left holding the bag as it wouldnāt be available for for them. Not really the global solution people would be looking for on this matter.
Letās be honest here, the posted guide is not going to be for the average user that downloads and installs FreePBX. Most barely know how to SSH into the server let alone be able to do a manual install of FreePBX. Hell, Iām not sure of the numbers but it seems the PBXact installs are growing pretty well which is a commercialized version of FreePBX. No module, no Google Voice because it will break all sorts of stuff you paid for in PBXact including being supported.
To recap, both Sangoma and Digium are pretty clear (at least for the present time) Google Voice support is gone for them. The only way to continue to use Asterisk/FreePBX + Google Voice is to use a forked version of Asterisk and a manual install of FreePBX. Which now makes this essentially an Expert Level process that will result in lack of support from both Digium and Sangoma for issues or bugs. So these are things that any āofficialā guide for this process should have stated in it, in big bold letters so people donāt miss it.
Wellā¦ Duh?
I think itās pretty clear that everything described here about Google Voice and patched Asterisk builds is for hobbyists and home users who arenāt using FreePBX as their business phone system.
This thread is about the open-source, hacker-friendly, experimental side of Asterisk and FreePBX, not the commercial Sangoma side you buy when you want a stable phone system for your 100-employee office.
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Nor can I, but those who willingly choose to jump thru GV hoops, do so with no detriment to anyone other than their own time.
naf has already submitted (as least some of) the gv mods to Asterisk for inclusion, and many have already been approved:
https://gerrit.asterisk.org/#/q/owner:naf%2540ou.edu
If and when these mods have been merged, it is likely that FreePBX GUI support will follow in some form.
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Iām no longer getting that issue (also have not done a git pull yet per your latest instructions). Only ERROR (every 2 minutes in log) still which according to naf is harmless and should be fix now with his latest commits is this:
[2018-07-19 13:18:58] ERROR[22517]: pjproject:0 <?>: sip_msg Header with no vptr encountered!!
I do not think I am seeing any frequent registrations. Seems to be once every hour of 90 min assuming this is the line in question when it does:
DEBUG[22517]: res_pjsip_outbound_registration.c:648 registration_client_send_manual: Registration re-using transport
EDIT: I had also posted above in my update to that error of some things I added to the config file. Perhaps doing that and rebooting may set things straight.
Yes, I think this is not for the mainstream. The only reason I still need GV is that my (and probably most) VoIP providers starts wanting to be paid for excessive outbound 800 calls (which I am on pretty much all day). Also, Canada calls via GV is free which we do tons of also. For a home user this adds up and I agree with a bit of technical know how and some Linux experience guides like this allows us to leverage what is out there for free. If it goes away, then too bad, weāll have to pay but why leave it on the table now if forking, patching etc. that takes an hour the most can give you some financial relief.
Do you build it from source when building in Docker? I tried to do this particular guide in Docker but the ncurses or whatever pop-up GUIs (ITU-T etc) created havoc. If you do build from source in a Dockerfile, appreciate if you can share. Thanks.
Providers do not charge the caller for Toll Free calls. Thatās why they are Toll Free. The owner of the Toll Free number pays those charges. If you call Sangomaās Toll Free number, Sangoma is paying for that call not you. Additionally, most ITSPs include Canada in their standard rate. Itās generally only the Yukon, Northwest Territories and Nunavut along with Alaska and sometimes Hawaii that have a higher cost or offshore US Territories.
If you donāt do excessive, then yes. But once they move you to premium for doing excessive you do get charged (and hours of use per day at 1c/min adds up):
Not sure about getting free calling to Canada from your VoIP provider but mine charges for that too:
Edit by xrobau: Embedded images, to save you a click. Original links below:
https://i.imgur.com/GIhZMoO.png
https://i.imgur.com/moupZ3s.png
You need to get a better carrier. Did I mention that if you use SIPStation, you directly support FreePX?
Happy with rates of current carrier especially for intl. Some countries I call a lot SIPStation is a whopping 25 cents more per minute. Iām happy with what my current carrier offer and they have been reliable and always available to answer and help with questions. Couple with GV, tremendous savings from what telco was charging.