Thank you for your help! using root user did let it install, after installing the dashboard was clean and appeared error free, but lots of features were not working at all including pjsip, looking in the log reveals Errors talking about LOTS of different missing .conf files. I would venture a guess that about half the .conf files were missing.
For now I am just going to use a Patch file and apply it to an old Debian install. I really wanted to get this going for Ubuntu, but I just don’t have any more free time.
The other option is I could just use the sample config files, that install appears to be working completely.
Maybe I could take the time to learn about the config files, then I could alter the sample files to address any concerns.
Also I did alter the guide to remove the ‘make samples’ command, instead it just copies the .conf files, which does a little bit less stuff than ‘make samples’ would normally do, and I added a disclaimer to the top of the page.
I am curious about something with this whole GVSIP thing and FreePBX. I’m hoping some people can share their thoughts and perhaps satiate my curiosity.
This requires a “patched/forked” version of Asterisk. So that removes the use of any Distro or other versions of FreePBX (PBXact, etc).
In order to use this you now have to go the route of a Manual install, which is generally not supported. It also means that it is now up to you, the user, to install all the software/dependencies/etc that is needed for features in FreePBX to function.
So far this is tested and the “official” instructions are for Ubuntu 18.04. That is an “un-supported” OS, as in while it works Sangoma support isn’t going to jump all over issues or bugs because FreePBX SNG7 is based on RHEL.
As SNG7 is based on RHEL and this based on Ubuntu that means no activation of the PBX and no use of commercial modules as they require both RHEL and an activated system to be used. This also means modules like Phone Apps and EPM will not work despite even having Sangoma phones. Because those are commercial modules even though free for use with Sangoma devices they are still encoded like every other commercial module.
Official Asterisk releases are already lagged in availability in the Sangoma repos for official versions. Each time a new release of Asterisk hits people start asking “When will it be in the FreePBX/Sangoma repos so I can download it?”. This is a third party fork that requires a full manual re-compile to update versions. No RPM or script to automate this process for the user. This means a couple things A) You now have to wait for the fork maintainer to update their fork. B) You have to hope the changes to the fork are inline with FreePBX because those changes could break how FreePBX expects Asterisk to behave or where it expects Asterisk to have certain information stored and in what format.
Both Google and ObiHai have put some money and time into this new deal they got going on. ObiTalk devices are now Google Voice branded, that’s not cheap on Obi’s part to have. As you can see in the samples and how-to to make GVSIP work, you’re connecting to the Obi branded network for Google. Now either Obi has an exclusivity deal with Google meaning they are the gods of the SIP network for Google OR Google is forging partnerships with vendors like Obi and in six months we could see “grandstream.sip.google.com” hosts for GVSIP services on Grandstream devices. Either way, Google and/or Obi are invested in this new move which means when they see “Asterisk/FreePBX” User Agents and/or non-Obi devices on the network they could shut you down. Not saying it will happen but to not consider it’s a possibility is naive.
I mean this isn’t like Chan_Motif which is a full signed module in FreePBX versions, doesn’t matter if it’s a manual install or Distro/PBXact install. Chan_Motif was there.
So I am very curious as with all the items I mentioned above, is having a “free” service like Google Voice on FreePBX really that important? Is it so vital that users would be willing to give up large features of FreePBX for it? What will current GV users do? I’m sure there are plenty (as I have 1 or 2 clients with GV on their FreePPX) that have GV on a system running commercial modules. Is this worth changing completely or even worse (to me) trying to make it work on your current setup and tanking the PBX?
Now I’m not curious about “home users”, people running FreePBX and GV for simple and basic use. I’m taking businesses, because plenty of those use GV as a solution. Because any real business risk assessment would show that there is more risk in this new GVSIP path for “free” service than there is in paying an ITSP like $12/year for the DID and maybe $300+ year if you’re doing 1,000+ minutes a month. I mean doesn’t $30/month seem more reasonable than paying Bob in IT $30/hour to spend 1-2 hours every few months to fix problems due to updates/changes and, of course, the eventual downtime experienced due to this?
This can be made to work in other Distros if somebody does the leg work. For now you would need to apply the patch files to Asterisk, but Naf has plans to submit the code to Asterisk for review after a testing period, so the code would be included with official Asterisk if all goes well.
Yes, this requires manual install, but Incredible PBX supposedly supports this out of the box, however I have not test it. Also tm1000 is working on a module for this so that the _custom edits are no longer necessary.
Yes this guide is for Ubuntu, which is what I am wanting to use. It does make sense that Sangoma would support SNG7 foremost. That is OK, atleast for me. This is Open Source, so users can solve and contribute the fix for any bugs as well.
Personally I do not use commercial modules, For me this gives me a simple home phone system supporting less than 5 people at 2 homes. There are however patches available so this could be made to work with SNG7. It just wont be me that write a guide for that since I am using Ubuntu, and my old system used Debian.
Good point, and I think thats the root of some of my issues with the sample config files. I have had a couple users private message me and they are reporting the same issues that I have found, one of them solves some of the issues after the install by copying only the necessary sample .conf files, I may test this if I find the time.
Completely agree and I am fairly sure most people using this have thought more or less the same thing. However with zero feedback from google regarding the new gvsip protocol, its pointless to speculate. It may work for years, or it may break tomorrow.
This is not Chan_Motif, but tm1000 has said they are working on a module.
Depending on your requirements you are absolutely right, at the moment about buying a DID.
chan_motif worked for 6+ YEARS!!! I did nothing to that box, and had stable uninterupted service!
GVsip is quickly becoming stable, so it might be the same story with it, stable uninterupted service,
however any bugs have to be worked out, and it has to be implemented into existing solutions.
Yes, it now includes all my latest changes. However the installer now checks database permissions and it will not work when running non root using sudo.
Checking if this is a new install...Yes (No /etc/freepbx.conf file detected)
Database Root installation checking credentials and permissions..Error!
Invalid Database Permissions. The error was: SQLSTATE[HY000] [2002] No such file or directory
This might mean somebody on the FreePBX team is checking the installer script over, which would be so AWESOME!
EDIT: I was mistaken, found the reason for the new Error. When I updated the guide to include all of my latest changes, I used a block quote for the mysql section instead of a code block, and the forum code changes the quotes to the type of quotes that do not work for the config file.
The guide now has all of my latest changes, and I have tested it start to finish.
Was that not already in the installation guide for Debian ?
Why don’t you start with the already working system install on Debian or whatever, and just insert your compile, make, make install installation of the patched Asterisk bearing in mind that as yet the version announced is not yet conformal, (you kinda fixed that in an ad-hoc way), I just don’t understand your ongoing angst, it’s all been working for years . . . .
(and seriously , . . … There is no magic involved.)
Available packages change and sometimes disappear all together, because they are deprecated.
What works in Debian 8.8 is not guaranteed to work in Ubuntu 18.04.
Honestly though I am over it and no longer trying to do anything more than I have here.
The goal was to get Google voice working on Ubuntu with Freepbx and Asterisk, and I have inbound and outbound calls so I am happy. The install works with the sample .conf files in place.
I do however plan to look those files over to get an understanding of what the different values do, etc.
Then all you need to do is install the debian/ubuntu/centos/whatever recipe (they have(almost) always worked for me) (apart from the attempt to install all modules, commercial included, just be sentient there ) and then when you suddenly realize it actually works, THEN compile your (or his) patched version for gv
sideline " (is it really worth your effort when you can buy calls for 0.008 cents a minute or less and a REAL DID for a buck a month ?)"
and then pretend it is “unpatched” by it’s announced version , It really is a as simple as that, (honestly,. . . . believe me . . . , been there done it, many , many times, and over many years and many many versions, absolutely never get your head up the “make samples” non-sequitor )
Is there a known issue with outbound calls with this setup?
Getting:
Connected to Asterisk GIT-15-759c0b60f8M currently running on freepbx (pid = 1735)
[2018-07-16 21:34:34] WARNING[1817]: pjproject:0 <?>: SSL 6 [SSL_ERROR_ZERO_RETURN] (Read) ret: 0 len: 32000
== Everyone is busy/congested at this time (1:0/1/0)
– Auto fallthrough, channel ‘PJSIP/701-0000000a’ status is ‘CONGESTION’
UPDATE: It’s working. Not sure what I changed but been looking at Naf’s conf file and this one and trying different things as there are some differences. Also, added this:
[gvsip1]
type=aor
contact=sip:obihai.sip.google.com
qualify_frequency=120
some of the small differences between his config file and this one, is that this one more tightly integrates with FreePBX(thanks billsimon) so that you can make use of the FreePBX gui features, such as inbound and outbound routes, etc. His config file is setup for straight asterisk, without FreePBX.
Sorry, there I was confused, for the longest time I thought “Incredible PBX” was in fact just FreePBX+ ( at least the open source parts of it) the + being , well +, (yes there was some temporary confusion about 3CX and financing, but that link I posted will really get you FreePBX( 13) on pretty well any OS of your choice ) just no commercial module support, that will just have to be be a “wait and see . . . . .” from Sangoma)
Incredible PBX is FreePBX 13. It’s Ward Mundy’s “Spoon” (he doesn’t like to say the word ‘fork’ because he doesnt change the code, just the OEM branding and he removed signature signing) of FreePBX 13
At one point in time Incredible PBX did support commercial modules. But the PBX in a Flash team removed it and said they’d never add it back again.
OK, thanks…it seems naf doesn’t think that qualify_frequency thing does anything much (it was mentioned here btw: <sorry, forum said new users can’t put link. Anyway it is in the DSLreports link naf has at the top of his github branch>).
Good that it integrates with FreePBX more. I have wasted a lot of time this weekend chasing IncrediblePBX and haven’t been successful. That new one may work but I rather stick with this. Good if you can keep this updated as it doesn’t have all the noise (too many features in IncrediblePBX that I do not use). I must say though, the Wazo based IncrediblePBX which I was using for the last year was solid and lean (the rest not so…various issues)…wish the Wazo one could have had gvzip support but alas good things must come to an end.
Keep up this excellent piece of work…with billsimon and other supporting here it seems to be the go to distribution that actually works.
BTW - I assume this supports more than one gvsip accounts?
any section that says gvsip1, gvtrunk1, gvout1, etc. you would duplicate those instructions and put gvsip2, gvsip3, etc. and get your Oauth credentials setup for the other accounts. Also anywhere it mentions the extension, in most setups, you will have different extensions per GV trunk. So that different phones/extensions use different GV accounts.
) and then when you suddenly realize it actually works, THEN compile your (or his) patched version for gv