How-To Guide for Google Voice with Freepbx 14 & asterisk gvsip, Ubuntu 18.04

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That is not a guide for installing Freepbx, its for Incredible PBX, and yes I agree, do whatever you can find that works!

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) :wink: 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 :slight_smile: ) just no commercial module support, that will just have to be be a “wait and see . . . . .” from Sangoma)

(and no ‘make samples’ involved :slight_smile: )

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.

12 posts were merged into an existing topic: Asterisk Patches to Digium

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?

Yes I have setup 3,

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.

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cool…btw, I didn’t see you recommend the backup and restore module. Would it work on this?

I have not tested it, but I would assume that MOST modules should work without issue. The ones I am using so far seem to be working fine. (I usually just backup the entire hard drive, with a backup utility such as Redo Backup and recovery, or Borg, DD, etc)

yes, I run my pbx in esxi so I can backup the VM but just wanted to backup the files in this tutorial (both the new ones and whatever the gui entries wrote).

BTW - do you see any reason g729 codec wouldn’t work? That’s one of the reason I want to rebuild is to include it.

The codec should work but gvsip only supports ulaw and opus, and for a codec to work, both ends must be capable of using said codec. So 99% of the time that will be ulaw.

however extension to extension you should be able to use any codec, and opus will also work under some conditions, when both ends are able to make use of it. (page 14 and 15 of DSL reports thread talks about the HD voice, user Sasa said they get HD voice under some conditions, which means the OPUS codec.)

yes, voip.ms supports g729 and that is my primary inbound. GV is mostly for all outbound except for overseas.

BTW - how would I add opus in? I see on the Asterisk SIP Setting --> General SIP Settings opus is there as well as g729 but when I select g729 for example I get errors and things break. All I had done is put the ‘.so’ file for g729 in the respective modules directory but I assume more would need to be done.

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You have to build that against asterisk.

Which means it was probably in the ‘make menuselect’ section?

No. It’s more complicated than that. g729 was patent encumbered. So it’s not currently included with Asterisk.

Also g729 is really old and horrible… @xrobau wanna comment?

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Thanks! Everything I have read about OPUS makes it sounds like in an ideal world, we would use it for everything.

It’s even used in WebRTC audio. It’s the preferred codec for Google.

I literally just bought the domain ‘StopUsingG729.com’ because people keep thinking it’s great. It’s not. It was great 15 years ago. Now, just use G722 (or opus) and move on with your life. No-one cares about 3kpbs when the slowest speed you can get is 1mbit+.

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A post was merged into an existing topic: Asterisk Patches to Digium

A post was merged into an existing topic: Asterisk Patches to Digium