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 wouldnāt work? Thatās 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 ā.soā 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 āmake menuselectā section?
No. Itās more complicated than that. g729 was patent encumbered. So itās not currently included with Asterisk.
Also g729 is really old and horribleā¦ @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.
Itās even used in WebRTC audio. Itās the preferred codec for Google.
I literally just bought the domain āStopUsingG729.comā because people keep thinking itās great. Itās 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 question for the experts here:
Do you believe that we will get to a point where a āMotif likeā add on will be available for installation on top of the off-the-shelf FreePBX distribution or will we have to go through the custom builds that you guys have worked out? (Iām using the RaspPBX/Raspbian version if that matters)
Iām happy to go through the detailed steps to get my GV back up and running but if I can wait a week or month and then just push a button to download some new module that will do the trick Iād prefer to stick with something that is widely released instead of a custom install.
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Are there steps to do a git pull on nafās branch and rebuild to get latest changes or is a complete rebuild necessary?
@tm1000 had said they were going to work on a module for this, maybe he can chime in letting you know if that is still the plan.
So from the FreePBX side of things, and after talking with @tm1000 on this, we do not plan on releasing a FreePBX module for this purpose moving forward.
Now if the relevant patches for the various configuration settings are accepted back into Asterisk, we will look at making those available in FreePBX as we do with any other Asterisk configuration options.
@GameGamer43 thanks for the reply, so it would happen if Nafs code gets included with asterisk.
@wmjackson depending on how soon you need your GV trunks back up and running you could try this guide, or try the incredible PBX install.
@gamegamer43 - sorry to hear that. Would have been convenient to just replace the GV/Motif module with something for GVSIP but I understand. There are only so many engineering cycles in a day and you have to prioritize them.
@xekon - is the first post in this thread the ālatestā guide for getting this going?
If Iām going to start with a āfreshā MicroSD card in my Pi, whatās the easiest way to get this going? Note that Iām a linux novice and know enough to be dangerous about the linux command line.
The settings specific to Asterisk will be added, yes, but a module that sets up Google Voice using the unauthorized obihai (now owned by Polycom) connection strings to google servers will not be created. I hope you can understand the legal reasoning here. Remember that Polycom owns Obihai and Obihai has a private agreement with Google (the connection string starts with obihai.). There are legal issues if Sangoma does something with this string because we are not sure what the contract between Obihai and Polycom and Google is.
Iām not a lawyer but you most likely donāt have to worry about this as an individual because you donāt have money like a corporation would.
Greetings,
Need a little help here. I was a happy Simon Telephonics user for some time. I started to get the same issue reported to the google change and want to move over to the GVSIP solution but am confused about chan_pjsip. I currently use chan_sip for my Cisco 7970 handset devices. Can I continue to use both types or am I out of luck?