ALERT GV: The Sky Has Fallen... Really

wardmundy

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There seems to be a fair amount of interest in GVsip for the PogoPlugs and/or the ARMel architecture. Personally, I'm using a PogoPlug E2. I did basically what Ward suggests: Since openssl 1.1.0 is not packaged (that I could find), I downloaded the source and compiled it using the configure from the script. I then applied the patches to the latest Asterisk and built that. Everything looked clean (except that there were a lot of patch failures, but virtually all of them were to the READMEs). However, when I run this new Asterisk with the 1.1.0 openssl, I get SSL errors:

[2018-07-15 12:25:06] WARNING[1622] pjproject: SSL SSL_ERROR_SSL (Handshake): Level: 0 err: <337092801> <SSL routines-tls_post_process_client_hello-no shared cipher> len: 0

Googling this yields a number of hits, but so far I haven't hit on the magic bullet to make it work.

I'd rather use the NAF approach, but I may give the PY code a try, although I really haven't taken the time to understand it. Right now I'm using OBis for GV access which usually work except for 5-ring
answering machines and voicemails.

If anyone succeeds in getting NAF going on ARMels, please let me know and I'll do the same.

Take a look at the link commands in the CentOS installer which also compiles openssl from source. I'm guessing that you haven't matched all of those up correctly. That's what NAF originally had to do on our server to get things working.

If that doesn't do it, post a response on DSLR Forum and perhaps NAF will have a suggestion or two.
 

kdthomas

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Did a complete install with CentOS 6.9 and all three of my extension are working (Desk, IVR, and Fax). Just make sure don't run the "Install-GVSIP" script until after you run the Enchilada script like I did or you'll have to do it all over again as well as delete the duplicate Obi device entries in our Google Voice settings.

Big thanks to everyone involved in getting this working again.
 

tycho

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I did basically what Ward suggests: Since openssl 1.1.0 is not packaged (that I could find), I downloaded the source and compiled it using the configure from the script. I then applied the patches to the latest Asterisk and built that.

I didn't get past looking for (and not finding) 1.1 packaged, so you have progressed much further than I have. I might try the hints Ward offers above, looking at the Centos installer for clues. But I have no sense of urgency because PYGV is working very, very well. At the moment. Which might or might not last...
 

kenn10

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Interesting circumstance with Google voice using MOTIF on my PIAF. While I can no longer use it for voice communications (the audio path cuts off when the far end answers,) it works flawlessly with AdvantFax to send and receive faxes.
 

troysmoke

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Hi.

First off, thanks for all of your hard work on getting all of this working. I'm new to the world of asterisk/FreePBX/IncrediblePBX (just trying to help some family members overseas have a US #).

I did the "pioneer" build and was able to use the latest Raspberry Pi build. I tried the Pi-specific build with no luck, but the gui-build finally worked. I was able to get two GV trunks registered and say the "Obitalk device" show up in the Google end.

However, now I have "two trunks down" in the Dashboard. With a soft phone (701), I can call outbound on either trunk. Both show as registered with "pjsip show registrations".[edit: had a mis-configuration in the outbound dialplan). Now when I call outbound, I don't get any audio and the call ends after about 20 seconds.

I created outbound routes for both trunks, with dialplans. First trunk is *41, second is *42.

Anyway, I don't know where to go to get the trunks online. Is it just a bug in the dashboard?

Thanks for any help/advice.

Edit: So, I'm wondering if the dashboard showing trunks down is a bug. I was able to dial out on both trunks with the softphone and then call the GV# and the extension picked up the call successfully. Anyway, hopefully everything sticks and keeps working.

Thanks, troy
 
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restamp

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Unless I have misunderstood something, I have some bad news to report. Yesterday, I posted about some SSL errors I was seeing after adding the NAF updates to an Asterisk server running on an ARM-based PogoPlug. At the time, I believed it was due to a cockpit error of some sort. However, after spending a lot of time today trying to resolve it, it dawned on me that my VPS server, on which I had been successfully testing the new software, did not have any encrypted PJsip extensions. When I added one, it exhibited the same SSL errors as did the PogoPlug.

I'm afraid I am coming around to the belief that, in it's current form, encrypted extensions and trunks (at least SIP and PJsip, IAX seems to do its own thing) cannot co-exist with the NAF software.

I suspect the problem is with openssl 1.1.0, which the NAF mods require. Googling the error message turns up lots of hits, but nothing definitive. It's enough however to leave me wondering whether this could be the reason the various flavors of *nix out there have been slow to upgrade to openssl 1.1.x.

Has anyone who had added the NAF software (along with openssl 1.1.x) been successful with also running encrypted links? If so, I'd be delighted to hear from you and how you did it.

In the meantime, though, I'm afraid we're either going to have to come up with yet another Asterisk mod to accommodate the openssl 1.1.x changes, or choose between having encrypted links and the NAF GVsip trunks.
 
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restamp

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Hi.
Edit: So, I'm wondering if the dashboard showing trunks down is a bug. I was able to dial out on both trunks with the softphone and then call the GV# and the extension picked up the call successfully.
I've proffered the following fix be before but so far it has not been picked up and made official:

To resolve this, edit pjsip_custom.conf and add the following line after each occurrence of "contact=sip: obihai.sip.google.com": qualify_frequency=120

(This should occur in each gvsipX context of type=aor.)

Then reload Asterisk.

And while we're in the process of polishing the apple, it would also be nice if the GV number were coded in the custom trunk's CALLERID field when it is created.
 
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wardmundy

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@restamp You are probably right on the encrypted extensions. I will post this on DSLR Forum if you haven't already. I will try your suggestion with qualify_frequency although I don't think that's the reason the FreePBX Dashboard doesn't show GVSIP trunks. Also, the DID of every GVSIP trunk is coded in the contact_user field which makes it usable as the DID for Inbound Routes in FreePBX. Setting the CallerID doesn't have any effect with GV trunks since Google doesn't let you set the CallerID anyway.
 

troysmoke

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Thanks for the info. I may give that a try. I'm mostly just happy that the system is operational. :)
 

wardmundy

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I've proffered the following fix be before but so far it has not been picked up and made official:

To resolve this, edit pjsip_custom.conf and add the following line after each occurrence of "contact=sip: obihai.sip.google.com": qualify_frequency=120

Then reload Asterisk.

Right you were. Thanks for the fix. I've updated the installers.
 

kdthomas

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I've proffered the following fix be before but so far it has not been picked up and made official:

To resolve this, edit pjsip_custom.conf and add the following line after each occurrence of "contact=sip: obihai.sip.google.com": qualify_frequency=120

(This should occur in each gvsipX context of type=aor.)

Then reload Asterisk.

And while we're in the process of polishing the apple, it would also be nice if the GV number were coded in the custom trunk's CALLERID field when it is created.

I added this in and I went from zero trunks online to seven registered. I have three GV numbers configured which is strange that it reports seven.
 

Eliad

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Incredible PBX 13-13 with GVSIP-NAF for CentOS/SL 6 (beta)

For the pioneers (only), we're ready for you to take a turnkey version of Incredible PBX with GVSIP-NAF for a spin. You'll need a CentOS 6 (64-bit) minimal platform to begin. Then just follow the original Incredible PBX 13-13 tutorial substituting the download link below. When the install finishes, you'll have Incredible PBX Lean with built-in GVSIP support but no extensions, routes, trunks, etc. You can run the Enchilada upgrade at this time, if desired, to add 30+ Asterisk apps. It will erase all existing data including any preconfigure trunks, extensions, and routes.

Installation: To get started, login to your CentOS/SL 6 server as root using SSH or Putty. Then...
Code:
cd /root
wget http://incrediblepbx.com/incrediblepbx-13-13-NAF.tar.gz
tar zxvf incrediblepbx-13-13-NAF.tar.gz
rm -f incrediblepbx-13-13-NAF.tar.gz
./Incredible*

For instructions on obtaining refresh_tokens for your Google Voice trunks, go here. For instructions on creating routes, go here.

Add GVSIP trunks by logging into your server as root and running /root/gvsip-naf/install-gvsip. It's a 10-second install per Google Voice trunk. You'll need your refresh_token and phone number for each GVSIP trunk you wish to add.

Delete existing GVSIP trunks (#1 through #9) by running /root/gvsip-naf/del-trunk. If you need to delete trunks higher than 9, edit del-trunk and add new sections for the number of trunks you have. Always delete trunks in reverse order. We don't recommend deleting GVSIP1, but it works.

Add an Inbound Route for each of your trunks using the 10-digit DID of the GV trunk and specifying a destination for the incoming calls. Except in Enchilada version, inbound calls won't be processed until you add an incoming route.

Add an Outbound Route for each trunk (named GVSIP1 through GVSIPn. Outbound calls will fail until you add an outbound route for the trunk.

To refresh the patched version of Asterisk, copy /etc/asterisk/pjsip_custom.conf to a safe place, delete the contents of pjsip_custom.conf, rerun the installer, and then copy your version of pjsip_custom.conf back to /etc/asterisk and restart Asterisk. That way you won't lose any of your previously configured GVSIP trunks.[/QUOTE
 

Eliad

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I installed a fresh Centos 6.9 updated it and then installed GVSIP-NAF. Then I tried to apply Full Enchilada. Full Enchilada does not install because it requires GV OAuth to be installed first. Did I miss a step?
 

restamp

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Also, the DID of every GVSIP trunk is coded in the contact_user field which makes it usable as the DID for Inbound Routes in FreePBX. Setting the CallerID doesn't have any effect with GV trunks since Google doesn't let you set the CallerID anyway.
Yeah, I realize it makes absolutely no difference to Asterisk or Google, but it does make it clearer which GVsip trunk is which when navigating the Connectivity > Trunks page. That's the only reason I suggested it.
 

wardmundy

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Looks Like it is installing Samba , CUPS, and Gnome. Why all that?

This is what I am trying now ISO did not work with GVSIP , nor did Virtual image in Virtualbox

It just rebooted and disconnected me from ssh, now I have no idea what it is doing!

Not going to explain why we install the 1000+ packages that we do. Do some reading...

We are in transition from installers that create a Google Voice XMPP platform to one using Google Voice GVSIP which is a completely different technology. To use the old installers including the ISO, you currently have to run the new GVSIP installer once you have the original working platform to bring it up to speed if you want to use Google Voice. The alternative is to use the new beta installer from an existing CentOS 6.10 minimal platform. It sets everything up in a single install and will become the default installer shortly.

If you're having specific problems, open a new thread after reviewing your install logs in /root to see what failed. Saying the install "did not work" doesn't provide much of a clue to let someone assist you.
 
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wardmundy

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Reports this morning suggest that Google has finally pulled the plug on Google Voice XMPP accounts other than a few that already had connections up. In short, the next time you attempt to login to Google Voice through an XMPP connection, it will fail.

As a result of this development, we are immediately releasing the new GVSIP editions of Incredible PBX 13-13 for CentOS and the Incredible PBX 13-13 ISO. Tutorials will be updated, but we wanted everyone to be aware that future downloads after noon EDT today will feature GVSIP, not XMPP, so there will no longer be a need to subsequently run the GVSIP-NAF upgrade with these two platforms only.

Updates for Ubuntu 18.04 and the Raspberry Pi will be available soon. In the meantime, on the Ubuntu and Raspbian platforms only, you will need to also run the GVSIP-NAF installer after completing a new install.
 
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Sajid

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I have 2 Google Voice Trunks up and running. When making outbound calls, after the call is answered the quality is great for 15 seconds or so then it gets choppy and then it gets to normal and then again and so on...I have turned of fail2ban and iptables to test this the results are the same. When I run pjsip show channelstats, on the receive section I can see under lost column count is increasing...Jitter is at 0.001. I dont see this when I run extension to extension calling. I am running this on a VM behind an L3 router. I even changed the internet provider (I have 2) to eliminate carrier related issue. Everything else is working except choppiness on the google voice outbound calls. Any help or has anyone else seen this. This is running on google voice on PJSIP latest naf script. any help??
 

windpoint

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I had an installation of Issabel 4.0 with IncrediblePBX. Is it possible. Is there anyway I can implement GVSIP without installing a new OS? I actually started loving Issabel UI.
Can I use this method:
http://nerdvittles.com/?p=26315
Thank you Ward.

-Mony

Reports this morning suggest that Google has finally pulled the plug on Google Voice XMPP accounts other than a few that already had connections up. In short, the next time you attempt to login to Google Voice through an XMPP connection, it will fail.

As a result of this development, we are immediately releasing the new GVSIP editions of Incredible PBX 13-13 for CentOS and the Incredible PBX 13-13 ISO. Tutorials will be updated, but we wanted everyone to be aware that future downloads after noon EDT today will feature GVSIP, not XMPP, so there will no longer be a need to subsequently run the GVSIP-NAF upgrade with these two platforms only.

Updates for Ubuntu 18.04 and the Raspberry Pi will be available soon. In the meantime, on the Ubuntu and Raspbian platforms only, you will need to also run the GVSIP-NAF installer after completing a new install.
 

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