ALERT GV: The Sky Has Fallen... Really

billsimon

Well-Known Member
Joined
Jan 2, 2011
Messages
1,534
Reaction score
727
not even hitting my firewall as far as I can see. Do we have an IP address range that GV uses for obitalk devices?

That's not the way this works. You register and it keeps a TLS connection open, from you to Google. Outgoing calls and incoming calls go over that open socket. The firewall does not matter here, because you have opened the connection to Google and keep it open.
 

mcfuzz

New Member
Joined
Oct 1, 2014
Messages
24
Reaction score
4
That's not the way this works. You register and it keeps a TLS connection open, from you to Google. Outgoing calls and incoming calls go over that open socket. The firewall does not matter here, because you have opened the connection to Google and keep it open.

Got it - then why is google not sending incoming calls to me? Any ideas?


edit: now I am getting outbound congestion messages from Google :\


Whyyyy is thissss happpeeenniiinnng :(
 

billsimon

Well-Known Member
Joined
Jan 2, 2011
Messages
1,534
Reaction score
727
Got it - then why is google not sending incoming calls to me? Any ideas?
edit: now I am getting outbound congestion messages from Google :\
Whyyyy is thissss happpeeenniiinnng :(

Hope you realize the software you are testing right now is still under heavy development, not necessarily even beta level yet. You should probably be an Asterisk expert to be testing it. My advice to you is WAIT.
 

mcfuzz

New Member
Joined
Oct 1, 2014
Messages
24
Reaction score
4
Hope you realize the software you are testing right now is still under heavy development, not necessarily even beta level yet. You should probably be an Asterisk expert to be testing it. My advice to you is WAIT.

I know I know - my "why is this happening" comment was more aimed at Google disabling XMPP support - which, honestly, I can't blame them for since it's their product but still - causing a bit of a headache.

My asterisk expertise days are from a decade ago when I did hosted call center software - a lot has changed since then and the braincells responsible for asterisk have long ago faded...
 

jhankins

New Member
Joined
Jul 2, 2018
Messages
1
Reaction score
1
I was able to get inbound calls working more reliably by registering a unique username and client_uri in pjsip_custom.conf. Details are here over at the thread on DSLReports:
You are right! you have to make sure to use a unique Username/client_uri, Mine was just "gv1"

Updated my pjsip_custom.conf using my google voice number for uniqueness.

client_uri=sip:[email protected]
to
client_uri=sip:gv{googlevoicenun}@obihai.sip.google.com
username=gv1
to
username=gv{googlevoicenum}

{googlevoicenum} = your Google voice number

Basically, replace "gv1" with something like "gv{yourPhoneNumber}". It seems the uniqueness may increase inbound call reliability.
 

yozh

Member
Joined
Nov 9, 2010
Messages
71
Reaction score
3
Haven't gotten that far yet. Stay tuned or feel free to experiment.


Did a little expiremnets, and adding another entry while stay established, called out of the same number. I have to spend time with this, which I dont have now :(
 

jtpiano

New Member
Joined
Nov 30, 2014
Messages
7
Reaction score
1
NAF got us going with Incredible PBX on the RasPi platform this afternoon. So I'll write it up for next week. Exciting times!

I tried running the NAF patch on my RasPi 2 as of 6/30. So call me a "Pioneer" Everything seems to work except that I am having one way audio issues.

Let me give you a little more background.
I have a spare Raspberry Pi 2. I downloaded and installed incrediblepbx13-raspbian8-gvoauth.zip dated 2017-11-06 from sourceforge. I then applied the NAF patch.

GV was working previously on my old system (ver 12) until the shut off date. I hopped on the forums here when things went south and my GV line stopped working. I didn't make any changes in my firewall or ATA setups.

It seem the only problem I am having is the one way audio issue. If I make an outbound call, the person I am calling can hear me but I cannot hear them.

Having said all that, I realize it is possible that I have made a simple mistake or overlooked something. This is after all, a new setup. ;-) Hope this feedback helps.

PS- I might have found the answer to my own question - https://pbxinaflash.com/community/threads/incredible-pbx-for-raspi3.21872/page-3 on post #43 and #44. I did my initial setup with a different IP and then moved it. I'll check on this when I get home and report back.

Well... the report is in. No joy here. ( as of 19:35 Central)

Suggestions?
 
Last edited:

Jebs2k

New Member
Joined
Jun 30, 2018
Messages
8
Reaction score
0
I was able to get inbound calls working more reliably by registering a unique username and client_uri in pjsip_custom.conf. Details are here over at the thread on DSLReports:
You are right! you have to make sure to use a unique Username/client_uri, Mine was just "gv1"

Updated my pjsip_custom.conf using my google voice number for uniqueness.

client_uri=sip:[email protected]
to
client_uri=sip:gv{googlevoicenun}@obihai.sip.google.com
username=gv1
to
username=gv{googlevoicenum}

{googlevoicenum} = your Google voice number

Basically, replace "gv1" with something like "gv{yourPhoneNumber}". It seems the uniqueness may increase inbound call reliability.


it works! i made the edit and so far all incoming calls are working, good find!
 

kdthomas

Member
Joined
May 13, 2016
Messages
57
Reaction score
11
I hadn't noticed any change of service as I rarely use my home phone, but I tested today and it looks like something has changed and the sky is halfway down for me using the old GV (Motif). Sad day indeed. Inbound voice works great, which is the main thing I use it for, however, outbound is only one-way. They can hear me, I can't hear them. Strange that they left it in this state, but maybe they're not done removing it?
 

2devnull

New Member
Joined
Nov 13, 2013
Messages
15
Reaction score
0
So, this is the issue I am having also. Wish I found this earlier, would have helped with all the hours of troubleshooting earlier. So I am on Wazo and have the issue of one-way on outbound where recipient can hear me but I cannot hear them. Everything else works as usual for now. I guess the recommendation is to move off GV entirely?
 

wardmundy

Nerd Uno
Joined
Oct 12, 2007
Messages
19,168
Reaction score
5,199
So, this is the issue I am having also. Wish I found this earlier, would have helped with all the hours of troubleshooting earlier. So I am on Wazo and have the issue of one-way on outbound where recipient can hear me but I cannot hear them. Everything else works as usual for now. I guess the recommendation is to move off GV entirely?


Yes, unless you want to move to an Incredible PBX platform that supports the new GVSIP implementation. See Nerd Vittles for details.
 

2devnull

New Member
Joined
Nov 13, 2013
Messages
15
Reaction score
0
Yes, unless you want to move to an Incredible PBX platform that supports the new GVSIP implementation. See Nerd Vittles for details.
I love the simplicity of the Wazo UI and it has been rock solid but if there isn't going to be a GVSIP update for it soon, I may need to do as suggested.
 

2devnull

New Member
Joined
Nov 13, 2013
Messages
15
Reaction score
0
Well, I tried the Ubuntu 18.04 Incredible PBX but ran into the system freeze (unresponsive system) issue that seems to be affecting 18.04 systems.
 

TopMark

New Member
Joined
Jul 11, 2018
Messages
6
Reaction score
1
Spent endless hours trying to figure out the refresh token and how to obtain one. Wondering if someone could point me in the right direction.
 

2devnull

New Member
Joined
Nov 13, 2013
Messages
15
Reaction score
0
I am getting this error after running the GVSIP install script and asterisk doesn't start (this is on the Virtualbox RasPi platform using this: http://incrediblepbx.com/gvsip-naf-raspi.tar.gz):

[2018-07-11 15:48:25] WARNING[3310] res_pjsip/config_auth.c: Unknown authentication storage type 'oauth' specified for auth_type
[2018-07-11 15:48:25] ERROR[3310] config_options.c: Error parsing auth_type=oauth at line 37 of /etc/asterisk/pjsip_custom.conf
[2018-07-11 15:48:25] ERROR[3310] res_sorcery_config.c: Could not create an object of type 'auth' with id 'gvsip1' from configuration file 'pjsip.conf'
[2018-07-11 15:48:25] WARNING[3310] res_pjsip/config_auth.c: Unknown authentication storage type 'oauth' specified for auth_type
[2018-07-11 15:48:25] ERROR[3310] config_options.c: Error parsing auth_type=oauth at line 82 of /etc/asterisk/pjsip_custom.conf
[2018-07-11 15:48:25] ERROR[3310] res_sorcery_config.c: Could not create an object of type 'auth' with id 'gvsip2' from configuration file 'pjsip.conf'​
 
Last edited:

2devnull

New Member
Joined
Nov 13, 2013
Messages
15
Reaction score
0
I understand this is a moving target and bleeding edge, but does someone have some files which all work together for the dual trunk GVSIP? I moved from the Virtualbox image (based on the issue a few posts above) to a raspberry pi 3 and although it seems the upgrade took correctly, I can't seem to find that *48 in the extensions_custom.conf file as the instructions say and therefore GVSIP isn't working.

Error when making outbound call is:
NOTICE[3975][C-0000000e]: chan_sip.c:26472 handle_request_invite: Call from '701' (192.168.1.17:5060) to extension '*481233456789' rejected because extension not found in context 'from-internal'​
 
Last edited:

wardmundy

Nerd Uno
Joined
Oct 12, 2007
Messages
19,168
Reaction score
5,199
GVSIP-NAF-GUI (beta)

UPDATE: Please follow the official Nerd Vittles tutorial now rather than relying on this beta version.


For the pioneers (only), we've got a new GVSIP-NAF implementation specifically tailored to management within the FreePBX GUI. It should work with CentOS/SL 6.9, Ubuntu 18.04, and Raspbian 8, not RASPBX. Works better with Incredible PBX because we know what's been installed.

Translation: You are responsible for creation of Inbound and Outbound Routes to manage your GVSIP trunks, numbered 1-n. The installer will handle creation of the GVSIP trunks themselves. It takes about 10 seconds to add a new GVSIP trunk. You can add as many as you like.

Overview: Running the installer (install-gvsip) the first time will get your Incredible PBX platform up to speed by installing the correct version of OpenSSL for your platform. Then it installs and patches Asterisk 13.21.1 13.22.0 to support GVSIP Google Voice trunks. Finally, it will let you create GVSIP trunks by simply entering a refresh_token and 10-digit phone number for your existing Google Voice trunk. For each trunk, the installer will create the necessary code to support a PJSIP trunk and a GVSIPn Custom Trunk to use for outbound routing. You can run the installer multiple times without worry. The second time you run it, you can install GVSIP trunk #2. The third time, it's GVSIP trunk #3. There's no limit. You can delete existing GVSIP trunks (#1 through #9) by running 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.

Setup: Once you have added at least one GVSIP trunk, you will need to go into FreePBX with a browser and add an Outbound Route for each of your trunks. Outbound calling with your new trunks will not work until you do this. We recommend dialing prefixes of *41-*49 for outgoing calls, but you can set things up however you like. That's what the GUI is for. Incoming calls to new trunks by default will go to Allison's Demo IVR if you're using Incredible PBX. You should 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. For non-Incredible PBX users, inbound calls won't be processed until you add an incoming route.

Installation: To get started, login to your Linux CLI as root. Make a backup. Be sure your MySQL root password is set to passw0rd (with a zero) before proceeding. Then...
Code:
cd /root
wget http://incrediblepbx.com/gvsip-naf-gui.tar.gz
tar zxvf gvsip-naf-gui.tar.gz
rm -f gvsip-naf-gui.tar.gz
cd gvsip-naf
./install-gvsip

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

Should you ever want to refresh the patched version of Asterisk, copy 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.
 
Last edited:

Eliad

Active Member
Joined
Aug 13, 2017
Messages
619
Reaction score
127
I installed it and seems to work well but it seems to break the Avantfax. this is the error that I am getting on the Avantfax

sendfax: Error creating cover sheet; command was "/var/www/html/avantfax/includes/faxcover.php -C '/var/www/html/avantfax/images/coverpage.html' -f 'atlasfax' -n '2132693228' -r 'Re: Cxxxx Kxxxxx' -s 'default' -t 'Hashenda Baxter' -x 'Release Point' -L 'Great Falls, MT' -N '406xxxxxxx' -V '406xxxxxxxx' -X 'Atlas Neurology' -M '[email protected]' -p '6'"; exit status ff00
 

wardmundy

Nerd Uno
Joined
Oct 12, 2007
Messages
19,168
Reaction score
5,199
I installed it and seems to work well but it seems to break the Avantfax. this is the error that I am getting on the Avantfax

sendfax: Error creating cover sheet; command was "/var/www/html/avantfax/includes/faxcover.php -C '/var/www/html/avantfax/images/coverpage.html' -f 'atlasfax' -n '2132693228' -r 'Re: Cxxxx Kxxxxx' -s 'default' -t 'Hashenda Baxter' -x 'Release Point' -L 'Great Falls, MT' -N '406xxxxxxx' -V '406xxxxxxxx' -X 'Atlas Neurology' -M '[email protected]' -p '6'"; exit status ff00

Not related. Open a new thread. No fax code is touched by this installer.
 

Members online

Forum statistics

Threads
25,778
Messages
167,504
Members
19,198
Latest member
serhii
Get 3CX - Absolutely Free!

Link up your team and customers Phone System Live Chat Video Conferencing

Hosted or Self-managed. Up to 10 users free forever. No credit card. Try risk free.

3CX
A 3CX Account with that email already exists. You will be redirected to the Customer Portal to sign in or reset your password if you've forgotten it.
Top