ALERT GV: The Sky Has Fallen... Really

Discussion in 'Today's Tech News & Events' started by wardmundy, May 10, 2018.

  1. billsimon

    billsimon Experienced in Asterisk, FreePBX, and SIP

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    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.
     
  2. mcfuzz

    mcfuzz New Member

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    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 :(
     
  3. billsimon

    billsimon Experienced in Asterisk, FreePBX, and SIP

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    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.
     
  4. mcfuzz

    mcfuzz New Member

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    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...
     
  5. jhankins

    jhankins New Member

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    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:gv1@obihai.sip.google.com
    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.
     
    Jebs2k likes this.
  6. yozh

    yozh Member

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    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 :(
     
  7. jtpiano

    jtpiano New Member

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    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?
     
    #47 jtpiano, Jul 2, 2018
    Last edited: Jul 2, 2018
  8. Jebs2k

    Jebs2k New Member

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

    kdthomas Member

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    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?
     
  10. 2devnull

    2devnull New Member

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    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?
     
  11. wardmundy

    wardmundy Nerd Uno

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    Yes, unless you want to move to an Incredible PBX platform that supports the new GVSIP implementation. See Nerd Vittles for details.
     
  12. 2devnull

    2devnull New Member

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    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.
     
  13. 2devnull

    2devnull New Member

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    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.
     
  14. TopMark

    TopMark New Member

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    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.
     
  15. 2devnull

    2devnull New Member

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    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'​
     
    #55 2devnull, Jul 11, 2018
    Last edited: Jul 11, 2018
  16. Eliad

    Eliad Member

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    nerdvittles.com
     
    #56 Eliad, Jul 11, 2018
    Last edited by a moderator: Jul 12, 2018
  17. 2devnull

    2devnull New Member

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    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'​
     
    #57 2devnull, Jul 11, 2018
    Last edited: Jul 11, 2018
  18. wardmundy

    wardmundy Nerd Uno

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    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.
     
    #58 wardmundy, Jul 12, 2018
    Last edited: Jul 20, 2018
    restamp and Waffull like this.
  19. Eliad

    Eliad Member

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    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 'atlxxxx@atlasneurology.com' -p '6'"; exit status ff00
     
  20. wardmundy

    wardmundy Nerd Uno

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

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