If you’ve been following along this past month as we rolled out a new SIP-based Google Voice implementation for Asterisk®, then you’ve gotten a first-hand look at how sausage is made. If you prefer free calling in the U.S. and Canada, then it was worth the effort. Today, we put the finishing touches on the new GVSIP-NAF implementation by introducing a version specifically tailored for those using FreePBX®. That, of course, includes the CentOS 6, Ubuntu 18, and Raspbian 8 versions of Incredible PBX®. While an Incredible PBX platform isn’t required to use today’s installer, it will make your life easier because Incredible PBX already includes the 1,000+ packages that transform Asterisk and FreePBX into a turnkey platform. If you’re rolling your own, then you’ll need a functioning Asterisk 13 platform with FreePBX that was built from source code rather than packages.

So, no, the FreePBX Distro®, Issabel®, AsteriskNOW®, Wazo®, VitalPBX®, and RASPBX® platforms are not suitable because there’s no easy way to recompile Asterisk. Now you know why we prefer to build Incredible PBX from source. Be advised that unresolved errors have been identified with Incredible PBX for CentOS 7 so steer clear of that platform for now.

October 14 UPDATE: Once again Google has "improved" things and changed the FQDN for GVSIP logins. The latest download (below) includes the fix. If you already have installed GVSIP and your trunks no longer register, here’s the fix:

sed -i 's|obihai.telephony.goog|voice.telephony.goog|' /etc/asterisk/pjsip_custom.conf
sed -i 's|obihai.telephony.goog|voice.telephony.goog|' /root/gvsip-naf/install-gvsip
amportal restart

October 1 UPDATE: There now is a GVSIP solution for the platforms mentioned above that don’t include a way to recompile Asterisk. Follow this link for the new tutorial.

July 25 NEWS UPDATE: Google now has discontinued support of their XMPP interface to Google Voice so the latest CentOS/SL, Ubuntu 18.04, and Raspberry Pi versions of Incredible PBX (13-13.7) including the Incredible PBX ISO and a VirtualBox image now incorporate NAF’s GVSIP interface to Google Voice natively. If you’re using one of those platforms, running the GVSIP installer as documented below will automatically skip the initial installation step and immediately let you add GVSIP trunks to your server. All you’ll need to add GVSIP trunks are a refresh token and phone number. Then create an outbound and inbound route for each trunk. It takes about 10 seconds. Keep reading. We’ll show you how.


After completion of today’s initial install, this release will let you deploy an unlimited number of Google Voice trunks on your Asterisk platform at a deployment rate of about 10 seconds per trunk. Once the trunks have been registered, we’ll use FreePBX to define an inbound and outbound route to manage the Google Voice calls. The entire setup process only takes a couple of minutes. We’ve also future-proofed this build with a simple way to not only add and delete trunks but also to rebuild Asterisk with new NAF releases should that ever become necessary. To get started, we recommend a clean install of Incredible PBX 13-13 on a platform of your choice. If you’re using an existing server, make sure you have uninstalled any Google Voice trunks you plan to use in today’s deployment. That includes Google Voice implementations on smartphones, Chromebooks, and other desktop machines.

Methodology. Our FreePBX integration approach blends what we believe are the best features of a command line interface with a GUI. The reason is fairly obvious. A typical GVSIP setup for Asterisk PJSIP trunking support involves more than three dozen complex settings together with more than 100 alphanumeric characters for OAuth 2 credentials. Setting that up in a GUI-style interface is obviously fraught with peril. A single typo and your Google Voice trunk won’t work. The error codes would take a genius to decipher. In addition, GVSIP configuration changes to Asterisk currently require an Asterisk restart. That requires root permissions. Assigning root permissions to any web application, especially FreePBX, would be extremely dangerous and is something we would strongly discourage.

So we’ve chosen to automate the creation of GVSIP trunks using a command-line interface available only to the root user. And you only need to enter two pieces of information, your refresh_token and your Google Voice 10-digit phone number. The installer handles everything else error-free. Typically, you set up trunks one time on a PBX and forget about it. So we think today’s approach offers the best of both worlds without exposing your GVSIP implementation to needless problems and security issues. Purists may (and probably will) have a different view, and that is perfectly fine by us. The beauty of open source platforms is you have choices. If you don’t like our approach, try something else if and when it’s available.

Special Thanks. Our extra special tip of the hat goes to @NAF of Oklahoma fame for his fantastic implementation of Google Voice on Asterisk’s new PJSIP platform. Without a shred of documentation, NAF almost single-handedly deciphered the secret sauce behind Google’s new SIP implementation with little more than a working Obi 200 device.

Initial FreePBX Platform Setup

The easiest way to begin is to install a fresh copy of Incredible PBX for CentOS 6.9, Ubuntu 18.04, or Raspbian. If you prefer a Cloud-based platform, you can’t beat our HiFormance build which installs in less than a minute and costs about $1/month. For other FreePBX implementations, you’ll need a working platform that supports recompiling Asterisk 13 from its original source code. Once you have a functioning server, you’re ready to install the GVSIP components.


Obtaining Google Voice Credentials for GVSIP

You’ll need at least one dedicated Google Voice account to use the new GVSIP implementation with Asterisk. If you’re new to all of this, our Getting Started with Google Voice tutorial will walk you through setting up an account and obtaining your OAuth 2 refresh token for GVSIP.

Installing GVSIP-NAF for FreePBX

With your Google Voice refresh token and 10-digit phone number in hand, you’re ready to begin the GVSIP-NAF upgrade. If you’re using one of the platforms with native GVSIP support, it will automatically skip the initial install step. Otherwise, make sure you have a backup of your server before you begin. Regardless of platform, don’t forget to disable and remove any Google Voice trunks you plan to use from all other platforms. Now log into your server as root and issue the following commands to begin:

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

Running the installer (install-gvsip) the first time on a platform without native GVSIP functionality will get your Asterisk/FreePBX platform up to speed by installing the correct version of OpenSSL for your platform. Then it installs and patches Asterisk 13.22.0 to support GVSIP Google Voice trunks.

For all platforms, the installer 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. To add additional trunks, simply run the installer again. Adding a new trunk takes about 10 seconds.

Should you ever want to refresh the patched version of Asterisk, copy pjsip_custom.conf from /etc/asterisk 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: amportal restart. That way you won’t lose your previously configured trunks. This is not yet available on the Raspberry Pi because of space constraints.

If you ever need to delete a GVSIP trunk that you previously have added, we’ve included a script which will perform the task for you. Just run del-trunk and specify the trunk to delete.

Configuring an Inbound Route for GVSIP Trunks

By default, incoming calls to GVSIP trunks on Incredible PBX servers will be sent to the Default Inbound Route configured on your PBX. As initially installed, that Default route points to Allison’s Demo IVR. This can be changed easily in the FreePBX GUI by modifying the Destination for the Default inbound route in Connectivity:Inbound Routes.

On other server platforms, you may not have a Default inbound route configured so you will need to create an inbound route to handle calls from each GVSIP trunk. Regardless of your server platform, we strongly recommend adding an Inbound Route for every GVSIP trunk using the 10-digit GVSIP phone number as the DID for the inbound route. Here’s an example of an Inbound Route created in Connectivity:Inbound Routes:Add Inbound Route:

If you have installed the Incredible Fax add-on, you can enable Fax Detection under the Fax tab. And, if you’d like CallerID Name lookups using CallerID Superfecta, you can enable it under the Other tab before saving your setup and reloading your dialplan.

Configuring an Outbound Route for GVSIP Trunks

By default, you cannot place outbound calls using your new GVSIP trunks. For each trunk, you first will need to create an Outbound Route specifying a Dial Pattern to use with each GVSIP trunk in Connectivity:Outbound Routes:Add Outbound Route. If you only have a single Google Voice trunk on your PBX and no other trunks, then you would probably want to specify that outbound calls be routed out the GVSIP1 trunk with a Dial Pattern of NXXNXXXXXX with 1 as the Prepend. This tells FreePBX to dial 18005551212 using the GVSIP1 trunk when a PBX user dials 8005551212. Google only accepts calls that include a country code (1=US/CAN).

HINT: If you ever forget which GVSIP trunks are associated with which phone numbers, simply run /root/gvsip-naf/del-trunk for a list of your trunks. Just press ENTER to exit without deleting any of your trunks.

There are a million ways to design outbound calling schemes on PBXs with multiple trunks. One of the simplest ways is to use no dial prefix for the primary trunk and then use dialing prefixes for the remaining trunks. As part of the install, the dialing prefixes of *41 through *49 were reserved for GVSIP trunks if you would like to use them. That’s totally up to you. Here’s what an Outbound Route would look like using this scheme for the GVSIP2 trunk:

Another outbound calling scheme would be to assign specific DIDs to individual extensions on your PBX. Here you could use NXXNXXXXXX with the 1 Prepend as the Dial Pattern with every Outbound Route and change the Extension Number in the CallerID field of the Dial Pattern. With this setup, you’d need a separate Outbound Route for each group of extensions using a specific trunk on your PBX. Additional dial patterns can be added for each extension designated for a particular trunk. A lower priority Outbound Route then could be added without a CallerID entry to cover extensions that weren’t restricted or specified.

HINT: Keep in mind that Outbound Routes are processed by FreePBX in top-down order. The first route with a matching dial pattern is the trunk that is selected to place the outbound call. No other outbound routes are ever used even if the call fails or the trunk is unavailable. To avoid failed calls, consider adding additional trunks to the Trunk Sequence in every outbound route. In summary, if you have multiple routes with the exact same dial pattern, then the match nearest to the top of the Outbound Route list wins. You can rearrange the order of the outbound routes by dragging them into any sequence desired.

Where Do We Go From Here?

We’ve now integrated GVSIP directly into all of the Incredible PBX 13-13 builds including the Incredible PBX ISO. This eliminates the need to run the GVSIP installer other than the 10-second procedure to add Google Voice trunks to your PBX. And we now have documented how to use a cloud-based Asterisk server as a Google Voice platform and host for many of the distributions which do not yet support GVSIP. Enjoy!

Dale Carnegie Award: Polycom’s PR Man of the Year

Originally published: Saturday, July 14, 2018  Updated: Sunday, October 14, 2018

Support Issues. With any application as sophisticated as this one, you’re bound to have questions. Blog comments are a difficult place to address support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forum. It’s the best Asterisk tech support site in the business, and it’s all free! Please have a look and post your support questions there. Unlike some forums, the PIAF Forum is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of users just like you. You won’t have to wait long for an answer to your question.

Need help with Asterisk? Join our new MeWe Support Site.


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This article has 32 comments

  1. Hi WM — do these instructions work for those who are running Incredible PBX 11.12 on the Pogoplug V4? Or are there some special compilation that we are required to do? Thanks.

    [WM: I’d suggest reviewing the install script and sorting out OpenSSL on the Pogoplug first. Once you get it up to version 1.1 (openssl version), it’s probably safe to run the installer which then will skip the OpenSSL step. Make a backup first.]

  2. Wow! I mean WOW! Keeping up with this recent drama has been more exciting than that time a TV show called The Sopranos came to a conclusion. But nevermind all that, what with the truly stellar dedicated and focused efforts of a few highly skilled folks that made such an effort and achieved such an esteemed level of progress for the international open-source VOiP community as a whole. Wow!

  3. Will this work with Incredible PBX also? I added a GVSip trunk and outgoing calls are being disconnected.

    [WM: You have to go through the upgrade procedure unless you installed the beta of Incredible PBX for GVSIP that was released yesterday. Head to the PIAF Forum if you need further help.]

  4. > NAF almost single-handedly deciphered the secret sauce behind Google’s new SIP implementation with little more than a working Obi 200 device.

    The dozen folks, including a Digium programmer, guiding him along on the dslreports forum, might say otherwise. Not to say he hasn’t done the heavy lifting, but let’s call it a community effort.

    [WM: Great clarification. No question that others including Digium and Simonics contributed valuable insights in moving things along. But 99% of the code and the reengineering of Asterisk’s PJSIP code to support the new Google Voice was all NAF’s handiwork.]

  5. So this is some amazing work as always! Quick question, if I already have few trunks in the pjsip_custom.conf how do I get them to show up in the GUI? Do I have to delete and add them again? Would figure since script can read existing trucks it would add them as well, but I only see the new additional one. AMAZING WORK !

    [WM: GUI representation of gvsip trunks is the missing piece at the moment.]

  6. maybe my last comment was misunderstood, so I have 3 trunks pre running the new GUI script. I ran the script, it asked me to add 4th trunk, I did, then I can use that NEW trunk #4 in FreePBX GUI as a trunk selection, but the other 3 older trunks are not in the list for me to select as outbound route…

    I figured since script parses the trunks it would add the existing trunks to FreePBX as well not only the new one.

    [WM: Take a look at the custom trunk added in FreePBX and then clone it for GVSIP1-GVSIP3.]

  7. Brian (tycho on PIAF forums; brg on DSLR)

    Is your reference to "Ubuntu 14″ in your first paragraph and error/typo that should instead be "Ubuntu 18″?

    [WM: It was an error. Fixed. Thanks for catching it. We used Ubuntu 14.04 for so long that we almost forgot they had newer releases.]

  8. Does this mess up a previous install-gvsip?

    [WM: Probably best to start over or at least remove your pjsip_custom.conf file before loading the update.]

  9. After 3 days of working on this. I finally have the following working
    1. Incredible PBX on a Ubuntu 1804 on a Synology NAS Virtual Manager – PBX
    2. SPA 3102 as a POTS or PSTN Gateway with analog extension (both registered using chan_sip & could not get PJSIP to work with this) – For PSTN calls
    3. naf – Googlevoice script (PJSIP Trunks) for Google Trunks (2 GVSIP added) – For USA/Canada & International calls. Needed to add STUN address for reliability of google (does not come with NAF script)
    4. Extensions on PJSIP (IP Phone & couple of SIP clients MAC/Android/IOS
    What is working?
    1. Inbound and outbound calls on Google voice – Works
    2. Inbound and outbound calls from PSTN via SPA3102 – Works
    3. CallerID works on both
    4. NEXT – CAllerID with Superfecta

    Thanks a ton for all the blogs and detailed instruction. I am compiling instructions and screenshots to post to help folks out.

  10. I’ve just finished installing an Ubuntu 18 based GVSIP PBX with 3 GV lines.
    Everything seems to work well. The voice quality is quite good although I have the impression that it is one notch below Motif’s quality (using only ulaw).

    I have noticed on the dashboard that the number of trunks varies between 5 and 7 and it changes quite often. Is that a malfunction of my system or normal behavior?

    Also, I sometimes get a message saying that the system can’t dial out because it is not available, or busy, etc. The message is not always the same, it varies and I didn’t get to do a full list of the different errors. But, I can’t correlate the error message with the number of trunks shown on the dashboard. I think they are unrelated. The problem goes away by itself after a little while, but if I want to recover I restart asterisk with "fwconsole restart". It seems to do the trick.

    Any ideas about how I should troubleshoot my system further?


  11. I’ve confirmed that the http://incrediblepbx.com/gvsip-naf-raspi.tar.gz
    has a corrupt asterisk-13.21.1.tar file in it

    http://incrediblepbx.com/gvsip-naf-gui.tar.gz file’s asterisk-13.22.0.tar in not corrupt but it’s a different version.

    It looks like the gvsip-naf-raspi.tar.gz file will not work for installing gvsip-naf-raspi.tar.gz.

    Where is the best place for this discussion to take place?

    [WM: We have removed the gvsip-naf-raspi installer. We will be unable to get to a RasPi device for testing for the next several days. In the meantime, let me recommend that you try the new release of the latest installer for all platforms which was just uploaded at 10 a.m. on July 20. We have not yet tested it, but we think it addresses the problems encountered with the original RasPi build. Here’s the link to the updated installer and article. You can post feedback in this thread on the PIAF Forum.]

  12. Ok, so I have this running on a RPI (older model), and it works well with a few hiccups. One, I noticed as I was doing unrelated work on the main network router, that when I outbound IP address changed, all the GV trunks didn’t work. A quick reboot of the RPI resolved that, but something to keep in mind. Second, this one seems like a bug, but the stats graph on the Dashboard shows all my GV trunks offline. I know this is not the case. Other wise, nice work!!!

    [WM: Thanks, Chris. You’ll need to reserve the DHCP address in your router for dedicated use with your RasPi. Otherwise, you are correct. It will kill off your GVSIP trunks every time the IP address changes. And, yes, the GUI display of the trunks being off-line is a FreePBX bug. There is a fix, but it causes an even worse bug that fills up your logs with incorrect error messages. So… someday (maybe) it will get fixed.]

  13. snovotill@gmail.com

    I have to use Ubuntu because my iNtel Cherry Trail box requires the latest 4.15 kernel (actual power dissipation only 2.49 watts in standby!). Both of my GV trunks are working FANTASTICALLY WELL. Thank you Ward =)

  14. I have the GV-SIP trunk configured but ran into a problem with google. The option to forward calls to Chat is now missing. anyone have any ideas on how to get calls to forward to chat to get my FreePBX working?

    [WM: Gone forever, I’m afraid.]

  15. I am able to install IncrediblePBX and I’m really impressed with the features (I started with Elastix and moved over to FreePBX before discovering this gem), the only problem I am running into is that I did a test installation on a virtual machine before trying to throwing it on to an actual physical system and now nothing other than the original test system will register the trunks, they just sit there as PJSIP unregistered.

    Is there something special I need to do to release them from the original server. Would this be related to port forwarding? Would a static IP address resolve this issue. I should mention the original virtual machine (the test machine), was behind a NAT’d network and didn’t require any port forwarding, so I’m not sure if that would even be an issue.

    [WM: Try running del-trunk to remove all the original trunks from your test machine.]

  16. I set everything up but using PAP2T no outgoing ringtone is played, call will connect and can pass audio without issue. Incoming calls work without issue.

    Tried on local network and outside with NAT still same issue no ringback. Anyone else having this issue or know a fix? Works fine with Cisco 7965.

    [WM: Lack of ringing on calls is a known issue with the new Google Voice.]

  17. Man you guys totally rock. I wired my house with Cisco 79xx phones and been able to call out via chan_motif/gv for the past 4 years without any issues until late last month.

    I got the GVSIP and PJSIP drivers working and kudos to you all for the reverse engineering, packet sniffing, and TLS proxy intercepts that must have been done to get this to work!!!

    [WM: Most of the design work was purely @NAF with some navigational tips from a few others along the way. My limited contribution was coming up with a simplified implementation so that mere mortals (like me) could use it.]

  18. AWS Cent OS 6.10 2 GV trunks works spot on and 8 years following WMs site…….never a dull moment…..awesome work sir WM. Thanks.

  19. This post says FreePBX but all links point to IncrediblePBX. I do not believe they are the same. Will the same file install in FreePBX?

    [WM: That depends upon the version of Asterisk, the version of FreePBX, and whether your platform was built from packages or source code. If your server includes Asterisk 13 source code, you’re probably well positioned to try the upgrade procedure assuming you also have a relatively recent version of FreePBX. Here’s a chart with all the available GVSIP options.]

  20. Looking to spin up a hosted solution with HiFormance. Anyone know if the July 25th image is going to be the one used if I sign up today?

    [WM: GVSIP is not included in the HiFormance image of Incredible PBX 13-13. You’d need to run through the upgrade procedure using the install-gvsip script. It takes about 20 minutes after you bring your server on line.

  21. So is GV as a Trunk for PBX no longer an option?

    [WM: Google Voice still can be used as a trunk with Incredible PBX, just not as an XMPP/Motif trunk. See this tutorial for how to upgrade existing servers. Or you can install one of the new Incredible PBX servers that has native GVSIP support rolled into it.

  22. Will this work with newer accounts that are not able to forward calls to google chat?

    [WM: It will work with all Google Voice accounts. In fact, no accounts can forward calls to Google Chat any longer.]

  23. Got everything installed, and you are the man. Only issue is that when I call my GV number, the PBX picks up but directs all calls to a message stating the number I have dialed can not be completed, check the number and try again.

    [WM: Sounds like you forgot to configure an Inbound Route in FreePBX for your GVSIP trunk. Review this tutorial and be sure you didn’t skip a step.]

  24. Thanks for responding Ward. You’re awesome! I’m the one who asked about being able to use this implementation even if I can’t forward calls to google chat. It seemed as if it would work with all accounts which is good and bad news. I’m glad it will work, but that means I’m doing something wrong. I’ve followed the tutorials to the best of my ability and have a working PBX up and running, but my google voice trunk is still showing as disconnected, despite my best efforts to fix it. I’m not stupid, just inexperienced. I could certainly explain the problem in more detail, but I get the feeling this isn’t the right place. Where should I go for help?

    [WM: The PIAF Forum is a great resource if you’re having trouble. Before posting there, I’d recommend you start over and the follow this tutorial to set up a GVSIP trunk. If it still doesn’t work, take a look at the log file in /root and see where the error(s) occurred. Then post away. Good luck!]

  25. I’m missing something. Fresh CentOS 6.9 install / fully updated
    Does the installer "/install-gvsip" install Asterisk? It seems to be missing.

    [WM: install-gvsip installs Asterisk on an Incredible PBX (or other) platform that already is running Asterisk 13 and FreePBX from source code builds, not packages.]

  26. Will the upgrade work on Ubuntu 14 also?

    [WM: Ubuntu 14 has reached EOL, and we haven’t tested with it. I suspect there may be a problem with OpenSSL. If you can get that upgraded to 1.1 before you begin, then the installer should work just fine.]

  27. So, I had this working before the post was published (that NEVER happens).

    Incoming calls have a +1 appended. There’s no context options for the custom trunk.

    Anyone know how to have calls that start with a +1 successfully go through? That’s the last thing I need for it to be perfect!

    [WM: Easiest way is to change the context for your GVSIP trunks. Instead of from-trunk, use from-pstn-e164.us]

  28. So I’m trying to install this IncrediblePBX from source… and I after running the install script and allowing it to compile asterisk, I can no longer start Asterisk and am getting this error:

    root@srv-pbx1:/usr/src/asterisk-13.22.0 $ asterisk
    FATAL: ThreadSanitizer can not mmap the shadow memory (something is mapped at 0x5611fe083000 < 0x7cf000000000)
    FATAL: Make sure to compile with -fPIE and to link with -pie.

    [WM: Don’t use CentOS 7.]

  29. Not having any luck getting this to work on my CentOS machine =(

    After compiling Asterisk, it won’t start, I keep getting:

    root@srv-pbx1:~ $ asterisk -vr
    FATAL: ThreadSanitizer can not mmap the shadow memory (something is mapped at 0x5620fcd1a000 < 0x7cf000000000)
    FATAL: Make sure to compile with -fPIE and to link with -pie.

    I even tried to remove LibTsan and it REALLY didn't like that =(

    [WM: Don’t use CentOS 7.]

  30. In my previous PBX, I had an extension that would use a Misc Destination to call my cell back through the same GV trunk a caller is calling in with. Is this no longer possible? I even registered a new GV account and used it as the outgoing trunk, but I am still getting a busy signal every time.

    [WM: Should still work fine. Open a thread on the PIAF Forum, and we’ll figure it out.]

  31. I have installed PBX 13-13 for VMware. I have 3 trunks + google voice. Google Voice works for 1 day and next day i have to reboot the server and it works again. What can be causing this that it disconnects after a period of time?

    [WM: We’ve seen similar issues with VMware ESXi. The absence of errors in the Asterisk logs suggests this has more to do with VMware failures than Google Voice or Asterisk problems. Only suggestion I can offer is to change platforms if you experience unexplained outages.]

  32. You script assumes that patch is available by default on a centos machine. It is not, you should yum install it.

    [WM: All necessary patches are included in our Incredible PBX 13-13 installer for CentOS.]

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