How to use Google Voice for free calls on an Asterisk 1.8+/FreePBX 2.8 system (the easy way)

December 21, 2010 at 9:35 pm · Filed under Asterisk, Google Voice ·Tagged Asterisk, Google, Google Talk, Google Voice, sudo, Voice over Internet Protocol

EDIT: This is one of the most popular articles on this site, so I really hate to have to say it but this method doesn’t work very well anymore, and that’s because Asterisk’s support of Google Voice, in a word, SUCKS (that is just my opinion, of course, but feel free to try it if you don’t believe me). Google Voice changed some things, and it sure appears as if Digium hasn’t even tried to keep up with the changes. Obihai devices, FreeSWITCH, and YATE all seem to have fewer or no issues communicating with Google Voice, while in Asterisk it’s a crap shoot whether you will receive incoming Google Voce calls (for the moment, outgoing calls still seem to work, though there is a rather long delay before the call connects through).

So I am going to make an unusual recommendation: Don’t use the method shown here, or ANY method you find that relies solely on Asterisk to connect to Google Voice. If you have an Obi100 or OBi110 adapter, consider using it as a gateway between Asterisk and Google Voice. If you don’t have an Obihai adapter, or need to bridge multiple Google Voice accounts, then take a look at the article Using FreeSWITCH to add Google Voice to Asterisk, or check the comment section of this article for some additional suggestions.  At the end of January, 2012 I used the technique in the aforementioned link to use FreeSWITCH as a gateway between Google Voice and Asterisk, and I must say that it works a LOT better than Asterisk’s Google Voice support.  I also added some comments to that article (probably too many!) that among other things show how I got it working for multiple Google Voice accounts.  So if you don’t have an Obihai, I would now recommend using that method, unless you are very short on memory and/or storage space — it just works, and calls connect faster.  If you do go that route, read the article AND the comments under it first, so you’ll know what to expect, and do be aware that it takes a relatively LONG time to compile and install FreeSWITCH (compared to Asterisk).  At points during the installation it may look like it’s stuck in an endless loop, but it really isn’t. Just go away and take a walk outside or something, and come back in a while and it should be done.

And if you just don’t want to be bothered with any of that, and you don’t mind using an external service, you could try using Bill Simon’s Google Voice-SIP gateway (the linked article shows how to use it with an Obihai device, but if you read through it you will see it also shows that the gateway can be used with Asterisk directly).

But just in case you think you can debug Asterisk’s buggy channel drivers and get this working as it should, here’s my original article. Unless you are a master coder, however, I strongly suggest that, at least for the time being, you do NOT follow these instructions, because you are most likely NOT going to be happy with the result!

* * * * *

This is a followup to my earlier article, How to use Google Voice for free outgoing calls on an Asterisk/FreePBX system (the easy way). Starting with Asterisk 1.8 it’s a whole lot easier, and doesn’t require you to have a separate DID or Gizmo5 account to bring in the calls. My goal was to simplify (as much as possible) the dial plan code used, especially the code that has to be entered into /etc/asterisk/extensions_custom.conf, and to speed up execution by getting rid of unnecessary waits (why make callers wait extra seconds if you don’t have to? That’s just inconsiderate).

This article comes with my standard disclaimer: Although it worked for me, it may or may not work for you, and I offer no warranties or guarantees of any kind. I present the following for experimental use only. Don’t put it on a production server unless you, personally, are satisfied that it works the way you need it to. If it stops working, you have my sympathy but that’s about it. If it breaks your system, you own all the pieces. I am not your technical support, so don’t complain to me if it doesn’t work, but if you leave a detailed comment maybe I or someone else can point you in the right direction. Just saying that something doesn’t work is not a detailed comment, and if you leave a comment like that, don’t be surprised if all you hear is crickets chirping. Still here? Let’s get started then.

Note that you should be running as root while you are configuring and/or installing Asterisk and any related programs, or else be using sudo in front of all commands entered at the Linux command prompt during the configuration and installation process.  Otherwise files may be installed with the wrong permissions and/or ownership, and things won’t work!

Until Google fixes the bug where your preference to disable Call Screening (where you are prompted to press “1″ to accept the call, etc.) is NOT honored when you use a Google Chat destination in Google Voice, the first thing you have to decide is how you want to answer incoming calls.  Will you want to answer a call as soon as it comes into your system, or defer answering until a destination extension picks up? There are pros and cons to each approach and rather than rehash those here, I’ll refer you to my previous article, Asterisk 1.8.x and FreePBX users: How to NOT answer Google Voice calls UNTIL the called extension answers. You may want to read the discussion there, and then come back here.

You also have to determine what version of Asterisk you are running. You need to be running at least Asterisk 1.8.1.1, or something in the Asterisk 1.8.2 branch or higher, for outgoing calls to work. From the Asterisk CLI you can enter core show version to see what version of Asterisk you are using.  If you don’t have the correct version you will need to upgrade Asterisk.  If you’re running an earlier version of Asterisk, see Links and Information: Updating an Asterisk/FreePBX system so you can use Google Voice for possible assistance.

Also, you need to make sure the proper modules are installed.  Go to your Asterisk source directory (usually it’s /usr/src/asterisk-1.8.something but be sure you’re at the correct version) and run make menuselect.  Look under Channel Drivers and make sure there’s an asterisk [*] next to both chan_gtalk and chan_jingle:

Also look under Resource Modules check to be sure there’s an asterisk next to res_jabber:

If everything looks good, just hit escape to exit (and skip the rest of this paragraph).  But, if you see XXX next to one or both, it probably means that iksemel is not installed on your system. The current version of iksemel (as of this writing) is 1.4 and you can get it here.  After you install and compile it (see How to fix the problem of missing modules res_jabber.so and/or chan_gtalk.so in your Asterisk installation if you need help with that), from within your Asterisk source directory, you’ll have to do contrib/scripts/get_mp3_source.sh (only if you’ve not previously done that), then ./configure, then make menselect (select chan_gtalk, chan_jingle and res_jabber if they aren’t already selected, and also app_mysql, app_saycountpl, cdr_mysql, format_mp3, and res_config_mysql if you just ran the get_mp3_source.sh script, then Save & Exit), then make, and finally make install to recompile Asterisk (if you’re going to do this anyway, this may be a good time to upgrade Asterisk to the latest version if you don’t already have it).

EDIT: Note that in many (possibly most) installations, you may have to go into your firewall and route UDP port 5222 to your Asterisk server, in the same way you probably had to map UDP port 5060 to your server in order to make SIP calls work. You can do this here at the start of the process as a precaution, or you can wait to see if you have problem making this work and if so, do it then. It’s up to you.

One final thing before we proceed — let’s talk a moment about Google Voice accounts. When you create a Google Voice account, it is a very good idea to first create a new Gmail account that you won’t actually use for e-mail, then tie the Google Voice account to that. Don’t tie your Google Voice account to your primary e-mail address because the account password has to be stored in plain text in the jabber.conf configuration file (so if someone hacks your Asterisk box, they could get into your e-mail account) AND because whenever you are logged into Gmail, there is some chance you might not receive incoming calls. So unless you are very attached to a particular Google Voice number (or managed to snag one in a Frontier exchange while they were still up for grabs), get a new Gmail account, use a good strong password when you create it, and then open your Google Voice account from there. And when you set up your Google Voice account, be sure to set it to forward to Google Chat, and be sure that Call Screening is off, that Caller ID (incoming) is set to Display caller’s number, that Do Not Disturb is NOT enabled (at least not whenever you actually want to receive calls!), and that Call Options are NOT enabled.

EDIT: At this point I should probably let you know that since this article was originally written, there is now an unofficial, third-party Google Voice FreePBX module by Marcus Brown (look at the “googlevoice” entries and pick the one with the highest version number).  Always use the latest version — older versions CAN break things on your system!  Those who have used it say it works well and makes configuration easier, and probably would eliminate the need to follow most or all of the instructions below.  The catch is that the original version only worked with FreePBX 2.8, due to internal changes made starting in FreePBX 2.9.  Patches have been provided to make it work with FreePBX 2.9, but by certain FreePBX developers, not by the original author.  So, while this module could make Google Voice configuration a whole lot easier, you do run the risk that it might no longer work if you upgrade FreePBX to a new major version at some point in the future.  Anyway, the following instructions are for those NOT using this module.

Once you have the prerequisites you can proceed to create the files you need.  You’re going to need to make or modify four files in the /etc/asterisk directory.  The first is gtalk.conf.  If you have nano installed, you can do this:

touch /etc/asterisk/gtalk.conf
nano /etc/asterisk/gtalk.conf

Or, use any editor you prefer (the same applies to all the files we’ll deal with here). Then paste the following text into that file:

[general]
allowguest=yes
context=googlein
stunaddr=stun01.sipphone.com

[guest]
disallow=all
allow=ulaw
connection=asterisk
context=googlein

Next, create the file jingle.conf (touch /etc/asterisk/jingle.conf) but don’t put anything in it. If it already exists, use an editor to make sure every line inside is commented out (with a semicolon at the start of the line) or remove it.

Next we must create jabber.conf (touch /etc/asterisk/jabber.conf). Into this file we will put our user details.

EDIT: I’m interrupting this article here to say that once you’ve got this far in the process, you have the option to beta test a BASH shell script to help you with the rest of the configuration.  If you’re not scratching your head wondering “what’s a shell script?” then I’d suggest you at least take a look at it, because using that script should make this process easier (particularly if you intend to have more than one Google Voice account coming into your system).  If you don’t wish to use the script then you can continue on with configuration of jabber.conf.

In the sample configuration below you will see username, where username is the part of the Gmail address before the @ on the associated Gmail account (don’t neglect to change the context name [username], in addition to replacing username in the line username=username@gmail.com/Talk).  For each additional Google Voice user you can add another username context.  So, here’s what jabber.conf must contain:

[general]
debug=no
autoprune=no
autoregister=yes
[username]
type=client
serverhost=talk.google.com
username=username@gmail.com/Talk
secret=account_password
priority=1
port=5222
usetls=yes
usesasl=yes
status=Available
statusmessage="I am an Asterisk Server"
timeout=100
keepalive=yes

The settings in the general context are just preferences that can be set differently if you prefer, but they’re not what we’re really concerned about. The main thing is the username context, and again, you should replace username with the part of the Gmail address before the @ sign. Duplicate the username context as often as you need to for each Google Voice user you have, and be sure you change the context name, username, and secret for each additional context.

Finally there are the additions to /etc/asterisk/extensions_custom.conf.  We will need to add at least two contexts to that file.  The first handles inbound calls from Google Voice.  You only need ONE of these no matter how many Google Voice users you have, but you have a choice between two contexts.  Remember that at the start of this article I said you had to decide whether you want to answer the call as soon as it comes into your system, or defer answering until a destination extension picks up.  Well, if you decided to let Asterisk answer the call as soon at it arrives, then you’ll need to add this context:

[googlein]
exten => _[0-9a-z].,1,Noop(Incoming Google Voice call for ${EXTEN})
exten => _[0-9a-z].,n,Set(CALLERID(name)=${CUT(CALLERID(name),@,1)})
exten => _[0-9a-z].,n,GotoIf($["${CALLERID(name):0:2}" != "+1"]?notrim)
exten => _[0-9a-z].,n,Set(CALLERID(name)=${CALLERID(name):2})
exten => _[0-9a-z].,n(notrim),Set(CALLERID(number)=${CALLERID(name)})
exten => _[0-9a-z].,n,Wait(1)
exten => _[0-9a-z].,n,Answer
exten => _[0-9a-z].,n,Wait(1)
exten => _[0-9a-z].,n,SendDTMF(1)
exten => _[0-9a-z].,n,Goto(from-trunk,gv-incoming-${CUT(EXTEN,@,1)},1)
exten => h,1,Macro(hangupcall,)

BUT if you want to defer answering until a destination extension picks up, then you’ll want to use this context instead:

[googlein]
exten => _[0-9a-z].,1,Noop(Incoming Google Voice call for ${EXTEN})
exten => _[0-9a-z].,n,Set(CALLERID(name)=${CUT(CALLERID(name),@,1)})
exten => _[0-9a-z].,n,GotoIf($["${CALLERID(name):0:2}" != "+1"]?notrim)
exten => _[0-9a-z].,n,Set(CALLERID(name)=${CALLERID(name):2})
exten => _[0-9a-z].,n(notrim),Set(CALLERID(number)=${CALLERID(name)})
exten => _[0-9a-z].,n,Set(DIAL_OPTIONS=${DIAL_OPTIONS}aD(:1))
exten => _[0-9a-z].,n,Goto(from-trunk,gv-incoming-${CUT(EXTEN,@,1)},1)
exten => h,1,Macro(hangupcall,)

DON’T USE BOTH OF THOSE! Pick one or the other. That will take care of your incoming calls (except for your FreePBX configuration, which we’ll talk about in a second). EDIT: Note that if you use the beta BASH shell script I mentioned above, it will generate a separate inbound context for each of your Google Voice accounts, which means that you can defer answering until the extension picks up on some of your accounts, but answer the call as soon as it arrives on your system with other accounts. The contexts above don’t give you that flexibility — you have to pick one method or the other for all your accounts (unless you know what you’re doing and can make the necessary tweaks yourself). Also note that there was an error in the above contexts prior to February 5, 2011 – where you see _[0-9a-z]. I was using _[@Xa-z]. which is wrong (and which I had seen in a post on a forum somewhere), but which would still work with any Gmail address that doesn’t begin with a number. The _[0-9a-z]. will work with any Gmail address that begins with a lowercase letter or a digit, and I’ve never seen a Gmail address with uppercase characters. Thanks to voipRookie for pointing this out in the comments.

Now, for OUTGOING calls, you’ll unfortunately need to add a separate context added to /etc/asterisk/extensions_custom.conf for each user (it’s just easier that way). Fortunately they are short — each context should look like this:

[gvoice-username]
exten => _X.,1,Dial(Gtalk/username/+${EXTEN}@voice.google.com)
exten => _X.,n,Noop(GVoice Call to ${EXTEN} failed)
exten => h,1,Macro(hangupcall,)

Once again you need to replace both instances of username (including in the context name [gvoice-username]) with the part of the Gmail address before the @ sign. Don’t leave out the “Noop” line — it’s important that there be at least one line (not including the hangup context) after the Dial line, otherwise unpredictable things might happen. Don’t forget to save your changes to extensions_custom.conf, and be happy — the hard part is over!

Now for the FreePBX configuration. Inbound calls will come in on inbound routes, the same as calls from DID’s or VoIP providers. The only difference is that instead of specifying a DID number, you’ll use gv-incoming-username as the DID, with the usual substitution for username. FreePBX will give you a hard time about it when you try to save it:

Just click OK and you should be fine.  If you’d prefer to use the Google Voice phone number(s) for your DID(s), I’ll tell you how you can do that at the end of this article, but I suggest you get at least one Google Voice number working the way I’ve shown here so you know that it works (it will be easy to change the DID later).

For outgoing calls, you’ll need to create a CUSTOM Trunk for each Google Voice user.  Click on Trunks, then Add Trunk, then Add Custom Trunk.  For the Trunk Name, use anything you like that makes sense to you, but I suggest including the username in the trunk name so you can keep the trunks straight if you have more than one account.  For the Dialed Number Manipulation Rules, put a 1 before the + (first field), and NXXNXXXXXX in the last field (leave the middle one blank).  For the Custom Dial String, use local/$OUTNUM$@gvoice-username (substituting for username as we have done previously). Remember that the gvoice-username part must match an outgoing context you previously created in /etc/asterisk/extensions_custom.conf (it looked like [gvoice-username] in the above instructions).

After you’ve created your trunk(s), just use them as you would any trunk. If you only want a particular user to have access to a particular Google Voice trunk, then set up an Outbound Route specifically for that user and make that trunk the first (or only) trunk selection. Keep in mind that you can restrict an Outbound Route to a particular extension (or group of extensions) by utilizing the fourth field of the “Dial Patterns that will use this Route” section — see A couple tidbits for Asterisk and FreePBX users for more information. Any such “restricted” Outbound Routes must be above (higher in priority than) any other routes containing the same or similar dial patterns but no extension restrictions.

One final thing you will need to do in order to make this work is to go to the Asterisk CLI and issue one of the following commands:  core restart now or, if you think there might be calls in progress,  core restart when convenient — the latter will wait until there are no calls in progress.  This is necessary to load the Gtalk configuration into Asterisk — for some reason an “orange bar reload” in FreePBX doesn’t do it.

Hopefully, if you do all that, it will work for you. If it doesn’t, check to see if you’ve missed any steps. The only thing I ask is that you don’t try to mix up my instructions with something you found on some other site, then come asking for help if it doesn’t work. It probably won’t work if you do it that way.

ADDENDUM: I suppose that some people would prefer to use the Google Voice number as the DID, rather than gv-incoming-username, if only to stop FreePBX from complaining every time you make a change to an inbound route.  If that’s your desire, it’s easy to accomplish.  Just change one line in your [googlein] context in extensions_custom.conf, as follows:

Change this line:

exten => _[0-9a-z].,n,Goto(from-trunk,gv-incoming-${CUT(EXTEN,@,1)},1)

to this:

exten => _[0-9a-z].,n,Goto(from-google-voice,${CUT(EXTEN,@,1)},1)

Next, underneath the end of the [googlein] context, create a new context called [from-google-voice] and put one line it it for each Google Voice account you have, followed by a hangup context as shown below:

[from-google-voice]
exten => username1,1,Goto(from-trunk,gvnumber1,1)
exten => username2,1,Goto(from-trunk,gvnumber2,1)
exten => username3,1,Goto(from-trunk,gvnumber3,1)
exten => h,1,Macro(hangupcall,)

Replace username1, username2, username3, etc. with the part of the Gmail address before the @ sign.  Replace gvnumber1, gvnumber2, gvnumber3, etc. with the ten digit Google Voice number associated with the account.  Of course, you would use as many lines as you have accounts (which, unless you’re doing something unusual, would be the same as the number of user accounts you have enabled in jabber.conf).  Save your changes, go to your inbound routes and change the DID(s) for each Google Voice account to the associated ten digit Google Voice number, and you’re all set — no more complaints from FreePBX!

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