Asterisk 11 + Google Voice

*** Update ***

Unfortunately, Google has decided to end support for the XMPP protocol in May of 2014. I will keep the post here in the hope that Google releases a new API, but also to share that information with any searchers. As of April 2016, the service is still operational, but there’s no telling for how long that will be the case.

Introduction: History

Earlier this year, the company I work for, Laird Technologies, converted their phone system to Cisco IP phones. In the process they threw out all of their old IP phones which were used with their Asterisk system. I acquired these, all Grandstream products (mostly GXP2020’s, 1 or 2 GXP2000’s and even a HandyTone-286!) and I had some ideas about what I could do with these, but I mostly just wanted to play around: set up phones around the house, make a doorbell/intercom system, etc. After I started reading and Asterisk and how people are using it, I got the idea to set up my own “land-line” service using Google Voice.

I think the first thing to understand about Asterisk is that it is a hugely powerful tool; you can run a call-center or office phone system on it. The second thing to understand is that it is Free and Open Source Software (FOSS)! These two factors combined mean that you can do almost anything Voice over IP (VoIP) related, but you end up reading a lot of blogs and documentation in order to really make it happen.

In my reading, I found that there are basically two different ways to accomplish my goal:

1. Set up Asterisk in conjunction with another Private Branch Exchange (PBX) and let the other PBX handle the GV transactions.
2. Take advantage of a new Google Voice engine in an upcoming version of Asterisk: Asterisk 11.

When presented with these two options, I really wanted to make a pure Asterisk system to reduce the complexity (Asterisk is complex enough as-is) and to really get to know Asterisk.

It took me quite a few tries to get this right. Setting up a PBX with calling inside of our home network was a piece of cake given how simple that configuration is. After I set up the Motif channel for Google Voice, I could see that it was connected (through Gmail Chat) and I could even make the phone ring by calling my Google Voice Number, but when I answered, I either got no audio or the call wouldn’t connect (this was because of how Motif answers a call, using a DTMF tone) so I figured it had to be a problem with my networking configuration. After checking and double-checking all of my port-forwarding through my NAT, I finally broke down and installed FreePBX to get it up and running more easily. Once I got that configuration working, I took all of the useful stuff out their generated configuration files and finally, I had a working system! Next, I’ll walk you through how to set up your own!

Walk-Through: Do it yourself

This walkthrough is based very heavily on these pages, I’ve just compiled the information for you:

• Downloading and Compiling Asterisk 11
• Configuring Asterisk 11 for Google Voice

My set-up is as follows:

• I have an Ubuntu 12.04 server on a LAN behind a NAT.
• The router is a WRT54G running DD-WRT and I use that to assign reserved IPs to the server and the phones.
• I have the following external ports forwarded through my NAT: 5060-5082 (SIP), 5222 (XMPP), 10000-12000 (RTP – note this for rtp.conf).

Currently, the Ubuntu Repository has the last Long-Term Support (LTS) version of Asterisk (1.8) so we need to download and compile Asterisk from source:

I’m going skip the DAHDI instructions in the link because I don’t have any Digium Hardware installed in my server.

# apt-get update && apt-get upgrade -y && reboot
# apt-get install build-essential wget libssl-dev libncurses5-dev libnewt-dev  libxml2-dev linux-headers-$(uname -r) libsqlite3-dev libiksemel-dev

Next, get the source files that we need

# cd /usr/src/
# wget http://downloads.asterisk.org/pub/telephony/libpri/libpri-1.4-current.tar.gz
# wget http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-11-current.tar.gz

Extract the files we’ll need to compile

# tar zxvf libpri*
# tar zxvf asterisk*

Install PRI (for ISDN, you may or may not need this as well):

# cd /usr/src/libpri*
# make && make install

Install Asterisk:

# cd /usr/src/asterisk*
# ./configure && make menuselect && make && make install && make config && make samples

When the “Asterisk Module and Build Option Selection” window pops up, navigate to “Channel Drivers” and ensure “chan_motif” is selected and navigate to “Resource Modules” and ensure “res_xmpp” is selected. Then you can select “Save & Exit” and finish the install. Start Asterisk to ensure it installed properly:

# /etc/init.d/asterisk start

Next, let’s look at configuration. All of the configuration files are located in /etc/asterisk. Let’s navigate there:

# cd /etc/asterisk
# ls

The files we’ll need to change are: extensions.conf, motif.conf, rtp.conf, sip.conf, and xmpp.conf. I trust that you know how to edit files in Linux so I’ll just post the relevant bits below.

extensions.conf: All of your dialing information goes here

[general]

static = yes
writeprotect = no
clearglobalvars = yes

[home]
exten => 200,1,Dial(SIP/200,20)
// Other local extensions go here

// These two expose Google Voice outbound calling with no extension prefix
exten => _1XXXXXXXXXX,1,Dial(Motif/google/${EXTEN}@voice.google.com,,r)
exten => _XXXXXXXXXX,1,Dial(Motif/google/${EXTEN}@voice.google.com,,r)

[incoming-motif] // This is verbatim from the second link I showed above.
exten => s,1,NoOp()
 same => n,Wait(1)
 same => n,Set(crazygooglecid=${CALLERID(name)})
 same => n,Set(stripcrazysuffix=${CUT(crazygooglecid,@,1)})
 same => n,Set(CALLERID(all)=${stripcrazysuffix})
 same => n,Dial(SIP/205&SIP/204,20,D(:1))

motif.conf: Connections using the Motif channel, if you had multiple Google Voice numbers, you would replicate these connections

[google]
context=incoming-motif
disallow=all
allow=ulaw
connection=google

rtp.conf: These are the ports used for UDP transmission, note that these must be forwarded through your NAT

[general]
rtpstart=10000
rtpend=12000
icesupport=yes

sip.conf: This is the most important file, it specifies how all of the calls are set up

[general]
context=home
callerid=Unknown
notifyringing=yes
notifyhold=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
alwaysauthreject=yes
disallow=all
allow=ulaw
allow=alaw
allow=gsm
callevents=no
jbenable=no
maxcallbitrate=384
maxexpiry=3600
minexpiry=60
notifyhold=yes
notifyringing=yes
registerattempts=0
registertimeout=20
rtpholdtimeout=300
rtpkeepalive=0
rtptimeout=30
srvlookup=no
t38pt_udptl=no
checkmwi=10
videosupport=no
canreinvite=no
allowguest=yes
defaultexpiry=120
g726nonstandard=no
nat=force_rport,comedia
// I have a dynamic IP I use dyndns.org for this service
externhost=<yourinternethostname>
externrefresh=120
localnet=192.168.1.1/255.255.255.0

[200]
deny=0.0.0.0/0.0.0.0
secret = <extensionpassword>
dtmfmode = rfc2833
canreinvite=no
context = home
host = dynamic
trustrpid=yes
sendrpid=no
type = friend
nat=no
port=5060
qualify=yes
qualifyfreq=60
transport=udp
encryption=no
callgroup=
pickupgroup=
dial=SIP/200
mailbox=200@home
permit=0.0.0.0/0.0.0.0
callerid = Master Bedroom
callcounter=yes
faxdetect=no

xmpp.conf: Where all of your Google account settings are stored

[google]
type=client
serverhost=talk.google.com
username=<gmailusername>
secret=<gmailpassword>
// Set Priority to at least 25 if you use this account for GMail/Chat
priority=25
port=5222
usetls=yes
usesasl=yes
status=available
statusmessage=Asterisk Phone
timeout=5

This is how I have mine configured. I have tried to highlight the settings which should be specific to your setup, but also keep in mind this is a pretty basic configuration.

Asterisk is still running with the base configuration, so let’s reload that:

# asterisk -rvvv
CLI> reload

Conclusion

This project was a lot of fun and I would recommend my approach to Asterisk: Start with simple tasks and add more complex features bit by bit. For example, my next post is about how to set up multiple Google Accounts on one server.

One of the coolest things about this, is you don’t even have to have your own server. If you ran this configuration on a Raspberry Pi, you could even have a standalone Google Voice phone!

One final thought, as of writing Google Voice provides free calling to the U.S. and Canada until at least the end of 2013. After that, I’m sure the rates will be cheap, but there’s no information about that.

Let me know if you try this and how far it gets you!