A Google Voice Landline with Asterisk

June 4, 2011

Inspired by a recent Maximum PC article, I recently set up a land line telephone that supports both incoming and outgoing calls from my Google Voice number. I opted to use an installation of Asterisk in on a standard Ubuntu server instead of using one of the pre-built PBX Linux distributions. The process involved enough fiddling and Google searching that I thought a summary of how my setup works could be helpful for others.

You can probably guess by the recent addition of calling to Gmail that Google is using VoIP protocols to shuffle Voice traffic around. Thankfully, they're somewhat standard, and Asterisk recently added official support for it. Just like the calling feature in Gmail, an Asterisk server can connect to Google and send/receive calls.

Chef Cookbook

I've written a Chef cookbook that does most of this configuration automatically. I'll go over the basics of the cookbook if you don't use Chef.

Install the Package

The best place to get the Asterisk package is the official asterisk.org repository. Add it to your /etc/apt/sources.list:

deb http://packages.asterisk.org/deb natty main

Then install the PGP key and update your package lists:

sudo apt-key adv --keyserver subkeys.pgp.net --recv-keys 175E41DF
sudo apt-get update

Finally, install the two packages we need:

sudo apt-get install asterisk-1.8 asterisk-dahdi

Configuration

There are five configuration files to change:

`/etc/asterisk/sip.conf`

Set localnet to the IP range of your LAN, e.g. 192.168.1.0/255.255.255.0
tcpenable = 'yes'
disallow = 'all'
allow = ... for ulaw, gsm, ilbc, and speex
externip must be set to your external (i.e. WAN) IP address

Finally, in the [authentication] section, add the details for your user account:

[DESIRED_ASTERISK_USERNAME]
secret=YOUR_DESIRED_PASSWORD
type=friend
callerid="Your Name <username>"
host=dynamic
context=outbound

`/etc/asterisk/extensions.conf`

This file sets up all routing paths for getting calls from Google to your user. This configuration is probably a bit more than is required (and is a mashup of a few different blog posts), but it's working for me at the moment:

[general]
static=yes
writeprotect=no
clearglobalvars=no

[globals]
CONSOLE=Console/dsp             ; Console interface for demo
IAXINFO=guest                   ; IAXtel username/password
TRUNK=Zap/G2                    ; Trunk interface
TRUNKMSD=1                  ; MSD digits to strip (usually 1 or 0)

[default]
exten => s,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
exten => s,n,Dial(SIP/YOUR_ASTERISK_USERNAME, 10)
exten => s,n, Hangup
exten => YOUR_ASTERISK_USERNAME, 1, Dial(SIP/YOUR_ASTERISK_USERNAME, 10)

[google-in]
exten => YOUR_ASTERISK_USERNAME, 1, GotoIf(${DB_EXISTS(gv_dialout/channel)}?bridged)
exten => YOUR_ASTERISK_USERNAME, n, NoOp(Callerid  ${CALLERID(name)})
exten => YOUR_ASTERISK_USERNAME, n, Set(CALLERID(num)=${SHIFT(CALLERID(name),@)})
exten => YOUR_ASTERISK_USERNAME, n, Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
exten => YOUR_ASTERISK_USERNAME, n, Dial(SIP/YOUR_ASTERISK_USERNAME, 20, aD(:1))
exten => YOUR_ASTERISK_USERNAME, n(bridged),Bridge(${DB_DELETE(gv_dialout/channel)}, p)

[outbound]
include => seven-digit
include => local-devices
include => tollfree
include => talk-gmail-outbound
include => talk-numeric-outbound
include => dial-uri

[local-devices]
exten => YOUR_ASTERISK_EXTENSION_NUM, 1, Dial(SIP/YOUR_ASTERISK_USERNAME, 10)

[tollfree]
exten => _411, 1, Dial(SIP/18004664411@proxy.ideasip.com,60)
exten => _1800NXXXXXX,1,Dial(SIP/${EXTEN}@proxy.ideasip.com,60)
exten => _1888NXXXXXX,1,Dial(SIP/${EXTEN}@proxy.ideasip.com,60)
exten => _1877NXXXXXX,1,Dial(SIP/${EXTEN}@proxy.ideasip.com,60)
exten => _1866NXXXXXX,1,Dial(SIP/${EXTEN}@proxy.ideasip.com,60)

[seven-digit]
exten => _NXXXXXX,1,Set(CALLERID(dnid)=1512${CALLERID(dnid)})
exten => _NXXXXXX,n,Goto(1512${EXTEN},1)
exten => _NXXNXXXXXX,1,Set(CALLERID(dnid)=1${CALLERID(dnid)})
exten => _NXXNXXXXXX,n,Goto(1${EXTEN},1)

[talk-gmail-outbound]
exten => _[a-z].@gmail.com,1,Dial(gtalk/google/${EXTEN}@gmail.com)
exten => _[A-Z].@gmail.com,1,Dial(gtalk/google/${EXTEN}@gmail.com)

[talk-numeric-outbound]
exten => _1XXXXXXXXXX,1,Dial(gtalk/google/+${EXTEN}@voice.google.com)
exten => _+1XXXXXXXXXX,1,Dial(gtalk/google/+${EXTEN}@voice.google.com)

[dial-uri]
exten => _[a-z].,1,Dial(SIP/${EXTEN}@${SIPDOMAIN},120,tr)
exten => _[A-Z].,1,Dial(SIP/${EXTEN}@${SIPDOMAIN},120,tr)
exten => _X.,1,Dial(SIP/${EXTEN}@${SIPDOMAIN},120,tr)

`/etc/asterisk/jabber.conf`

This file configures the authentication details for your Google account (Jabber is the protocol used by GTalk, and it's technically how you log in). This file contains your Google password, so make sure it's only readable by your user or root.

[general]
autoregister=yes

[google]
type=client
serverhost=talk.google.com
username=YOUR_GOOGLE_ACCOUNT@gmail.com/Talk
secret=YOUR_GOOGLE_PASSWORD
port=5222
statusmessage=Asterisk Server - Not a Human
status=xaway

`/etc/asterisk/gtalk.conf`

Finally, this file links together your Google account (defined previous in the jabber.conf file) with Google Voice.

[general]
context=google-in       ; Context to dump call into
allowguest=yes

[guest]         ; special account for options on guest account
disallow=all
allow=ulaw

[gtalk]
username=YOUR_GOOGLE_USER@gmail.com
disallow=all
allow=ulaw
context=google-in
connection=google

Google Voice Configuration

The last step is to enable Google Talk as an extension in your Google Voice account. This should be pretty straightforward.

Caveats

There are a few known issues with this setup.

The Google account you use will appear logged into Google Talk anytime the server is running. Messages sent to the server will just get lost in the ether. I set up a separate Google account with my Voice number so this wouldn't be an issue. If you already have an established Voice number that you don't want to change, another option is to create a second Google account for chat only - although then you lose the ability to use the inlined chat in Gmail. If anyone has a better solution for this, I'd love to know about it.

Incoming calls are somewhat unreliable. Sometimes the phone will ring, but after picking it up you just get dead air. My cell phone keeps ringing when this happens so I'm still able to pick up the call. It's annoying, however, and I'd love to find the root cause.

I suspect that Google's requirement that you hit "1" when accepting a call via SIP is the problem. This Asterisk config attempts to do that automatically for you, but it took me quite a while to nail down that process and it's most likely still not working. Sometimes jabbing "1" when I get dead air seems to wake it up, but not every time.

Otherwise, outgoing calls work quite well. The only issue I've ever had is if I'm using all of my upstream bandwidth doing something else, but that's not unique to Asterisk. Some QoS controls in your router should help out with that.