How to make and receive calls using Google Voice without XMPP

I.    Introduction

Around 2010, before Asterisk 1.8 and Gtalk, I did some work to set up calls using Google Voice with callback through a DID channel.  I didn’t touch it for quite some time because gtalk with XMPP works very well.  Since Google is going to drop XMPP protocol in May 2014, people started to looking into the old fashioned way again.  Following approaches described in this blog (http://tech.iprock.com/?p=9784) and others’ comments, especially from those by davidnewt, I did some experiments and the set-up is pretty stable now.  I think it will be worthwhile to share with people have the same interest.  Below, I will describe the steps-by-step implementation of such an approach and what I have learned in the process.

The platform that I prefer most is Dockstar, which is an ARM based box, running Debian.  However, since it is no longer widely available, I repeated the installation process using Raspberry Pi with a version of Debian Wheezy.   The installation process should work without modification on any Linux box running Debian Wheezy and Ubuntu 12.04 or later.  With some minor modification, the process is also applicable to any linux systems.  Generally speaking, the GV callback approach can be run for Asterisk running on any ARM and X86 based systems.

To keep the implementation effort to be minimal, I use only plain Asterisk without GUI.  Only the basic system with call-out and incoming call functionalities are implemented.  With some knowledge of Asterisk, the system functionalities can be expanded as much as you like.

Now, let the fun part begin!

II.    The step-by-step guide of setting up an Asterisk PBX with GV callback.

A.    Set up a DID route

There are different ways to have DID services.  I feel the most available and easiest way to set up a DID is to get a DID number from IPKall (http://phone.ipkall.com/) and point it to your registration of a SIP portel.  Among the SIP portels, I had good experience with IPTel (http://www.iptel.org/) and Linphone (http://www.linphone.org/eng/linphone/register-a-linphone-account.html).  All of them are free.  However, you can use any DID route that you prefer.  What you need to do are as follows:

1. Register with a SIP portel of your choice.  Below I will use IPTel in the examples.  After registering, you will have a user name [sipusername], and a password [sippassword].  The web address of IPTel is iptel.org.
2. At the IPKall web page, you can choose an area code, enter sipusername@iptel.org in SIP URI fields, provide a valid email address and choose a password [DIDpassword] for IPKall.
3.  You will receive an email to inform the DID number assigned to you.  Below I will use 253xxxxxxx as the DID number.
4. You can test the DID number by using a softphone and enter your SIP Portel registration information [sipusername, sippassword, iptel.org].  (You can use an ATA with a phone attached to it instead of using the softphone as well.)
5. If the softphone shows you are registered, you can call the DID number from another phone.  If your soft phone rings and you can receive the call, your DID number is ready.

(Note, the IPKall number is free but you have to at least use it once a month.  Otherwise, your number will be taken away.)

B.    Set up Google Voice (GV) and callback number.

If you already have signed up for Google Voice that you want to use, then skip the next steps and go to step 2

1. Go to www.gmail.com and create a gmail account with gvusername and gvpassword.
2.  Once the gmail account is established, go to www.google.com/voice and apply for a new GV number (gvnumber), if you don’t have one already.
3. In www.google.com/voice webpage click at the symbol on the right of the second line and select settings.
4. In the Settings menu, disable call screening in the Calls submenu
5.  In the Phone submenu click on “add a phone” and use the [didnumber] that you got from IPKall in step A.3 as the number you want.  Google will try to verify that you have the number so you will need to have a softphone, or ATA, connect to the SIP account that you registered in step A.1.  Verify the number with GV.  The DID number [253xxxxxxx] will be your GV call back number.

C.    Install Asterisk

I use Asterisk 1.8 and Raspberry as an example.  I have tried Asterisk 11 and everything appeared also work but I didn’t test extensively.  Asterisk 1.6 will probably also work but I haven’t tested it.  It should work on any Linux box running Debian Wheezy.  Other Linux distributions should also work but may need some modifications.

1. SSH into and log on your Raspberry Pi.
2.  apt-get update
3. apt-get install asterisk

For Debian Wheezy, Asterisk 1.8 will be automatically installed and that’s it.

D.    Install Python and pygooglevoice

Do the following under the Linux # prompt:

cd /root
apt-get -f install python-setuptools
easy_install simplejson
wget http://pygooglevoice.googlecode.com/files/pygooglevoice-0.5.tar.gz
tar zxvf pygooglevoice-0.5.tar.gz
cd pygooglevoice-0.5
python setup.py install
sed -i 's|https://www.google.com/accounts/ServiceLoginAuth?service=grandcentral|https://accounts.google.com/ServiceLogin?service=grandcentral\&continue=https://www.google.com/voice|' /usr/local/lib/python2.7/dist-packages/googlevoice/settings.py
sed -i 's|      galx.*|      galx = re.search(r\"name=\\"GALX\\" type=\\"hidden\\"\\n *value=\\"(.+)\\"\", content).group(1)|' /usr/local/lib/python2.7/dist-packages/googlevoice/voice.py

Make a test call using Python at Linux prompt (not Asterisk):

# gvoice -e [gvusername@gmail.com] -p [gvpassword] call NXXNXXXXXX 253xxxxxxx 1

where NXXNXXXXXX is any US number.  You need to make sure the softphone or ATA is registered to the SIP account connected to the IPKall DID number and ready to receive the call.  If it rings, the phone at the other end will ring after you picking up the softphone.

E. Configure Asterisk

1. For the basic calling, all need to be done is replacing two default Asterisk conf files, sip.conf and extensions.conf in the /etc/asterisk/ directory.  The examples of the files are given below.  Of course, you will need to insert your own GV and DID and SIP credentials into the proper place inside these two files.  The Asterisk PBX is set as to have three extensions: 201, 202 and 203.  Any of your softphones or SIP phones registered with the extensions can make outgoing calls.  If there are incoming calls to your GV number or the DID number, the phones registered with 201 and 203 will ring until one of them is picked up.
2. In the sip.conf file, there is a register statement.  After replacing the original files with these two example files, restart the Asterisk by doing a “/etc/init.d/asterisk restart”.  Then do “asterisk –rvvv” to get to the CLI interface.  Inside the CLI interface, type “sip show registry” and hit the return.  It should show you that your SIP service is registered.
3. Leave the CLI interface open and register your softphone with the Asterisk PBX and them make a call.  The CLI will show you all these activities.  Hopefully, you will be able to make a successful call after listen to a few seconds of music :-).
4. In some Asterisk distributions, Music on hold functions may not installed correctly.  If you don’t here music after you dial a number and you really want to, please check the following:
   1. In the musiconhold.conf file, check the directory=… statement in the [default] section.  It is simply “directory=moh”, you may want to change it to “directory=/var/lib/asterisk/moh”.
   2. Look if there are music files in /var/lib/asterisk/moh directory with extensions of .gsm or .wav.  If there’s no files, you can get the files from here, or you can use your favorite music files.

Good luck!

III.    Acknowledgements and Disclaimers

First the acknowledgement:

I am not a network/software/VOIP person.  All of what I did are based the knowledge that I gained from Googling and exchanging with other more experienced people.  Thus, nothing described in this note is invented by myself.  All the credit should go to the people who proposed this approach and worked out the details, specifically, the authors, commenters and contributors of this blog mentioned previously.  What I did is just to make sure all of details that I described are correct and to verify it indeed works.  I would appreciate to have your feedback and comments.  If you have any questions, please let me know also.  I will try to answer them as much as I can.

Now the disclaimer:

As stated above, what I did is simply trying the GV callback approach designed and described by others.  It is experimental to say the most.  As we all know, whenever one goes online including performing VOIP activities, there are always security risks.  Thus, if you decide to follow what described in this blog, you should know the risks involved and you will be responsible for whatever action you are taken.  You will be responsible for any consequences of your actions.

Another point is that we don’t know what Google is going to do after May 15.  Even though the set-up following the above descriptions works today, there’s no guarantee it will work at that time.  We just have to hope it will.

IV.    Appendix

The configure files:

sip.conf (source)

[general]
context=phone                 ; Default context for incoming calls
allowoverlap=no                 ; Disable overlap dialing support. (Default is yes)
bindport=5060                  ; UDP Port to bind to (SIP standard port is 5060)
externrefresh=60
localnet=192.168.1.0/255.255.255.0
udpbindaddr=0.0.0.0             ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
canreinvite=no
dtmfmode = rfc2833
tcpenable=no
;directmedia=yes
register => sipusername:sippassword@iptel.org:5060/253xxxxxxx
session-timers=refuse
disallow=all
allow=ulaw
allow=gsm

[201]
defaultuser=201
secret=secret1
type=peer
callerid="user1 "
host=dynamic
context=phone
outgoinglimit=1
incominglimit=1
canreinvite=no
nat=yes
qualify=yes

[202]
defaultuser=202
secret=secret2
type=peer
callerid="user2 "
host=dynamic
context=phone
outgoinglimit=1
incominglimit=1
canreinvite=no
nat=yes
qualify=yes

[203]
defaultuser=203
secret=secret3
type=peer
callerid="user3 "
host=dynamic
context=phone
outgoinglimit=1
incominglimit=1
canreinvite=no
nat=yes
qualify=yes

[DID]
context=sip
defaultuser=sipusername
type=peer
secret=sippassword
host=iptel.org
fromdomain=iptel.org
fromuser=253xxxxxxx
trustrpid = yes
sendrpid = yes
canreinvite = no
insecure=port,invite
nat=yes

extensions.conf (source)

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

[globals]
gtimeout=50    ; timeout value
; initialize
gvuser=10000

[sip]
exten => _253xxxxxxx,1,ExecIf($[${gvuser}!=10000]?Bridge(${gvuser}):Dial(SIP/201&SIP/203,60,D(:1)))
exten => _253xxxxxxx,n, Set(GLOBAL(gvuser)=10000)
exten => _253xxxxxxx, n, Hangup()
[phone]
include => sip
include => gv-outbound
[gv-outbound]
exten => _NXXNXXXXXX,1,GoTo(1${EXTEN},1)
exten => _1NXXNXXXXXX,1,Answer
exten => _1NXXNXXXXXX,n,Set(GLOBAL(gvuser)=${CHANNEL})
exten => _1NXXNXXXXXX,n,System(gvoice -e gvusername@gmail.com -p gvpassword call ${EXTEN} 1253xxxxxxx 1 &)
exten => _1NXXNXXXXXX,n,Ringing
exten => _1NXXNXXXXXX,n,Wait(30)
exten => _X.,n,Noop(Never received callback from Google Voice on channel ${gvuser} . exiting)
exten => h,1,GotoIf($["${CHANNEL(state)}" = "Ring"]?:bridged)
exten => h,n,Noop(Hangup on channel ${gvuser})
exten => h,n,System(gvoice -e gvusername@gmail.com -p gvpassword cancel &)
exten => h,n, Set(GLOBAL(gvuser)=10000)
exten => h,n,Hangup()
exten => h,n(bridged),Noop(The channel has been bridged successfully)
exten => h,n, Set(GLOBAL(gvuser)=10000)