Using Google Voice as a landline

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Have you ever wanted to stop paying for a landline but didn't know how?  In this article I'll review the steps I used to configure a VoIP land line using a SIP interface through a Raspberry Pi based PBX with Freeswitch and Google Voice.  In other words, make free calls!  NOTE: You will not have 911/Emergency services with this solution!

I spent the better part of the weekend reading wiki's, blogs, manuals and other posts and had to piece them all together to get my own personal solution working.  I thought it would be useful to write it up and share my experiences.

I've tried to capture my steps as best as I could .  Please let me know if anything does not work or needs to be amended.

Things you'll need:

1. Raspberry PI with "wheezy-raspbian" connected to your network.
2. A SIP device or software, like XLITE.  I'm using an old Sipura SPA-2100.  One end connected to your physical phone line and the other to your network.
3. A Google voice account.
4. Test your Google voice account, try calling a number from the web UI.
5. Log into Google Voice:

• Click on Settings > Phones
• Uncheck all phones
• Check Google Chat
• Log out of gmail ( Or turn off chat at the bottom of the gmail page)

Steps: (on your Raspberry Pi as root)

• Install dependencies 

• #apt-get install autoconf automake gawk g++ git-core libjpeg62-dev libncurses5-dev libtool make python-dev gawk pkg-config  libperl-dev libgdbm-dev libdb-dev libssl-dev

• Download, Compile and install freeswitch. NOTE: This step takes a few hours to compile.  

• #mkdir /usr/local/freeswitch 
  #useradd freeswitch -d /usr/local/freeswitch
  #chown -R freeswitch:freeswitch /usr/local/freeswitch

• #cd /usr/local/src
  #git clone git://git.freeswitch.org/freeswitch.git
  #cd /usr/local/src/freeswitch

  #./bootstrap.sh && ./configure --prefix=/usr/local/freeswitch && make clean && make clean modwipe && make && make install

• Make sure the following line is present and uncommented in /usr/src/freeswitch/modules.conf

  endpoints/mod_dingaling
  

  And build mod_dingaling:

  #make mod_dingaling-install

• Make sure mod_dingaling is not commented out in file conf/autoload_configs/modules.conf.xml

•  <load module="mod_dingaling"/>

• Edit the conf/jingle_profiles/client.xml and replace all its contents with the following.  Then replace  only the highlighted fields with your Gmail username and password.

• <include>
    <!-- Client Profile (Original mode) -->
    <!-- to use this profile take the x- away from the open and close tags so its <profile> and </profile> -->
  <include>
    <profile type="client">
      <param name="name" value="gtalk"/>
      <param name="login" value="YOUR_GMAIL@gmail.com/talk"/>
      <param name="password" value="GMAIL.PASSWORD"/>
      <param name="server" value="talk.google.com" />
      <param name="message" value="Thanks Google!" />
      <param name="dialplan" value="XML"/>
      <param name="context" value="default"/>
      <param name="exten" value="2001"/>
      <param name="rtp-ip" value="auto"/>
      <param name="auto-login" value="true"/>
      <param name="sasl" value="plain"/>
      <param name="server" value="talk.google.com"/>
      <param name="tls" value="true"/>
      <param name="use-rtp-timer" value="false"/>
      <param name="vad" value="none"/>
      <param name="candidate-acl" value="wan.auto"/>
      <param name="local-network-acl" value="localnet.auto"/>
   </profile>
  
  </include>
  

  
  
• Start freeswitch manually and test module

• #/usr/local/freeswitch/bin/freeswitch 
• You should get a console like this:

• Try running reload mod_dingaling  if you entered the correct credentials and all was compiled 
• correctly, you should see the following message:

• 
  +OK Reloading XML
  +OK module unloaded
  +OK module loaded
  
  freeswitch@pbx> 2012-12-30 19:52:59.136376 [NOTICE] libdingaling.c:1674 XMPP server connected
  
  2012-12-30 19:52:59.356369 [NOTICE] libdingaling.c:1686 XMPP authenticated
  

  
  
• Exit the freeswitch console using the shutdown command, and return to the shell prompt.
• Edit the conf/directory/default.xml and replace all its contents with the following.  Then replace  only the highlighted fields with your SIP device IP address

• <include>
    <!--the domain or ip (the right hand side of the @ in the addr-->
    <domain name="192.168.0.XXX">
      <params>
        <param name="dial-string" value="{^^:sip_invite_domain=${dialed_domain}:presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(*/${dialed_user}@${dialed_domain})}"/>
      </params>
  
      <variables>
        <variable name="record_stereo" value="true"/>
        <variable name="default_gateway" value="$${default_provider}"/>
        <variable name="default_areacode" value="$${default_areacode}"/>
        <variable name="transfer_fallback_extension" value="operator"/>
      </variables>
  
      <groups>
        <group name="public">
          <users>
            <X-PRE-PROCESS cmd="include" data="default/*.xml"/>
          </users>
        </group>
      </groups>
  
    </domain>
  </include>
  
  

  
  

• Create a file called conf/directory/default/2001.xml and paste the following contents, replacing only the highlighted fields with any random password (save it for later) and your name, which will be used for caller id.

• <include>
    <user id="2001">
      <params>
        <param name="password" value="MAKEUPONE"/>
        <param name="vm-password" value="1000"/>
      </params>
      <variables>
        <variable name="toll_allow" value="domestic,international,local"/>
        <variable name="user_context" value="default"/>
        <variable name="effective_caller_id_name" value="John Doe"/>
        <variable name="effective_caller_id_number" value="2001"/>
        <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/>
        <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>
        <variable name="callgroup" value="default"/>
      </variables>
    </user>
  </include>
  

  
  
• Login to your SIP (Or SIP software), and enter the following . (NOTE I will only demonstrate on the SPA-2100, other solutions will have different screens).  
• In the proxy (or domain) type in the IP address or hostname of your freeswitch PBX.
• The Display Name can be anything you want.
• User ID should be 2001 (because that is what I set it to).
• The passowrd  is from the one you made up above.

• Start freeswitch on the command line /usr/local/freeswitch/bin/freeswitch
• Save and reboot the SIP device.  You SIP device should now be registered. You can test by way of hearing a dial tone on your phone.
• Open another login window to your PBX and leave the previous freeswitch console open in the the other window. We'll reference it later.
• Edit the conf/dialp/default.xml and add the following, right after the "unloop" extension section.  Then replace only the highlighted fields with your SIP device IP address.

•  <extension name="gvoice_in">
    <condition field="source" expression="^mod_dingaling$">
      <!--<action application="info" />-->
      <action application="log" data="CONSOLE GV CALL IN!" />
      <action application="log" data="CONSOLE ${destination_number}"/>
      <action application="start_dtmf" />
      <action application="set" data="execute_on_answer=send_dtmf 1@2001"/>
      <!--<action application="cidlookup" data="$1"/>-->
      <action application="set" data="hangup_after_bridge=true" />
      <!--<action application="set" data="originate_continue_on_timeout=true"/>-->
      <!--<action application="set" data="call_timeout=35"/>-->
      <action application="bridge" data="user/2001@192.168.0.XXX"/>
      <action application="answer"/>
    </condition>
  </extension>
  
  <extension name="gvoice_out">
    <condition regex="any">
      <regex field="destination_number" expression="^(\d{10})$" />
      <regex field="dialed_extension" expression="^\+1(\d{10})@voice.google.com$" />
      <regex field="destination_number" expression="\+1(\d{10})$" />
      <action application="set" data="hangup_after_bridge=true"/>
      <action application="set" data="ringback=${us-ring}"/>
      <action application="set" data="call_timeout=45" />
      <action application="ring_ready"/>
      <action application="bridge" data="dingaling/gtalk/+1$1@voice.google.com"/>
    </condition>
  </extension>
  
  

• Edit the conf/autoload_configs/dingaling.conf.xml and replace all its contents with the following.  

• <configuration name="dingaling.conf" description="XMPP Jingle Endpoint">
    <settings>
      <param name="debug" value="0"/>
      <param name="codec-prefs" value="PCMU"/>
    </settings>
  
    <X-PRE-PROCESS cmd="include" data="../jingle_profiles/*.xml"/>
  
  </configuration>
  
  
  

• Go back to the freeswitch console window and type reloadxml  then  reload mod_dingaling 
• Now you are can test making outgoing and receiving calls.
• If all works to your satisfaction, its time to make freespace start automatically at boot time.

Creating a Freeswitch boot service

• Create a new file called /etc/init.d/freeswitch paste the contents from this file.

#chown -R freeswitch:freeswitch /etc/init.d/freeswitch

#chmod +x /etc/init.d/freeswitch

#update-rc.d  freeswitch defaults

#/etc/init.d/freeswitch start

• Reboot the pbx and check that the daemon started successfully and that everything is working.

Voice Codec Optimization
I found that the default codec causes outgoing calls to be choppy.  To fix this, I changed the default codec to PCMA.

Replace the following in conf/vars.xml

<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>

with the following

<X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMA"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMA"/>

Save the file and stop and start the service

#service freeswitch stop  && sleep 15 && service freeswitch start