Introduction: Free Landline Using Google Voice and a RaspberryPi

About: DIY home automation enthusiast.

Disclaimer: The following article is intended for users comfortable working on Linux based machines.

In this article I'll review the steps I used to configure a VoIP landline using a SIP interface through a Raspberry Pi based PBX with Freeswitch and Google Voice.  In other words, a free landline!

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.

NOTE: You do not have 911 capability with this solution.

Step 1: Things You'll Need

    - A Raspberry PI with "wheezy-raspbian" connected to your network.  Please see the Raspberry Pi Wiki for instructions on how to set this up.
    - A physical 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.  If you decide to use a software solution, like XLITE, then this isn't a "true" landline replacement because you'll need your PC always on.  Using a SIP device, allows
    - A Google voice account.

     - Test your Google voice account, try calling a number from the web UI.

      Then 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)

Step 2: Get Started

Install dependencies on your Raspberry pi.

    #apt-get install autoconf automake gawk g++ git-core libjpeg62-dev libncurses5-dev libtool make python-dev gawk pkg-config libtiff4-dev 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://
    #cd /usr/local/src/freeswitch

    #./ && ./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


    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.

      <!-- Client Profile (Original mode) -->
      <!-- to use this profile take the x- away from the open and close tags so its <profile> and </profile> -->
      <profile type="client">
        <param name="name" value="gtalk"/>
        <param name="login" value=""/>
        <param name="password" value="GMAIL.PASSWORD"/>
        <param name="server" value="" />
        <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=""/>
        <param name="tls" value="true"/>
        <param name="use-rtp-timer" value="false"/>
        <param name="vad" value="none"/>
        <param name="candidate-acl" value=""/>
        <param name="local-network-acl" value=""/>


Step 3: Start Freeswitch

Start freeswitch manually and test module


You should get a console like in the picture .

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

      <!--the domain or ip (the right hand side of the @ in the addr-->
      <domain name="192.168.0.XXX">
          <param name="dial-string" value="{^^:sip_invite_domain=${dialed_domain}:presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(*/${dialed_user}@${dialed_domain})}"/>

          <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"/>

          <group name="public">
              <X-PRE-PROCESS cmd="include" data="default/*.xml"/>


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.

      <user id="2001">
          <param name="password" value="MAKEUPONE"/>
          <param name="vm-password" value="1000"/>
          <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"/>

Step 4: Configure Your SIP Device

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).   NOTE: your configuration page may vary based on your SIP solution.

    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"/>

        <extension name="gvoice_out">
          <condition regex="any">
            <regex field="destination_number" expression="^(\d{10})$" />
            <regex field="dialed_extension" expression="^\+1(\d{10})$" />
            <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$"/>

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

        <configuration name="dingaling.conf" description="XMPP Jingle Endpoint">
            <param name="debug" value="0"/>
            <param name="codec-prefs" value="PCMU"/>

          <X-PRE-PROCESS cmd="include" data="../jingle_profiles/*.xml"/>


    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 (Optional)
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

freeswitch startup script

# Start/stop the freeswitch daemon.
# Provides:          freeswitch
# Required-Start:    $network
# Required-Stop:
# Should-Start:
# Should-Stop:
# Default-Start:     2 3 4 5
# Default-Stop:  0 1 6
# Short-Description:
# Description:


test -f $DAEMON || exit 0

. /lib/lsb/init-functions

# there because it should be in /etc/default/locale.
parse_environment ()
    for ENV_FILE in /etc/environment /etc/default/locale; do
        [ -r "$ENV_FILE" ] || continue
        [ -s "$ENV_FILE" ] || continue

         for var in LANG LANGUAGE LC_ALL LC_CTYPE; do
             value=`egrep "^${var}=" "$ENV_FILE" | tail -n1 | cut -d= -f2`
             [ -n "$value" ] && eval export $var=$value

             if [ -n "$value" ] && [ "$ENV_FILE" = /etc/environment ]; then
                 log_warning_msg "/etc/environment has been deprecated for locale information; use /etc/default/locale for $var=$value instead"

# Get the timezone set.
    if [ -z "$TZ" -a -e /etc/timezone ]; then
        TZ=`cat /etc/timezone`

# Parse the system's environment
if [ "$READ_ENV" = "yes" ] ; then

case "$1" in
start)  log_daemon_msg "Starting freeswitch " "freeswitch"
        ulimit -s 240
        OLDPID=`pgrep $NAME`
        NEWPID=`cat $PIDFILE` 2>/dev/null
        if [ "$OLDPID" =  "$NEWPID" ] ; then
                echo "Service already running " && exit 1
                su - $NAME -c "$DAEMON $DARGS " >/dev/null 2>&1
        sleep 1
        ps -ef|grep $DAEMON|grep -v grep|awk '{print$2}' > $PIDFILE
        log_end_msg $?
stop)   log_daemon_msg "Stopping freeswitch" "freeswitch"
        kill `cat $PIDFILE` >/dev/null 2>&1
        [ $RETVAL -eq 0 ] && [ -e "$PIDFILE" ] && rm -f $PIDFILE
        log_end_msg $RETVAL
restart) log_daemon_msg "Restarting freeswitch" "freeswitch"
        $0 stop
        $0 start
        log_end_msg 0
        status_of_proc -p $PIDFILE $DAEMON $NAME && exit 0 || exit $?
*)      log_action_msg "Usage: /etc/init.d/freeswitch {start|stop|status|restart|reload|force-reload}"
        exit 2
exit 0