Sunday, December 30, 2012

Using Google Voice as a landline


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://
      #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.
    • <include>
        <!-- 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=""/>

  • 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">
            <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.
    • <include>
        <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"/>

  • 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="$1"/>-->
          <action application="set" data="hangup_after_bridge=true" />
          <!--<action application="set" data="originate_continue_on_timeout=true"/>-->
          <!--<action application="set" data="call_timeout=35"/>-->
          <action application="bridge" data="user/2001@192.168.0.XXX"/>
          <action application="answer"/>
      <extension name="gvoice_out">
        <condition regex="any">
          <regex field="destination_number" expression="^(\d{10})$" />
          <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
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