Raspberry Pi - My Hell & Fixes: September 2013

tep 1
Plug it in. Doesn’t matter which port – just plug the thing in.
Step 2
Boot it up. I’m assuming here you have used your Pi before and know how to get into an X session.
Step 3
On your desktop you’ll see an icon like this:

Double-click on it – you’ll see a new window launched entitled wpa_gui (you can also launch this by running sudo wpa_gui from the terminal). This is what it’ll look like:

Step 4
Now here’s where you actually have to do something. As you’ll notice in the image above, there is a single entry in the Adapter list – wlan0. This is the N150 wireless dongle (unless you have other wireless devices). If you see this, you’re OK to proceed.
Next, click on the Scan button. This will bring up a new window:

Step 5
Once you have this window up and you can see your wireless network (you might need to click scan again), double-click on it. You’ll be presented with the following window:

Assuming you don’t want to change any settings (I use MAC filtering rather than encryption, so it was easy enough – I’m assuming it’ll be fairly straight-forward to set up encryption), just click on the Add button. This will add the selected wireless network to a list you want be able to use.
Step 6
Once you’ve done this, close down this winder and the previous one until you return to the main wpa_gui window. You’ll see you now have your wireless network in the Network list, and looking below that you’ll see a status of Completed (station).

And that’s pretty much it. I’ve been using it all evening without any issue. Hopefully this’ll get you up and running as easily as it did me

taken from : http://www.myraspberrypiexperience.com/

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:

Raspberry PI with "wheezy-raspbian" connected to your network.
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.
A Google voice account.
Test your Google voice account, try calling a number from the web UI.
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

freeswitch startup script

#!/bin/sh
# Start/stop the freeswitch daemon.
#
### BEGIN INIT INFO
# 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:
### END INIT INFO

PATH=/bin:/usr/bin:/sbin:/usr/sbin
DESC="freeswitch"
NAME=freeswitch
DAEMON=/usr/local/freeswitch/bin/freeswitch
DARGS="-nc"
PIDFILE=/var/run/freeswitch.pid
SCRIPTNAME=/etc/init.d/"$NAME"

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"
             fi
         done
     done



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

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


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
        else
                su - $NAME -c "$DAEMON $DARGS " >/dev/null 2>&1
        fi
        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 $PIDFILE` >/dev/null 2>&1
        RETVAL=$?
        [ $RETVAL -eq 0 ] && [ -e "$PIDFILE" ] && rm -f $PIDFILE
        log_end_msg $RETVAL
        ;;
restart) log_daemon_msg "Restarting freeswitch" "freeswitch"
        $0 stop
        $0 start
        log_end_msg 0
        ;;
status)
        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
        ;;
esac
exit 0

Taken From http://raspberrypipbx.blogspot.co.uk/

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