Incoming SIP INVITE
       │
       ▼
   mod_sofia (SIP stack)
       │
       ▼
   Profile (internal or external)
       │
       ▼
   Context (default or public)
       │
       ▼
   Extension matching (top-to-bottom)
       │
       ▼
   Condition evaluation (regex)
       │
       ▼
   Actions execute (answer, bridge, playback, etc.)

/etc/freeswitch/                  # All configuration
├── freeswitch.xml                # Master config (includes everything)
├── vars.xml                      # Global variables (domain, passwords, IPs)
├── autoload_configs/             # Module configurations
│   ├── modules.conf.xml          # Which modules to load
│   ├── sofia.conf.xml            # SIP module config (includes profiles)
│   ├── conference.conf.xml       # Conference profiles
│   ├── voicemail.conf.xml        # Voicemail settings
│   ├── ivr.conf.xml              # IVR menu definitions
│   ├── cdr_csv.conf.xml          # CDR output config
│   └── event_socket.conf.xml     # ESL listener config
├── sip_profiles/                 # SIP profile definitions
│   ├── internal.xml              # Internal profile (phones, port 5060)
│   ├── external.xml              # External profile (trunks, port 5080)
│   └── external/                 # Gateway (trunk) definitions
│       └── my_provider.xml       # One file per trunk
├── dialplan/                     # Call routing rules
│   ├── default.xml               # Internal context (phone-to-phone)
│   ├── public.xml                # External context (inbound from trunks)
│   └── default/                  # Additional dialplan fragments
├── directory/                    # User/phone definitions
│   └── default/                  # Domain directory
│       ├── 1000.xml              # Extension 1000
│       ├── 1001.xml              # Extension 1001
│       └── ...
└── lang/                         # Language/sound file mappings

/var/lib/freeswitch/              # Runtime data
├── db/                           # SQLite databases (registrations, etc.)
├── recordings/                   # Call recordings
├── storage/                      # Voicemail, fax storage
└── sounds/                       # Audio/prompt files

/var/log/freeswitch/              # Logs
├── freeswitch.log                # Main log (rotated)
└── cdr-csv/                      # CDR files

# Install prerequisites
apt-get update && apt-get install -y gnupg2 wget lsb-release

# Set your SignalWire PAT
TOKEN="YOUR_SIGNALWIRE_PAT_HERE"

# Add SignalWire GPG key
wget --http-user=signalwire --http-password=$TOKEN \
  -O /usr/share/keyrings/signalwire-freeswitch-repo.gpg \
  https://freeswitch.signalwire.com/repo/deb/debian-release/signalwire-freeswitch-repo.gpg

# Add repository
echo "machine freeswitch.signalwire.com login signalwire password $TOKEN" \
  > /etc/apt/auth.conf.d/freeswitch.conf
chmod 600 /etc/apt/auth.conf.d/freeswitch.conf

echo "deb [signed-by=/usr/share/keyrings/signalwire-freeswitch-repo.gpg] \
  https://freeswitch.signalwire.com/repo/deb/debian-release/ bookworm main" \
  > /etc/apt/sources.list.d/freeswitch.list

# Install prerequisites
apt-get update && apt-get install -y gnupg2 wget lsb-release

# Set your SignalWire PAT
TOKEN="YOUR_SIGNALWIRE_PAT_HERE"

# Add SignalWire GPG key
wget --http-user=signalwire --http-password=$TOKEN \
  -O /usr/share/keyrings/signalwire-freeswitch-repo.gpg \
  https://freeswitch.signalwire.com/repo/deb/debian-release/signalwire-freeswitch-repo.gpg

# Add repository
echo "machine freeswitch.signalwire.com login signalwire password $TOKEN" \
  > /etc/apt/auth.conf.d/freeswitch.conf
chmod 600 /etc/apt/auth.conf.d/freeswitch.conf

echo "deb [signed-by=/usr/share/keyrings/signalwire-freeswitch-repo.gpg] \
  https://freeswitch.signalwire.com/repo/deb/debian-release/ noble main" \
  > /etc/apt/sources.list.d/freeswitch.list

apt-get update

# Full install (recommended for learning — includes all modules)
apt-get install -y freeswitch-meta-all

# OR minimal install (production — add modules as needed)
# apt-get install -y freeswitch-meta-vanilla

# Enable and start FreeSWITCH
systemctl enable freeswitch
systemctl start freeswitch

# Verify it is running
systemctl status freeswitch

# Check ownership (FreeSWITCH runs as freeswitch user)
ls -la /etc/freeswitch/
ls -la /var/lib/freeswitch/
ls -la /var/log/freeswitch/

# Fix ownership if needed
chown -R freeswitch:freeswitch /etc/freeswitch
chown -R freeswitch:freeswitch /var/lib/freeswitch
chown -R freeswitch:freeswitch /var/log/freeswitch

# Install build dependencies
apt-get update && apt-get install -y \
  build-essential cmake automake autoconf libtool pkg-config \
  libssl-dev zlib1g-dev libdb-dev libexpat1-dev libcurl4-openssl-dev \
  libpcre3-dev libspeex-dev libspeexdsp-dev libsqlite3-dev \
  libedit-dev libldns-dev libpq-dev libtiff-dev libjpeg-dev \
  libavformat-dev libswscale-dev liblua5.3-dev \
  libopus-dev libsndfile1-dev uuid-dev \
  python3-dev erlang-dev yasm nasm \
  git wget unzip

# Clone the repository
cd /usr/local/src
git clone https://github.com/signalwire/freeswitch.git -b v1.10 freeswitch
cd freeswitch

# Install libks and signalwire-c dependencies
git clone https://github.com/signalwire/libks.git
cd libks && cmake . -DCMAKE_INSTALL_PREFIX=/usr && make install && cd ..

git clone https://github.com/signalwire/signalwire-c.git
cd signalwire-c && cmake . -DCMAKE_INSTALL_PREFIX=/usr && make install && cd ..

# Bootstrap and configure
cd /usr/local/src/freeswitch
./bootstrap.sh -j

# Edit modules.conf to enable/disable modules before building
# nano modules.conf

./configure --prefix=/usr/local/freeswitch

# Build and install
make -j$(nproc)
make install

# Install sounds and music on hold
make cd-sounds-install cd-moh-install

# Create system user
useradd -r -s /bin/false freeswitch
chown -R freeswitch:freeswitch /usr/local/freeswitch

# SIP signaling (internal profile)
ufw allow 5060/tcp comment "FreeSWITCH SIP TCP internal"
ufw allow 5060/udp comment "FreeSWITCH SIP UDP internal"

# SIP signaling (external profile)
ufw allow 5080/tcp comment "FreeSWITCH SIP TCP external"
ufw allow 5080/udp comment "FreeSWITCH SIP UDP external"

# RTP media (voice/video)
ufw allow 16384:32768/udp comment "FreeSWITCH RTP media"

# ESL (Event Socket Layer) - restrict to trusted IPs only
ufw allow from 10.0.0.0/8 to any port 8021 proto tcp comment "FreeSWITCH ESL"

# WebRTC (if using mod_verto)
# ufw allow 8081/tcp comment "FreeSWITCH Verto WSS"
# ufw allow 8082/tcp comment "FreeSWITCH Verto WSS"

ufw enable

# SIP
iptables -A INPUT -p udp --dport 5060 -j ACCEPT
iptables -A INPUT -p tcp --dport 5060 -j ACCEPT
iptables -A INPUT -p udp --dport 5080 -j ACCEPT
iptables -A INPUT -p tcp --dport 5080 -j ACCEPT

# RTP
iptables -A INPUT -p udp --dport 16384:32768 -j ACCEPT

# ESL (restrict source)
iptables -A INPUT -p tcp -s 10.0.0.0/8 --dport 8021 -j ACCEPT
iptables -A INPUT -p tcp --dport 8021 -j DROP

# Connect to FreeSWITCH CLI
fs_cli

# Inside fs_cli, check status
freeswitch@server> status

# Expected output:
# UP 0 years, 0 days, 0 hours, 5 minutes, 23 seconds, 456 milliseconds, 789 microseconds
# FreeSWITCH (Version 1.10.x ...) is ready
# 0 session(s) since startup
# 0 session(s) - peak 0, last 5min 0
# 0 session(s) per Sec out of max 30, peak 0, last 5min 0
# 1000 session(s) max
# min idle cpu 0.00/99.67

# Check SIP profiles
freeswitch@server> sofia status

# Expected output:
#                      Name          Type                               Data      State
# =================================================================================================
#                  internal       profile   sip:mod_sofia@YOUR_SERVER_IP:5060      RUNNING (0)
#                  external       profile   sip:mod_sofia@YOUR_SERVER_IP:5080      RUNNING (0)
# ... (other profiles)

# Check loaded modules
freeswitch@server> module_exists mod_sofia
# true

# Exit CLI
freeswitch@server> /exit

# Edit global variables
nano /etc/freeswitch/vars.xml

<!-- CHANGE THIS — default password for all extensions -->
<X-PRE-PROCESS cmd="set" data="default_password=PUT_A_STRONG_PASSWORD_HERE"/>

<!-- Domain — set to your server IP or FQDN -->
<X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/>

nano /etc/freeswitch/autoload_configs/event_socket.conf.xml

<configuration name="event_socket.conf" description="Socket Client">
  <settings>
    <param name="nat-map" value="false"/>
    <param name="listen-ip" value="127.0.0.1"/>
    <param name="listen-port" value="8021"/>
    <!-- CHANGE THIS from ClueCon -->
    <param name="password" value="YOUR_SECURE_ESL_PASSWORD"/>
    <!--<param name="apply-inbound-acl" value="loopback.auto"/>-->
  </settings>
</configuration>

# From fs_cli
fs_cli -x "reloadxml"

# Or restart the service
systemctl restart freeswitch

<profile name="internal">
  <settings>
    <!-- Network -->
    <param name="sip-ip" value="$${local_ip_v4}"/>
    <param name="sip-port" value="5060"/>
    <param name="rtp-ip" value="$${local_ip_v4}"/>

    <!-- NAT: Set this to your public IP if behind NAT -->
    <!-- <param name="ext-sip-ip" value="YOUR_PUBLIC_IP"/> -->
    <!-- <param name="ext-rtp-ip" value="YOUR_PUBLIC_IP"/> -->

    <!-- Dialplan context for calls arriving on this profile -->
    <param name="context" value="default"/>

    <!-- Codec preferences (in order) -->
    <param name="inbound-codec-string" value="OPUS,G722,PCMU,PCMA"/>
    <param name="outbound-codec-string" value="OPUS,G722,PCMU,PCMA"/>

    <!-- DTMF handling -->
    <param name="dtmf-type" value="rfc2833"/>

    <!-- Registration -->
    <param name="inbound-reg-force-matching-username" value="true"/>
    <param name="auth-calls" value="true"/>
    <param name="apply-nat-acl" value="nat.auto"/>

    <!-- Recording -->
    <param name="record-path" value="$${recordings_dir}"/>
    <param name="record-template" value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>

    <!-- Timers -->
    <param name="session-timeout" value="1800"/>
    <param name="rtp-timeout-sec" value="300"/>
    <param name="rtp-hold-timeout-sec" value="1800"/>

    <!-- Hold music -->
    <param name="hold-music" value="$${hold_music}"/>
  </settings>
</profile>

<profile name="external">
  <settings>
    <param name="sip-ip" value="$${local_ip_v4}"/>
    <param name="sip-port" value="5080"/>
    <param name="rtp-ip" value="$${local_ip_v4}"/>

    <!-- NAT: uncomment and set if behind NAT -->
    <!-- <param name="ext-sip-ip" value="YOUR_PUBLIC_IP"/> -->
    <!-- <param name="ext-rtp-ip" value="YOUR_PUBLIC_IP"/> -->

    <!-- Inbound calls from trunks go to 'public' context -->
    <param name="context" value="public"/>

    <!-- Codecs -->
    <param name="inbound-codec-string" value="PCMU,PCMA,G722"/>
    <param name="outbound-codec-string" value="PCMU,PCMA,G722"/>

    <!-- Do NOT require authentication (trunks use IP-based auth) -->
    <param name="auth-calls" value="false"/>

    <!-- DTMF -->
    <param name="dtmf-type" value="rfc2833"/>

    <!-- Aggressively reclaim failed channels -->
    <param name="rtp-timeout-sec" value="300"/>
  </settings>

  <!-- Gateway definitions are in sip_profiles/external/ directory -->
  <gateways>
    <X-PRE-PROCESS cmd="include" data="external/*.xml"/>
  </gateways>
</profile>

<include>
  <user id="1001">
    <params>
      <param name="password" value="Str0ng_P@ss_1001!"/>
      <param name="vm-password" value="1001"/>
    </params>
    <variables>
      <variable name="toll_allow" value="domestic,international,local"/>
      <variable name="accountcode" value="1001"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="John Smith"/>
      <variable name="effective_caller_id_number" value="1001"/>
      <variable name="outbound_caller_id_name" value="My Company"/>
      <variable name="outbound_caller_id_number" value="15551234567"/>
      <variable name="callgroup" value="sales"/>
    </variables>
  </user>
</include>

#!/bin/bash
# create-extensions.sh
# Creates 10 SIP extensions (1001-1010) for FreeSWITCH

DIRECTORY="/etc/freeswitch/directory/default"

declare -A USERS=(
  [1001]="Alice Johnson:sales"
  [1002]="Bob Williams:sales"
  [1003]="Carol Davis:sales"
  [1004]="Dan Miller:support"
  [1005]="Eve Wilson:support"
  [1006]="Frank Brown:support"
  [1007]="Grace Taylor:billing"
  [1008]="Hank Anderson:billing"
  [1009]="Ivy Martinez:management"
  [1010]="Jack Thompson:management"
)

for EXT in "${!USERS[@]}"; do
  IFS=':' read -r NAME GROUP <<< "${USERS[$EXT]}"

  # Generate a random password
  PASS=$(openssl rand -base64 12 | tr -dc 'A-Za-z0-9' | head -c 16)

  cat > "${DIRECTORY}/${EXT}.xml" << XMLEOF
<include>
  <user id="${EXT}">
    <params>
      <param name="password" value="${PASS}"/>
      <param name="vm-password" value="${EXT}"/>
    </params>
    <variables>
      <variable name="toll_allow" value="domestic,local"/>
      <variable name="accountcode" value="${EXT}"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="${NAME}"/>
      <variable name="effective_caller_id_number" value="${EXT}"/>
      <variable name="outbound_caller_id_name" value="My Company"/>
      <variable name="outbound_caller_id_number" value="15551234567"/>
      <variable name="callgroup" value="${GROUP}"/>
    </variables>
  </user>
</include>
XMLEOF

  echo "Created ${EXT} — ${NAME} (${GROUP}) — Password: ${PASS}"
done

chown -R freeswitch:freeswitch "${DIRECTORY}"
echo ""
echo "Done. Run 'fs_cli -x reloadxml' to apply."
echo "IMPORTANT: Save the passwords above — they will not be displayed again."

chmod +x create-extensions.sh
./create-extensions.sh

<include>
  <gateway name="my_provider">
    <!-- Provider credentials -->
    <param name="username" value="your_sip_username"/>
    <param name="password" value="your_sip_password"/>
    <param name="realm" value="sip.provider.com"/>
    <param name="proxy" value="sip.provider.com"/>

    <!-- Registration (some providers require it, some use IP auth) -->
    <param name="register" value="true"/>
    <param name="register-transport" value="udp"/>

    <!-- Caller ID -->
    <param name="caller-id-in-from" value="true"/>

    <!-- Retry on failure -->
    <param name="retry-seconds" value="30"/>

    <!-- Codec preferences for this trunk -->
    <param name="codec-prefs" value="PCMU,PCMA,G722"/>

    <!-- Ping the provider to detect failures -->
    <param name="ping" value="25"/>
    <param name="ping-max" value="3"/>
    <param name="ping-min" value="1"/>
  </gateway>
</include>

<include>
  <gateway name="ip_auth_provider">
    <param name="username" value="not_used"/>
    <param name="password" value="not_used"/>
    <param name="realm" value="sip.provider.com"/>
    <param name="proxy" value="sip.provider.com"/>
    <param name="register" value="false"/>
    <param name="caller-id-in-from" value="true"/>
  </gateway>
</include>

# From fs_cli
fs_cli -x "sofia profile external rescan"

# Check gateway status
fs_cli -x "sofia status gateway my_provider"

# Expected output:
# Name         my_provider
# Profile      external
# Scheme       sip
# Realm        sip.provider.com
# Username     your_sip_username
# ...
# State        REGED  <-- Successfully registered
# Status       UP

<include>
  <context name="public">
    <!-- Route DID +15551234567 to extension 1001 -->
    <extension name="inbound_main">
      <condition field="destination_number" expression="^(\+?1?5551234567)$">
        <action application="set" data="domain_name=$${domain}"/>
        <action application="transfer" data="1001 XML default"/>
      </condition>
    </extension>

    <!-- Route DID +15559876543 to IVR -->
    <extension name="inbound_ivr">
      <condition field="destination_number" expression="^(\+?1?5559876543)$">
        <action application="set" data="domain_name=$${domain}"/>
        <action application="transfer" data="5000 XML default"/>
      </condition>
    </extension>

    <!-- Route DID +15555551234 to ring group (sales) -->
    <extension name="inbound_sales">
      <condition field="destination_number" expression="^(\+?1?5555551234)$">
        <action application="set" data="domain_name=$${domain}"/>
        <action application="transfer" data="9001 XML default"/>
      </condition>
    </extension>

    <!-- Catch-all: reject unknown DIDs -->
    <extension name="public_reject">
      <condition field="destination_number" expression="^(.*)$">
        <action application="log" data="WARNING Rejecting unknown inbound DID: ${destination_number} from ${sip_from_uri}"/>
        <action application="hangup" data="CALL_REJECTED"/>
      </condition>
    </extension>
  </context>
</include>

<X-PRE-PROCESS cmd="set" data="external_sip_ip=YOUR_PUBLIC_IP"/>
<X-PRE-PROCESS cmd="set" data="external_rtp_ip=YOUR_PUBLIC_IP"/>

<!-- Replace the sip-ip and rtp-ip lines with: -->
<param name="sip-ip" value="$${local_ip_v4}"/>
<param name="ext-sip-ip" value="$${external_sip_ip}"/>
<param name="rtp-ip" value="$${local_ip_v4}"/>
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>

<!-- Enable NAT handling -->
<param name="apply-nat-acl" value="nat.auto"/>
<param name="aggressive-nat-detection" value="true"/>
<param name="local-network-acl" value="localnet.auto"/>

fs_cli -x "sofia status profile internal reg"

# Output shows registered endpoints:
# Call-ID:        xxxxx@yyyy
# User:           1001@YOUR_SERVER_IP
# Contact:        "Alice Johnson" <sip:1001@phone_ip:port>
# Agent:          Ooh, a softphone!
# Status:         Registered(UDP)(unknown) EXP(2024-01-01 12:00:00)
# ...

<context name="default">          <!-- Container: who can use these rules -->
  <extension name="my_rule">      <!-- Named rule -->
    <condition field="X" expression="regex">  <!-- When to match -->
      <action application="Y" data="Z"/>      <!-- What to do -->
      <action application="Y2" data="Z2"/>    <!-- Actions run in sequence -->
    </condition>
  </extension>
</context>

^1001$              — Exact match for "1001"
^(10[0-1][0-9])$    — Match 1000-1019
^(\d{10})$          — Match any 10-digit number
^9(\d+)$            — Match 9 + any digits (capture digits after 9)
^(\+?1?\d{10})$     — Match with optional +1 prefix
^.*$                — Match anything (catch-all)

<!-- Set a variable -->
<action application="set" data="my_var=hello"/>

<!-- Use a variable in an action -->
<action application="log" data="INFO The value is ${my_var}"/>

<!-- Use a variable in a condition -->
<condition field="${my_var}" expression="^(hello)$">

<extension name="business_hours">
  <condition field="time_of_day" expression="09:00-17:30">
    <!-- This runs during business hours -->
    <action application="transfer" data="5000 XML default"/>
    <!-- This runs OUTSIDE business hours -->
    <anti-action application="playback" data="ivr/ivr-please_call_back_during_business_hours.wav"/>
    <anti-action application="voicemail" data="default $${domain} 1001"/>
  </condition>
</extension>

<include>
  <context name="default">

    <!-- ============================================ -->
    <!-- INTERNAL EXTENSION CALLING (1001-1099)       -->
    <!-- ============================================ -->
    <extension name="internal_extensions">
      <condition field="destination_number" expression="^(10[0-9]{2})$">
        <action application="set" data="dialed_extension=$1"/>
        <action application="export" data="dialed_extension=$1"/>

        <!-- Set ring timeout to 30 seconds -->
        <action application="set" data="call_timeout=30"/>

        <!-- Set caller ID -->
        <action application="set"
          data="effective_caller_id_name=${outbound_caller_id_name}"/>
        <action application="set"
          data="effective_caller_id_number=${outbound_caller_id_number}"/>

        <!-- Ring the extension, then send to voicemail on no answer -->
        <action application="set"
          data="hangup_after_bridge=true"/>
        <action application="set"
          data="continue_on_fail=true"/>
        <action application="bridge"
          data="user/${dialed_extension}@$${domain}"/>

        <!-- No answer — go to voicemail -->
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="voicemail"
          data="default $${domain} ${dialed_extension}"/>
      </condition>
    </extension>

    <!-- ============================================ -->
    <!-- RING GROUPS                                  -->
    <!-- ============================================ -->

    <!-- Sales ring group (9001) — ring all sales phones simultaneously -->
    <extension name="ring_group_sales">
      <condition field="destination_number" expression="^(9001)$">
        <action application="set" data="call_timeout=25"/>
        <action application="set" data="continue_on_fail=true"/>
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="bridge"
          data="user/1001@$${domain},user/1002@$${domain},user/1003@$${domain}"/>
        <!-- All busy/unavailable — voicemail for sales -->
        <action application="voicemail" data="default $${domain} 1001"/>
      </condition>
    </extension>

    <!-- Support ring group (9002) — sequential ring (try each for 15s) -->
    <extension name="ring_group_support">
      <condition field="destination_number" expression="^(9002)$">
        <action application="set" data="continue_on_fail=true"/>
        <action application="set" data="hangup_after_bridge=true"/>
        <!-- Try first support agent -->
        <action application="set" data="call_timeout=15"/>
        <action application="bridge" data="user/1004@$${domain}"/>
        <!-- Try second -->
        <action application="bridge" data="user/1005@$${domain}"/>
        <!-- Try third -->
        <action application="bridge" data="user/1006@$${domain}"/>
        <!-- All unavailable -->
        <action application="voicemail" data="default $${domain} 1004"/>
      </condition>
    </extension>

    <!-- ============================================ -->
    <!-- OUTBOUND CALLS VIA TRUNK                     -->
    <!-- ============================================ -->

    <!-- Local/National calls: dial 9 + number -->
    <extension name="outbound_national">
      <condition field="destination_number" expression="^9(\d{10,11})$">
        <action application="set" data="effective_caller_id_number=15551234567"/>
        <action application="set" data="effective_caller_id_name=My Company"/>
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set" data="call_timeout=60"/>
        <action application="bridge"
          data="sofia/gateway/my_provider/$1"/>
      </condition>
    </extension>

    <!-- International calls: dial 00 + country code + number -->
    <extension name="outbound_international">
      <condition field="destination_number" expression="^(00\d{7,15})$">
        <!-- Check if user has international permission -->
        <action application="set"
          data="continue_on_fail=false"/>
        <action application="set"
          data="effective_caller_id_number=15551234567"/>
        <action application="bridge"
          data="sofia/gateway/my_provider/$1"/>
      </condition>
    </extension>

    <!-- ============================================ -->
    <!-- TIME-BASED ROUTING                           -->
    <!-- ============================================ -->
    <extension name="main_number_time_routing">
      <condition field="destination_number" expression="^(5000)$"/>
      <!-- Check business hours: Mon-Fri 9:00-17:30 -->
      <condition field="time_of_day" expression="09:00-17:30">
        <action application="transfer" data="5001 XML default"/>
        <anti-action application="transfer" data="5002 XML default"/>
      </condition>
    </extension>

    <!-- Business hours handler -->
    <extension name="business_hours_handler">
      <condition field="destination_number" expression="^(5001)$">
        <action application="answer"/>
        <action application="playback"
          data="ivr/ivr-welcome.wav"/>
        <action application="transfer" data="main_ivr XML default"/>
      </condition>
    </extension>

    <!-- After hours handler -->
    <extension name="after_hours_handler">
      <condition field="destination_number" expression="^(5002)$">
        <action application="answer"/>
        <action application="playback"
          data="ivr/ivr-after_hours_message.wav"/>
        <action application="voicemail"
          data="default $${domain} 1001"/>
      </condition>
    </extension>

    <!-- ============================================ -->
    <!-- UTILITIES                                    -->
    <!-- ============================================ -->

    <!-- Echo test (dial 9196) -->
    <extension name="echo_test">
      <condition field="destination_number" expression="^(9196)$">
        <action application="answer"/>
        <action application="echo"/>
      </condition>
    </extension>

    <!-- Music on hold test (dial 9664) -->
    <extension name="moh_test">
      <condition field="destination_number" expression="^(9664)$">
        <action application="answer"/>
        <action application="playback"
          data="$${hold_music}"/>
      </condition>
    </extension>

    <!-- Voicemail access (dial *98) -->
    <extension name="voicemail_check">
      <condition field="destination_number" expression="^\*98$">
        <action application="answer"/>
        <action application="sleep" data="500"/>
        <action application="voicemail"
          data="check default $${domain}"/>
      </condition>
    </extension>

    <!-- Direct voicemail for extension (dial *99 + ext) -->
    <extension name="voicemail_direct">
      <condition field="destination_number" expression="^\*99(\d{4})$">
        <action application="answer"/>
        <action application="sleep" data="500"/>
        <action application="voicemail"
          data="default $${domain} $1"/>
      </condition>
    </extension>

    <!-- Call parking (dial 5900) -->
    <extension name="park_call">
      <condition field="destination_number" expression="^(5900)$">
        <action application="set" data="fifo_music=$${hold_music}"/>
        <action application="fifo" data="park@$${domain} in"/>
      </condition>
    </extension>

    <!-- Retrieve parked call (dial 5901) -->
    <extension name="retrieve_parked_call">
      <condition field="destination_number" expression="^(5901)$">
        <action application="answer"/>
        <action application="fifo" data="park@$${domain} out wait"/>
      </condition>
    </extension>

  </context>
</include>

<configuration name="ivr.conf" description="IVR Menus">
  <menus>

    <!-- ============================================ -->
    <!-- MAIN MENU (Level 1)                          -->
    <!-- ============================================ -->
    <menu name="main_ivr"
          greet-long="ivr/ivr-welcome_to_our_company.wav"
          greet-short="ivr/ivr-please_make_selection.wav"
          invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
          exit-sound="voicemail/vm-goodbye.wav"
          confirm-macro=""
          confirm-key=""
          tts-engine=""
          tts-voice=""
          confirm-attempts="3"
          timeout="10000"
          inter-digit-timeout="2000"
          max-failures="3"
          max-timeouts="3"
          digit-len="1">

      <!-- Press 1 for Sales -->
      <entry action="menu-exec-app"
             digits="1"
             param="transfer 9001 XML default"/>

      <!-- Press 2 for Support -->
      <entry action="menu-exec-app"
             digits="2"
             param="transfer 9002 XML default"/>

      <!-- Press 3 for Billing -->
      <entry action="menu-sub"
             digits="3"
             param="billing_ivr"/>

      <!-- Press 4 for Company Directory -->
      <entry action="menu-exec-app"
             digits="4"
             param="transfer 411 XML default"/>

      <!-- Press 0 for Operator -->
      <entry action="menu-exec-app"
             digits="0"
             param="transfer 1009 XML default"/>

      <!-- Press * to repeat -->
      <entry action="menu-top"
             digits="*"/>

    </menu>

    <!-- ============================================ -->
    <!-- BILLING SUBMENU (Level 2)                    -->
    <!-- ============================================ -->
    <menu name="billing_ivr"
          greet-long="ivr/ivr-billing_menu.wav"
          greet-short="ivr/ivr-please_make_selection.wav"
          invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
          exit-sound="voicemail/vm-goodbye.wav"
          timeout="10000"
          inter-digit-timeout="2000"
          max-failures="3"
          max-timeouts="3"
          digit-len="1">

      <!-- Press 1 for Account Balance -->
      <entry action="menu-exec-app"
             digits="1"
             param="transfer 8001 XML default"/>

      <!-- Press 2 for Payment -->
      <entry action="menu-exec-app"
             digits="2"
             param="transfer 8002 XML default"/>

      <!-- Press 3 for Billing Agent -->
      <entry action="menu-exec-app"
             digits="3"
             param="transfer 1007 XML default"/>

      <!-- Press 9 to go back to main menu -->
      <entry action="menu-back"
             digits="9"/>

      <!-- Press 0 for Operator -->
      <entry action="menu-exec-app"
             digits="0"
             param="transfer 1009 XML default"/>

    </menu>

    <!-- ============================================ -->
    <!-- SUPPORT SUBMENU (Level 2)                    -->
    <!-- ============================================ -->
    <menu name="support_ivr"
          greet-long="ivr/ivr-support_menu.wav"
          greet-short="ivr/ivr-please_make_selection.wav"
          invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
          exit-sound="voicemail/vm-goodbye.wav"
          timeout="10000"
          inter-digit-timeout="2000"
          max-failures="3"
          max-timeouts="3"
          digit-len="1">

      <!-- Press 1 for Technical Support -->
      <entry action="menu-sub"
             digits="1"
             param="tech_support_ivr"/>

      <!-- Press 2 for General Inquiries -->
      <entry action="menu-exec-app"
             digits="2"
             param="transfer 1005 XML default"/>

      <!-- Press 9 to go back -->
      <entry action="menu-back"
             digits="9"/>

    </menu>

    <!-- ============================================ -->
    <!-- TECH SUPPORT (Level 3)                       -->
    <!-- ============================================ -->
    <menu name="tech_support_ivr"
          greet-long="ivr/ivr-tech_support_menu.wav"
          greet-short="ivr/ivr-please_make_selection.wav"
          invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
          exit-sound="voicemail/vm-goodbye.wav"
          timeout="10000"
          inter-digit-timeout="2000"
          max-failures="3"
          max-timeouts="3"
          digit-len="1">

      <!-- Press 1 for Internet Issues -->
      <entry action="menu-exec-app"
             digits="1"
             param="transfer 1004 XML default"/>

      <!-- Press 2 for Phone/VoIP Issues -->
      <entry action="menu-exec-app"
             digits="2"
             param="transfer 1005 XML default"/>

      <!-- Press 3 for Email Issues -->
      <entry action="menu-exec-app"
             digits="3"
             param="transfer 1006 XML default"/>

      <!-- Press 9 to go back -->
      <entry action="menu-back"
             digits="9"/>

      <!-- Press 0 for Operator -->
      <entry action="menu-exec-app"
             digits="0"
             param="transfer 1009 XML default"/>

    </menu>

  </menus>
</configuration>

<!-- IVR entry point (dial 5000 or transfer from public context) -->
<extension name="main_ivr">
  <condition field="destination_number" expression="^(main_ivr|5000)$">
    <action application="answer"/>
    <action application="sleep" data="500"/>
    <action application="ivr" data="main_ivr"/>
  </condition>
</extension>

<!-- Collect a 5-digit account number -->
<extension name="account_lookup">
  <condition field="destination_number" expression="^(8001)$">
    <action application="answer"/>
    <action application="sleep" data="500"/>

    <!-- play_and_get_digits parameters:
         min_digits max_digits max_tries timeout terminators
         audio_file bad_input_file variable_name regex_pattern
         digit_timeout transfer_on_failure -->
    <action application="play_and_get_digits"
      data="5 5 3 10000 # ivr/ivr-enter_account_number.wav ivr/ivr-that_was_an_invalid_entry.wav account_number \d{5} 15000"/>

    <!-- Now use the collected digits -->
    <action application="log"
      data="INFO Account number entered: ${account_number}"/>

    <!-- You could use mod_xml_curl to look up the account,
         or use Lua/Python to query a database -->
    <action application="playback"
      data="ivr/ivr-thank_you.wav"/>
    <action application="transfer"
      data="1007 XML default"/>
  </condition>
</extension>

<action application="speak" data="flite|kal|Welcome to our company. Press 1 for sales."/>

<configuration name="tts_commandline.conf" description="TTS Command">
  <settings>
    <param name="command" value="echo '${text}' | /usr/bin/piper --model /opt/tts/en_US-lessac-medium.onnx --output_file ${file}"/>
  </settings>
</configuration>

<action application="speak"
  data="tts_commandline|default|Welcome to our company. Please hold while we connect you."/>

Caller dials main number
       │
       ▼
   Level 1: Main Menu
   1 → Sales (ring group 9001)
   2 → Support submenu
   3 → Billing submenu
   4 → Company directory
   0 → Operator (ext 1009)
       │
       ├── Level 2: Support
       │   1 → Tech Support submenu
       │   2 → General Inquiries (ext 1005)
       │   9 → Back to Main
       │       │
       │       └── Level 3: Tech Support
       │           1 → Internet (ext 1004)
       │           2 → VoIP (ext 1005)
       │           3 → Email (ext 1006)
       │           9 → Back to Support
       │           0 → Operator
       │
       └── Level 2: Billing
           1 → Account Balance (collect acct#)
           2 → Payment (ext 8002)
           3 → Billing Agent (ext 1007)
           9 → Back to Main
           0 → Operator

<configuration name="voicemail.conf" description="Voicemail">
  <settings>
  </settings>

  <profiles>
    <profile name="default">
      <!-- Storage -->
      <param name="file-extension" value="wav"/>
      <param name="record-silence-threshold" value="200"/>
      <param name="record-silence-hits" value="5"/>
      <param name="max-record-len" value="300"/>
      <param name="max-retries" value="3"/>

      <!-- Greeting -->
      <param name="terminator-key" value="#"/>
      <param name="play-new-messages-key" value="1"/>
      <param name="play-saved-messages-key" value="2"/>

      <!-- Message controls (during playback) -->
      <param name="skip-greet-key" value="#"/>
      <param name="config-menu-key" value="5"/>
      <param name="record-greeting-key" value="1"/>
      <param name="choose-greeting-key" value="2"/>
      <param name="change-pass-key" value="6"/>

      <!-- Playback controls -->
      <param name="listen-key" value="1"/>
      <param name="save-key" value="2"/>
      <param name="delete-key" value="7"/>
      <param name="forward-key" value="8"/>
      <param name="repeat-key" value="0"/>

      <!-- Notification -->
      <param name="notify-mailto" value=""/>
      <param name="notify-email-body" value="You have a new voicemail from ${caller_id_number} (${caller_id_name}). The message is ${message_len} seconds long."/>
      <param name="notify-email-subject" value="New voicemail from ${caller_id_number}"/>

      <!-- Email with attachment -->
      <param name="email-from" value="voicemail@YOUR_DOMAIN"/>
      <param name="email-body" value="You have a new voicemail.\n\nFrom: ${caller_id_name} (${caller_id_number})\nDate: ${left_epoch}\nDuration: ${message_len} seconds\n\nThe recording is attached."/>
      <param name="email-subject" value="[Voicemail] New message from ${caller_id_number}"/>

      <!-- Attach the recording to the email -->
      <param name="vm-email-all-messages" value="true"/>
      <!-- Delete from server after emailing (set false to keep) -->
      <param name="vm-delete-file" value="false"/>
      <!-- Keep the message as new after emailing -->
      <param name="vm-keep-local-after-email" value="true"/>

      <!-- MWI (Message Waiting Indicator — lights up the phone's voicemail LED) -->
      <param name="vm-message-ext" value="wav"/>

      <!-- Storage location -->
      <param name="storage-dir" value="$${storage_dir}/voicemail/default"/>

      <!-- Operator extension (press 0 during greeting) -->
      <param name="operator-extension" value="operator XML default"/>
      <param name="operator-key" value="0"/>
    </profile>
  </profiles>
</configuration>

<!-- Send to voicemail (internal — after no answer on bridge) -->
<!-- This is already in the internal_extensions example above -->

<!-- Check own voicemail: dial *98 -->
<extension name="voicemail_check">
  <condition field="destination_number" expression="^\*98$">
    <action application="answer"/>
    <action application="sleep" data="500"/>
    <action application="voicemail"
      data="check default $${domain}"/>
  </condition>
</extension>

<!-- Leave voicemail directly for someone: dial *99 + extension -->
<extension name="voicemail_leave">
  <condition field="destination_number" expression="^\*99(\d{4})$">
    <action application="answer"/>
    <action application="sleep" data="500"/>
    <action application="voicemail"
      data="default $${domain} $1"/>
  </condition>
</extension>

<!-- Check specific mailbox (for shared mailboxes): dial *97 + extension -->
<extension name="voicemail_check_specific">
  <condition field="destination_number" expression="^\*97(\d{4})$">
    <action application="answer"/>
    <action application="sleep" data="500"/>
    <action application="voicemail"
      data="check default $${domain} $1"/>
  </condition>
</extension>

# Install a lightweight MTA
apt-get install -y msmtp msmtp-mta

# Configure SMTP relay
cat > /etc/msmtprc << 'EOF'
defaults
auth           on
tls            on
tls_trust_file /etc/ssl/certs/ca-certificates.crt
logfile        /var/log/msmtp.log

account        default
host           smtp.gmail.com
port           587
from           voicemail@YOUR_DOMAIN
user           your_email@gmail.com
password       your_app_password
EOF

chmod 600 /etc/msmtprc

<!-- In /etc/freeswitch/directory/default/1001.xml -->
<include>
  <user id="1001">
    <params>
      <param name="password" value="Str0ng_P@ss_1001!"/>
      <param name="vm-password" value="1001"/>
      <param name="vm-mailto" value="alice@company.com"/>
      <param name="vm-email-all-messages" value="true"/>
      <param name="vm-attach-file" value="true"/>
      <param name="vm-keep-local-after-email" value="true"/>
    </params>
    <!-- ... variables ... -->
  </user>
</include>

#!/bin/bash
# /usr/local/bin/cleanup-voicemail.sh
# Delete voicemail messages older than 30 days

STORAGE_DIR="/var/lib/freeswitch/storage/voicemail"
MAX_AGE=30

echo "$(date '+%Y-%m-%d %H:%M:%S') — Voicemail cleanup starting"

# Find and delete old voicemail recordings
find "${STORAGE_DIR}" -name "*.wav" -type f -mtime +${MAX_AGE} -delete
find "${STORAGE_DIR}" -name "*.mp3" -type f -mtime +${MAX_AGE} -delete

# Clean up empty directories
find "${STORAGE_DIR}" -type d -empty -delete

echo "$(date '+%Y-%m-%d %H:%M:%S') — Voicemail cleanup complete"

chmod +x /usr/local/bin/cleanup-voicemail.sh

# Run weekly
echo "0 3 * * 0 root /usr/local/bin/cleanup-voicemail.sh >> /var/log/voicemail-cleanup.log 2>&1" \
  > /etc/cron.d/voicemail-cleanup

<extension name="recorded_internal_call">
  <condition field="destination_number" expression="^(10[0-9]{2})$">
    <action application="set" data="dialed_extension=$1"/>

    <!-- Start recording BEFORE bridging -->
    <action application="set"
      data="RECORD_STEREO=true"/>
    <action application="set"
      data="media_bug_answer_req=true"/>
    <action application="record_session"
      data="/var/lib/freeswitch/recordings/${strftime(%Y/%m/%d)}/${uuid}_${caller_id_number}_to_${dialed_extension}.wav"/>

    <!-- Now bridge the call -->
    <action application="set" data="call_timeout=30"/>
    <action application="set" data="hangup_after_bridge=true"/>
    <action application="set" data="continue_on_fail=true"/>
    <action application="bridge"
      data="user/${dialed_extension}@$${domain}"/>

    <!-- Voicemail on no answer (recording continues into VM) -->
    <action application="answer"/>
    <action application="voicemail"
      data="default $${domain} ${dialed_extension}"/>
  </condition>
</extension>

<!-- Date-based subdirectories + descriptive filenames -->
<action application="record_session"
  data="/var/lib/freeswitch/recordings/${strftime(%Y/%m/%d)}/${uuid}_${caller_id_number}_to_${destination_number}_${strftime(%Y%m%d-%H%M%S)}.wav"/>

/var/lib/freeswitch/recordings/2026/03/14/abc123_15551234567_to_1001_20260314-143022.wav

<!-- Enable stereo recording -->
<action application="set" data="RECORD_STEREO=true"/>

<!-- Record in stereo WAV -->
<action application="record_session"
  data="/var/lib/freeswitch/recordings/${strftime(%Y/%m/%d)}/${uuid}.wav"/>

# Start recording on an active call (from fs_cli or ESL)
uuid_record <call-uuid> start /var/lib/freeswitch/recordings/mid_call_recording.wav

# Stop recording
uuid_record <call-uuid> stop /var/lib/freeswitch/recordings/mid_call_recording.wav

# Stop all recordings on a call
uuid_record <call-uuid> stop all

<!-- Agent presses *1 to start recording, *2 to stop -->
<extension name="record_on_demand">
  <condition field="destination_number" expression="^(10[0-9]{2})$">
    <action application="set" data="dialed_extension=$1"/>

    <!-- Bind DTMF keys for recording control -->
    <action application="bind_digit_action"
      data="rec,*1,exec:record_session,/var/lib/freeswitch/recordings/${uuid}_on_demand.wav"/>
    <action application="bind_digit_action"
      data="rec,*2,exec:stop_record_session,/var/lib/freeswitch/recordings/${uuid}_on_demand.wav"/>
    <action application="digit_action_set_realm" data="rec"/>

    <action application="bridge"
      data="user/${dialed_extension}@$${domain}"/>
  </condition>
</extension>

#!/bin/bash
# /usr/local/bin/convert-recordings.sh
# Convert WAV recordings to MP3 and archive

RECORDINGS_DIR="/var/lib/freeswitch/recordings"
ARCHIVE_DIR="/var/lib/freeswitch/recordings-mp3"
LOG="/var/log/recording-convert.log"

# Install lame if not present
which lame > /dev/null 2>&1 || apt-get install -y lame

# Process WAV files older than 5 minutes (avoid in-progress recordings)
find "${RECORDINGS_DIR}" -name "*.wav" -type f -mmin +5 | while read WAV_FILE; do
  # Build MP3 path (mirror directory structure)
  REL_PATH="${WAV_FILE#${RECORDINGS_DIR}/}"
  MP3_FILE="${ARCHIVE_DIR}/${REL_PATH%.wav}.mp3"
  MP3_DIR=$(dirname "${MP3_FILE}")

  # Create directory
  mkdir -p "${MP3_DIR}"

  # Convert
  if lame --quiet -V2 "${WAV_FILE}" "${MP3_FILE}" 2>/dev/null; then
    echo "$(date '+%Y-%m-%d %H:%M:%S') Converted: ${REL_PATH}" >> "${LOG}"
    # Delete original WAV after successful conversion
    rm -f "${WAV_FILE}"
  else
    echo "$(date '+%Y-%m-%d %H:%M:%S') FAILED: ${REL_PATH}" >> "${LOG}"
  fi
done

# Clean up empty directories
find "${RECORDINGS_DIR}" -type d -empty -delete 2>/dev/null

chmod +x /usr/local/bin/convert-recordings.sh

# Run every 30 minutes
echo "*/30 * * * * root /usr/local/bin/convert-recordings.sh" \
  > /etc/cron.d/recording-convert

#!/bin/bash
# /usr/local/bin/cleanup-recordings.sh
# Delete recordings older than retention period

RECORDINGS_DIR="/var/lib/freeswitch/recordings"
MP3_DIR="/var/lib/freeswitch/recordings-mp3"
RETENTION_DAYS=90

echo "$(date '+%Y-%m-%d %H:%M:%S') — Recording cleanup starting"
echo "Retention: ${RETENTION_DAYS} days"

# Delete old WAV files
WAV_COUNT=$(find "${RECORDINGS_DIR}" -name "*.wav" -type f -mtime +${RETENTION_DAYS} | wc -l)
find "${RECORDINGS_DIR}" -name "*.wav" -type f -mtime +${RETENTION_DAYS} -delete
echo "Deleted ${WAV_COUNT} WAV files older than ${RETENTION_DAYS} days"

# Delete old MP3 files
MP3_COUNT=$(find "${MP3_DIR}" -name "*.mp3" -type f -mtime +${RETENTION_DAYS} | wc -l)
find "${MP3_DIR}" -name "*.mp3" -type f -mtime +${RETENTION_DAYS} -delete
echo "Deleted ${MP3_COUNT} MP3 files older than ${RETENTION_DAYS} days"

# Clean empty directories
find "${RECORDINGS_DIR}" -type d -empty -delete 2>/dev/null
find "${MP3_DIR}" -type d -empty -delete 2>/dev/null

# Report disk usage
echo "Current recording storage:"
du -sh "${RECORDINGS_DIR}" "${MP3_DIR}" 2>/dev/null

echo "$(date '+%Y-%m-%d %H:%M:%S') — Cleanup complete"

chmod +x /usr/local/bin/cleanup-recordings.sh

# Run daily at 4 AM
echo "0 4 * * * root /usr/local/bin/cleanup-recordings.sh >> /var/log/recording-cleanup.log 2>&1" \
  > /etc/cron.d/recording-cleanup

<configuration name="conference.conf" description="Audio Conference">
  <advertise>
    <!-- Advertise conference rooms via SIP SUBSCRIBE -->
    <room name="3001@$${domain}" status="FreeSWITCH"/>
  </advertise>

  <caller-controls>
    <!-- Default key bindings for participants -->
    <group name="default">
      <control action="mute" digits="0"/>
      <control action="deaf mute" digits="*"/>
      <control action="energy up" digits="9"/>
      <control action="energy equ" digits="8"/>
      <control action="energy dn" digits="7"/>
      <control action="vol talk up" digits="3"/>
      <control action="vol talk zero" digits="2"/>
      <control action="vol talk dn" digits="1"/>
      <control action="vol listen up" digits="6"/>
      <control action="vol listen zero" digits="5"/>
      <control action="vol listen dn" digits="4"/>
      <control action="hangup" digits="#"/>
    </group>

    <!-- Moderator key bindings -->
    <group name="moderator">
      <control action="mute" digits="0"/>
      <control action="deaf mute" digits="*"/>
      <control action="hangup" digits="#"/>
      <control action="lock" digits="*1"/>
      <control action="mute non_moderator" digits="*5"/>
      <control action="unmute non_moderator" digits="*6"/>
      <control action="kick last" digits="*7"/>
      <control action="transfer" digits="*9" data="1009 XML default"/>
    </group>
  </caller-controls>

  <profiles>
    <!-- Standard conference profile -->
    <profile name="default">
      <param name="domain" value="$${domain}"/>
      <param name="rate" value="16000"/>
      <param name="interval" value="20"/>
      <param name="energy-level" value="100"/>

      <!-- Comfort noise for silence -->
      <param name="comfort-noise" value="true"/>

      <!-- Sounds -->
      <param name="muted-sound" value="conference/conf-muted.wav"/>
      <param name="unmuted-sound" value="conference/conf-unmuted.wav"/>
      <param name="alone-sound" value="conference/conf-alone.wav"/>
      <param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/>
      <param name="exit-sound" value="tone_stream://%(500,0,300,200,100,50,25)"/>
      <param name="kicked-sound" value="conference/conf-kicked.wav"/>
      <param name="locked-sound" value="conference/conf-locked.wav"/>
      <param name="is-locked-sound" value="conference/conf-is-locked.wav"/>
      <param name="is-unlocked-sound" value="conference/conf-is-unlocked.wav"/>
      <param name="pin-sound" value="conference/conf-pin.wav"/>
      <param name="bad-pin-sound" value="conference/conf-bad-pin.wav"/>

      <!-- Controls -->
      <param name="caller-controls" value="default"/>
      <param name="moderator-controls" value="moderator"/>

      <!-- Auto-record all conferences -->
      <!-- <param name="auto-record"
        value="/var/lib/freeswitch/recordings/conference/${conference_name}_${strftime(%Y%m%d-%H%M%S)}.wav"/> -->

      <!-- Max members (0 = unlimited) -->
      <param name="max-members" value="100"/>
      <!-- Announce count of members when joining -->
      <param name="announce-count" value="5"/>

      <!-- Codec preferences -->
      <param name="conference-flags"
        value="wait-mod|audio-always|waste-bandwidth"/>
    </profile>

    <!-- PIN-protected conference profile -->
    <profile name="secure">
      <param name="domain" value="$${domain}"/>
      <param name="rate" value="16000"/>
      <param name="interval" value="20"/>
      <param name="energy-level" value="100"/>
      <param name="comfort-noise" value="true"/>

      <!-- PIN required -->
      <param name="pin" value="12345"/>
      <param name="pin-retries" value="3"/>
      <param name="pin-sound" value="conference/conf-pin.wav"/>
      <param name="bad-pin-sound" value="conference/conf-bad-pin.wav"/>

      <param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/>
      <param name="exit-sound" value="tone_stream://%(500,0,300,200,100,50,25)"/>
      <param name="caller-controls" value="default"/>
      <param name="moderator-controls" value="moderator"/>
      <param name="max-members" value="50"/>
      <param name="conference-flags" value="wait-mod|audio-always"/>
    </profile>

    <!-- Webinar profile: listeners muted by default -->
    <profile name="webinar">
      <param name="domain" value="$${domain}"/>
      <param name="rate" value="16000"/>
      <param name="interval" value="20"/>
      <param name="energy-level" value="100"/>
      <param name="comfort-noise" value="true"/>

      <param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/>
      <param name="exit-sound" value="tone_stream://%(500,0,300,200,100,50,25)"/>

      <param name="caller-controls" value="default"/>
      <param name="moderator-controls" value="moderator"/>
      <param name="max-members" value="500"/>

      <!-- Members join muted -->
      <param name="member-flags" value="mute"/>
      <param name="conference-flags" value="wait-mod|audio-always"/>

      <!-- Auto-record webinars -->
      <param name="auto-record"
        value="/var/lib/freeswitch/recordings/webinar/${conference_name}_${strftime(%Y%m%d-%H%M%S)}.wav"/>
    </profile>
  </profiles>
</configuration>

<!-- ============================================ -->
<!-- CONFERENCE ROOMS                             -->
<!-- ============================================ -->

<!-- Standard conference rooms: dial 3001-3099 -->
<extension name="conference_standard">
  <condition field="destination_number" expression="^(30[0-9]{2})$">
    <action application="answer"/>
    <action application="sleep" data="500"/>
    <action application="conference" data="room_$1@default"/>
  </condition>
</extension>

<!-- PIN-protected conference rooms: dial 3100-3199 -->
<extension name="conference_secure">
  <condition field="destination_number" expression="^(31[0-9]{2})$">
    <action application="answer"/>
    <action application="sleep" data="500"/>
    <action application="conference" data="room_$1@secure"/>
  </condition>
</extension>

<!-- Moderator entry: dial 3200 + room number (e.g., 32003001 for room 3001) -->
<extension name="conference_moderator">
  <condition field="destination_number" expression="^3200(30[0-9]{2})$">
    <action application="answer"/>
    <action application="sleep" data="500"/>
    <action application="set" data="conference_member_flags=moderator"/>
    <action application="conference" data="room_$1@default+flags{moderator}"/>
  </condition>
</extension>

<!-- Webinar rooms: dial 3300-3399 (listeners muted) -->
<extension name="conference_webinar">
  <condition field="destination_number" expression="^(33[0-9]{2})$">
    <action application="answer"/>
    <action application="sleep" data="500"/>
    <action application="conference" data="webinar_$1@webinar"/>
  </condition>
</extension>

<!-- Webinar presenter: dial 3400 + room number -->
<extension name="conference_webinar_presenter">
  <condition field="destination_number" expression="^3400(33[0-9]{2})$">
    <action application="answer"/>
    <action application="sleep" data="500"/>
    <action application="conference" data="webinar_$1@webinar+flags{moderator}"/>
  </condition>
</extension>

<extension name="conference_dynamic_pin">
  <condition field="destination_number" expression="^(35[0-9]{2})$">
    <action application="answer"/>
    <action application="sleep" data="500"/>
    <!-- Collect PIN from caller -->
    <action application="play_and_get_digits"
      data="4 6 3 10000 # conference/conf-pin.wav conference/conf-bad-pin.wav entered_pin \d{4,6} 10000"/>
    <!-- Verify PIN (in production, check against a database) -->
    <action application="set" data="expected_pin=7890"/>
    <action application="execute_extension"
      data="check_pin_${entered_pin} XML features"/>
    <!-- If we get here, PIN was correct -->
    <action application="conference" data="room_$1@default"/>
  </condition>
</extension>

# List active conferences
fs_cli -x "conference list"

# List members in a specific conference
fs_cli -x "conference room_3001 list"

# Mute a participant (by member ID)
fs_cli -x "conference room_3001 mute 3"

# Unmute a participant
fs_cli -x "conference room_3001 unmute 3"

# Kick a participant
fs_cli -x "conference room_3001 kick 3"

# Lock the conference (no new members)
fs_cli -x "conference room_3001 lock"

# Unlock
fs_cli -x "conference room_3001 unlock"

# Mute all non-moderators
fs_cli -x "conference room_3001 mute non_moderator"

# Start recording a conference
fs_cli -x "conference room_3001 record /var/lib/freeswitch/recordings/conference/room_3001_$(date +%Y%m%d).wav"

# Stop recording
fs_cli -x "conference room_3001 norecord all"

# Play a file into the conference
fs_cli -x "conference room_3001 play /var/lib/freeswitch/sounds/announcement.wav"

# Get conference count
fs_cli -x "conference room_3001 count"

<configuration name="event_socket.conf" description="Socket Client">
  <settings>
    <!-- Listen on localhost only (secure) -->
    <param name="listen-ip" value="127.0.0.1"/>
    <param name="listen-port" value="8021"/>
    <param name="password" value="YOUR_SECURE_ESL_PASSWORD"/>

    <!-- Optional: allow connections from specific IPs -->
    <!--<param name="listen-ip" value="0.0.0.0"/>-->
    <!--<param name="apply-inbound-acl" value="loopback.auto"/>-->
  </settings>
</configuration>

# Connect to local FreeSWITCH
fs_cli

# Connect to remote FreeSWITCH
fs_cli -H 10.0.0.5 -P 8021 -p YOUR_SECURE_ESL_PASSWORD

# Execute a single command and exit
fs_cli -x "sofia status"

# Execute API command
fs_cli -x "show channels"

# Execute background API command
fs_cli -x "bgapi originate user/1001 &echo()"

pip3 install greenswitch

#!/usr/bin/env python3
"""
freeswitch_monitor.py
Monitor FreeSWITCH events via ESL (inbound mode).
Logs call activity and tracks concurrent calls.
"""

import greenswitch
import json
from datetime import datetime


class FreeSWITCHMonitor:
    def __init__(self, host='127.0.0.1', port=8021, password='YOUR_SECURE_ESL_PASSWORD'):
        self.host = host
        self.port = port
        self.password = password
        self.active_calls = {}
        self.conn = None

    def connect(self):
        """Establish ESL connection."""
        self.conn = greenswitch.InboundESL(
            host=self.host,
            port=self.port,
            password=self.password
        )
        self.conn.connect()
        print(f"[{self._now()}] Connected to FreeSWITCH at {self.host}:{self.port}")

    def subscribe_events(self):
        """Subscribe to call-related events."""
        # Subscribe to specific events
        self.conn.send(
            'event plain CHANNEL_CREATE CHANNEL_ANSWER '
            'CHANNEL_HANGUP_COMPLETE DTMF CODEC'
        )

        # Register event handlers
        self.conn.register_handle('CHANNEL_CREATE', self._on_channel_create)
        self.conn.register_handle('CHANNEL_ANSWER', self._on_channel_answer)
        self.conn.register_handle('CHANNEL_HANGUP_COMPLETE', self._on_channel_hangup)
        self.conn.register_handle('DTMF', self._on_dtmf)

    def _on_channel_create(self, event):
        """Called when a new channel is created."""
        uuid = event.headers.get('Unique-ID', 'unknown')
        caller = event.headers.get('Caller-Caller-ID-Number', 'unknown')
        dest = event.headers.get('Caller-Destination-Number', 'unknown')
        direction = event.headers.get('Call-Direction', 'unknown')

        self.active_calls[uuid] = {
            'caller': caller,
            'destination': dest,
            'direction': direction,
            'created': datetime.now(),
            'state': 'ringing'
        }
        print(f"[{self._now()}] NEW CALL: {caller} -> {dest} ({direction}) UUID={uuid}")
        print(f"  Active calls: {len(self.active_calls)}")

    def _on_channel_answer(self, event):
        """Called when a channel is answered."""
        uuid = event.headers.get('Unique-ID', 'unknown')
        if uuid in self.active_calls:
            self.active_calls[uuid]['state'] = 'answered'
            caller = self.active_calls[uuid]['caller']
            dest = self.active_calls[uuid]['destination']
            print(f"[{self._now()}] ANSWERED: {caller} -> {dest} UUID={uuid}")

    def _on_channel_hangup(self, event):
        """Called when a channel hangs up."""
        uuid = event.headers.get('Unique-ID', 'unknown')
        cause = event.headers.get('Hangup-Cause', 'unknown')
        duration = event.headers.get('variable_billsec', '0')

        if uuid in self.active_calls:
            call = self.active_calls.pop(uuid)
            print(
                f"[{self._now()}] HANGUP: {call['caller']} -> {call['destination']} "
                f"Duration={duration}s Cause={cause} UUID={uuid}"
            )
            print(f"  Active calls: {len(self.active_calls)}")

    def _on_dtmf(self, event):
        """Called when DTMF is received."""
        uuid = event.headers.get('Unique-ID', 'unknown')
        digit = event.headers.get('DTMF-Digit', '?')
        print(f"[{self._now()}] DTMF: digit={digit} UUID={uuid}")

    def run_api(self, command):
        """Execute an API command and return the result."""
        result = self.conn.send(f'api {command}')
        return result.data if result else None

    def originate_call(self, endpoint, app='&park()'):
        """Place a new call."""
        cmd = f'api originate {endpoint} {app}'
        result = self.conn.send(cmd)
        print(f"[{self._now()}] Originate result: {result.data if result else 'failed'}")
        return result

    def _now(self):
        return datetime.now().strftime('%Y-%m-%d %H:%M:%S')

    def run(self):
        """Main event loop."""
        self.connect()
        self.subscribe_events()
        print(f"[{self._now()}] Listening for events... (Ctrl+C to stop)")

        # Get initial status
        status = self.run_api('status')
        if status:
            print(f"\n--- FreeSWITCH Status ---\n{status}\n")

        try:
            # This blocks and processes events
            self.conn.process_events()
        except KeyboardInterrupt:
            print(f"\n[{self._now()}] Shutting down...")


if __name__ == '__main__':
    monitor = FreeSWITCHMonitor(
        host='127.0.0.1',
        port=8021,
        password='YOUR_SECURE_ESL_PASSWORD'
    )
    monitor.run()

python3 freeswitch_monitor.py

#!/usr/bin/env python3
"""
call_queue.py
Simple call queue: callers hear hold music, agents dial in to take the next call.
Uses ESL inbound mode.
"""

import greenswitch
from collections import deque
from datetime import datetime
import threading


class CallQueue:
    def __init__(self, host='127.0.0.1', port=8021, password='YOUR_SECURE_ESL_PASSWORD'):
        self.conn = greenswitch.InboundESL(host=host, port=port, password=password)
        self.waiting_callers = deque()  # Queue of caller UUIDs
        self.available_agents = []       # List of agent UUIDs
        self.lock = threading.Lock()

    def connect(self):
        self.conn.connect()
        self.conn.send('event plain CHANNEL_ANSWER CHANNEL_HANGUP_COMPLETE')
        print(f"[{self._now()}] Queue system connected")

    def add_caller(self, uuid):
        """Add a caller to the queue and play hold music."""
        with self.lock:
            self.waiting_callers.append(uuid)
            # Play music on hold to the caller
            self.conn.send(
                f'api uuid_broadcast {uuid} '
                f'local_stream://moh both'
            )
            print(f"[{self._now()}] Caller {uuid} added to queue. "
                  f"Queue depth: {len(self.waiting_callers)}")
            self._try_connect()

    def add_agent(self, uuid):
        """Register an agent as available."""
        with self.lock:
            self.available_agents.append(uuid)
            print(f"[{self._now()}] Agent {uuid} available. "
                  f"Agents: {len(self.available_agents)}")
            self._try_connect()

    def _try_connect(self):
        """Try to bridge a waiting caller with an available agent."""
        if self.waiting_callers and self.available_agents:
            caller_uuid = self.waiting_callers.popleft()
            agent_uuid = self.available_agents.pop(0)

            # Bridge the two calls
            self.conn.send(
                f'api uuid_bridge {caller_uuid} {agent_uuid}'
            )
            print(
                f"[{self._now()}] CONNECTED: caller={caller_uuid} "
                f"agent={agent_uuid}"
            )

    def get_stats(self):
        """Return current queue statistics."""
        with self.lock:
            return {
                'waiting_callers': len(self.waiting_callers),
                'available_agents': len(self.available_agents),
                'timestamp': self._now()
            }

    def _now(self):
        return datetime.now().strftime('%Y-%m-%d %H:%M:%S')


if __name__ == '__main__':
    queue = CallQueue()
    queue.connect()
    # In production, callers and agents would be added via
    # dialplan outbound ESL or API calls
    print("Queue system running. Use ESL commands to add callers/agents.")

npm install modesl

// freeswitch_monitor.js
// FreeSWITCH event monitor using Node.js + modesl

const esl = require('modesl');

const connection = new esl.Connection('127.0.0.1', 8021, 'YOUR_SECURE_ESL_PASSWORD', () => {
    console.log(`[${timestamp()}] Connected to FreeSWITCH via ESL`);

    // Get initial status
    connection.api('status', (result) => {
        console.log(`\n--- FreeSWITCH Status ---\n${result.getBody()}\n`);
    });

    // Subscribe to events
    connection.subscribe([
        'CHANNEL_CREATE',
        'CHANNEL_ANSWER',
        'CHANNEL_HANGUP_COMPLETE'
    ], () => {
        console.log(`[${timestamp()}] Subscribed to call events`);
    });
});

// Track active calls
const activeCalls = new Map();

connection.on('esl::event::CHANNEL_CREATE::*', (event) => {
    const uuid = event.getHeader('Unique-ID');
    const caller = event.getHeader('Caller-Caller-ID-Number') || 'unknown';
    const dest = event.getHeader('Caller-Destination-Number') || 'unknown';
    const direction = event.getHeader('Call-Direction') || 'unknown';

    activeCalls.set(uuid, { caller, dest, direction, created: new Date() });
    console.log(`[${timestamp()}] NEW: ${caller} -> ${dest} (${direction}) [${activeCalls.size} active]`);
});

connection.on('esl::event::CHANNEL_ANSWER::*', (event) => {
    const uuid = event.getHeader('Unique-ID');
    const call = activeCalls.get(uuid);
    if (call) {
        console.log(`[${timestamp()}] ANSWER: ${call.caller} -> ${call.dest}`);
    }
});

connection.on('esl::event::CHANNEL_HANGUP_COMPLETE::*', (event) => {
    const uuid = event.getHeader('Unique-ID');
    const cause = event.getHeader('Hangup-Cause') || 'unknown';
    const duration = event.getHeader('variable_billsec') || '0';
    const call = activeCalls.get(uuid);

    if (call) {
        activeCalls.delete(uuid);
        console.log(
            `[${timestamp()}] HANGUP: ${call.caller} -> ${call.dest} ` +
            `Duration=${duration}s Cause=${cause} [${activeCalls.size} active]`
        );
    }
});

connection.on('error', (error) => {
    console.error(`[${timestamp()}] ESL Error:`, error);
});

function timestamp() {
    return new Date().toISOString().replace('T', ' ').substring(0, 19);
}

// Originate a call example (uncomment to use):
// connection.api('originate user/1001 &echo()', (result) => {
//     console.log('Originate result:', result.getBody());
// });

node freeswitch_monitor.js

<extension name="esl_controlled_ivr">
  <condition field="destination_number" expression="^(7000)$">
    <action application="socket" data="127.0.0.1:9090 async full"/>
  </condition>
</extension>

#!/usr/bin/env python3
"""
outbound_ivr.py
Simple outbound ESL IVR server.
FreeSWITCH connects to this app for call handling.
"""

import socket
import threading


def handle_call(client_socket, addr):
    """Handle one incoming ESL connection (one call)."""
    print(f"New call connection from {addr}")

    # Read the initial connect message
    data = client_socket.recv(65536).decode()

    # Send connect command
    client_socket.sendall(b'connect\n\n')
    data = client_socket.recv(65536).decode()

    # Parse caller info from headers
    headers = {}
    for line in data.split('\n'):
        if ':' in line:
            key, value = line.split(':', 1)
            headers[key.strip()] = value.strip()

    caller = headers.get('Caller-Caller-ID-Number', 'unknown')
    dest = headers.get('Caller-Destination-Number', 'unknown')
    uuid = headers.get('Unique-ID', 'unknown')

    print(f"Call from {caller} to {dest} (UUID: {uuid})")

    # Answer the call
    send_command(client_socket, 'answer')

    # Play a greeting
    send_command(client_socket, 'playback /var/lib/freeswitch/sounds/ivr/ivr-welcome.wav')

    # Collect digits
    send_command(
        client_socket,
        'play_and_get_digits 1 1 3 10000 # '
        '/var/lib/freeswitch/sounds/ivr/ivr-please_make_selection.wav '
        '/var/lib/freeswitch/sounds/ivr/ivr-that_was_an_invalid_entry.wav '
        'selection \\d 10000'
    )

    # Read the response to get the collected digit
    # (In production, parse CHANNEL_EXECUTE_COMPLETE events)

    # Transfer based on selection
    send_command(client_socket, 'transfer 1001 XML default')

    client_socket.close()


def send_command(sock, command):
    """Send an ESL command."""
    msg = f'sendmsg\ncall-command: execute\nexecute-app-name: {command}\n\n'
    sock.sendall(msg.encode())
    # Read response
    try:
        return sock.recv(65536).decode()
    except Exception:
        return ''


def main():
    server = socket.socket(socket.AF_INET, socket.SOCK_STREAM)
    server.setsockopt(socket.SOL_SOCKET, socket.SO_REUSEADDR, 1)
    server.bind(('127.0.0.1', 9090))
    server.listen(50)
    print("Outbound ESL server listening on 127.0.0.1:9090")

    while True:
        client, addr = server.accept()
        thread = threading.Thread(target=handle_call, args=(client, addr))
        thread.daemon = True
        thread.start()


if __name__ == '__main__':
    main()

<configuration name="cdr_csv.conf" description="CDR CSV Format">
  <settings>
    <!-- Master CSV file for all calls -->
    <param name="default-template" value="default"/>

    <!-- Rotate log file when it reaches this size (bytes) -->
    <param name="rotate-on-hup" value="true"/>

    <!-- Legs: a-leg only, b-leg only, or both -->
    <param name="legs" value="a"/>
  </settings>

  <templates>
    <!-- Default template — one line per call -->
    <template name="default">"${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${accountcode}","${read_codec}","${write_codec}","${sip_hangup_disposition}","${sip_from_uri}","${sip_to_uri}"</template>

    <!-- Detailed template with more fields -->
    <template name="detailed">"${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${progress_stamp}","${progress_media_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}","${accountcode}","${read_codec}","${write_codec}","${sip_hangup_disposition}","${endpoint_disposition}","${sip_from_uri}","${sip_to_uri}","${sip_call_id}","${network_addr}","${bridge_channel}","${last_app}","${last_arg}"</template>
  </templates>
</configuration>

<configuration name="cdr_sqlite.conf" description="CDR SQLite">
  <settings>
    <param name="db-name" value="cdr"/>
    <param name="db-table" value="cdr"/>
    <param name="legs" value="a"/>
    <param name="default-template" value="default"/>
  </settings>

  <templates>
    <template name="default">
      INSERT INTO cdr (
        caller_id_name, caller_id_number, destination_number, context,
        start_stamp, answer_stamp, end_stamp, duration, billsec,
        hangup_cause, uuid, accountcode, read_codec, write_codec,
        sip_hangup_disposition, network_addr
      ) VALUES (
        '${caller_id_name}', '${caller_id_number}', '${destination_number}',
        '${context}', '${start_stamp}', '${answer_stamp}', '${end_stamp}',
        '${duration}', '${billsec}', '${hangup_cause}', '${uuid}',
        '${accountcode}', '${read_codec}', '${write_codec}',
        '${sip_hangup_disposition}', '${network_addr}'
      );
    </template>
  </templates>
</configuration>

sqlite3 /var/lib/freeswitch/db/cdr.db

-- Show recent calls
SELECT caller_id_number, destination_number, duration, hangup_cause, start_stamp
FROM cdr
ORDER BY start_stamp DESC
LIMIT 20;

-- Call volume by hour
SELECT strftime('%H', start_stamp) AS hour, COUNT(*) AS calls
FROM cdr
WHERE date(start_stamp) = date('now')
GROUP BY hour
ORDER BY hour;

-- Average call duration by destination
SELECT destination_number,
       COUNT(*) AS calls,
       AVG(billsec) AS avg_duration,
       SUM(billsec) AS total_seconds
FROM cdr
WHERE billsec > 0
GROUP BY destination_number
ORDER BY calls DESC;

<configuration name="cdr_pg.conf" description="CDR PostgreSQL">
  <settings>
    <param name="host" value="127.0.0.1"/>
    <param name="port" value="5432"/>
    <param name="dbname" value="freeswitch_cdr"/>
    <param name="user" value="freeswitch"/>
    <param name="password" value="YOUR_DB_PASSWORD"/>
    <param name="legs" value="a"/>
    <param name="default-template" value="default"/>
    <!-- Log errors to file if DB is unavailable -->
    <param name="spool-format" value="csv"/>
    <param name="log-dir" value="/var/log/freeswitch/cdr-pg-errors"/>
  </settings>

  <templates>
    <template name="default">
      INSERT INTO cdr (
        caller_id_name, caller_id_number, destination_number, context,
        start_stamp, answer_stamp, end_stamp, duration, billsec,
        hangup_cause, uuid, accountcode, read_codec, write_codec,
        sip_hangup_disposition, network_addr
      ) VALUES (
        '${caller_id_name}', '${caller_id_number}', '${destination_number}',
        '${context}',
        to_timestamp(${start_epoch}), to_timestamp(${answer_epoch}),
        to_timestamp(${end_epoch}),
        ${duration}, ${billsec}, '${hangup_cause}', '${uuid}',
        '${accountcode}', '${read_codec}', '${write_codec}',
        '${sip_hangup_disposition}', '${network_addr}'
      );
    </template>
  </templates>
</configuration>