Using Bitcall with FreeSWITCH (SIP Trunk Setup)

“Turn your FreeSWITCH server into a Bitcall-powered telecom powerhouse… but without the headache.”

Let’s walk you through adding Bitcall step-by-step.

🔧 What You’ll Need

A running FreeSWITCH server
A Bitcall account
A SIP account in Bitcall (with credentials or with your IP whitelisted)
SSH access (for editing configs, running commands)

⚙️ Step 1: Create Your SIP Gateway

A) Using

SIP Credentials

(user/password)

Create a file:

/usr/local/freeswitch/conf/sip_profiles/external/bitcall.xml

<include>
  <gateway name="bitcall">
    <param name="username" value="YOUR_USER"/>
    <param name="password" value="YOUR_PASS"/>
    <param name="proxy" value="gateway.bitcall.io"/>
    <param name="register" value="true"/>
    <param name="caller-id-in-from" value="true"/>
    <param name="expire-seconds" value="3600"/>
    <param name="inbound-codec-prefs" value="PCMU,PCMA"/>
    <param name="outbound-codec-prefs" value="PCMU,PCMA"/>
  </gateway>
</include>

💡 Replace YOUR_USER and YOUR_PASS with your Bitcall SIP credentials.

B) Using

IP Authentication

(Trusted IP)

Create the same file but without user/pass:

<include>
  <gateway name="bitcall-ip">
    <param name="proxy" value="gateway.bitcall.io"/>
    <param name="register" value="false"/>
    <param name="caller-id-in-from" value="true"/>
  </gateway>
</include>

⚠️ Make sure your server’s public IP is whitelisted in the Bitcall portal.

🔄 Step 2: Reload FreeSWITCH Settings

Run:

fs_cli -x "reloadxml"
fs_cli -x "sofia status gateway bitcall"

You should see REGED if using credentials .

📞 Step 3: Define Outbound Dialplan

Create file:

/usr/local/freeswitch/conf/dialplan/default/30_bitcall_out.xml

<extension name="bitcall_out">
  <condition field="destination_number" expression="^(\+33\d{9})$">
    <!-- random French CLI from your signed DIDs -->
    <action application="set" data="effective_caller_id_number=${caller_id_number}"/>
    <action application="bridge" data="sofia/gateway/bitcall/$1"/>
  </condition>
</extension>

To randomize French CLI, leverage a script or DID pool — see our PHP/JS snippet below.

🎲 Step 4: Random CLI Setup

If you have multiple DIDs in a database or file, use a small script to pick one per call:

<action application="set" data="effective_caller_id_number=${curl('http://localhost/random_cli.php')}"/>

random_cli.php might contain:

<?php
$lines = file('/etc/freeswitch/french_dids.txt', FILE_IGNORE_NEW_LINES);
echo trim($lines[array_rand($lines)]);

Saves DIDs like:

+33612345678 +33798765432

🔍 Step 5: Reload Dialplan & Test

fs_cli -x "reloadxml"

Test a call:

fs_cli -x "originate sofia/gateway/bitcall/+33123456789 &playback(silence_stream://200)"

If you hear silence, your trunk is working!

🛠️ Troubleshooting Tips

Problem	Quick Fix
Gateway shows NOREG or down	Check credentials or ensure IP-auth is whitelisted
Calls don’t go out	Confirm dialplan syntax and reload XML
One-way audio	Verify ext-sip-ip and ext-rtp-ip are set properly
CLI not working	Ensure script runs and returns correctly formatted DID
Codec mismatch	Adjust inbound-codec-prefs or outbound-codec-prefs

🧠 TL;DR Recap

✅ Add gateway in sip_profiles/external (with creds or IP)  
✅ Reload XML and check gateway status  
✅ Create outbound dialplan rule for +33 numbers  
✅ (Optional) Add random French CLI using script  
✅ Reload and test a call  
✅ Bam! Bitcall + FreeSWITCH = 🔥