FreeSWITCH configuration guide

⚠ These are placeholder values — set the real ones in the editor (Global tab) and they update on every guide at once.

1. Create the gateway

Create conf/sip_profiles/external/sipandvoip.xml:

<include>
  <gateway name="SipandVoip">
    <param name="username"    value="YOUR_USERNAME"/>
    <param name="password"    value="YOUR_PASSWORD"/>
    <param name="realm"       value="sip.sipandvoip.com.cy"/>
    <param name="proxy"       value="sip.sipandvoip.com.cy"/>
    <param name="register"    value="true"/>
    <param name="expire-seconds" value="3600"/>
    <param name="retry-seconds"  value="30"/>
    <param name="from-user"   value="YOUR_USERNAME"/>
    <param name="from-domain" value="sip.sipandvoip.com.cy"/>
    <param name="extension"   value="YOUR_USERNAME"/>
    <param name="context"     value="public"/>
    <param name="caller-id-in-from" value="false"/>
    <param name="ping"        value="30"/>
  </gateway>
</include>

2. Set the external profile for NAT

If FreeSWITCH sits behind NAT, make sure vars.xml has your public address:

<X-PRE-PROCESS cmd="set" data="external_rtp_ip=YOUR.PUBLIC.IP"/>
<X-PRE-PROCESS cmd="set" data="external_sip_ip=YOUR.PUBLIC.IP"/>

Use stun:stun.freeswitch.org instead of a literal IP only if your public address is dynamic.

3. Outbound dialplan

In conf/dialplan/default.xml:

<extension name="sipandvoip_outbound">
  <condition field="destination_number" expression="^(\+?\d{6,15})$">
    <action application="set" data="effective_caller_id_number=YOUR_DID"/>
    <action application="set" data="hangup_after_bridge=true"/>
    <action application="bridge" data="sofia/gateway/SipandVoip/$1"/>
  </condition>
</extension>

4. Inbound dialplan

In conf/dialplan/public.xml, match your DID and send it internally:

<extension name="sipandvoip_inbound">
  <condition field="destination_number" expression="^(YOUR_DID)$">
    <action application="set" data="domain_name=$${domain}"/>
    <action application="transfer" data="1000 XML default"/>
  </condition>
</extension>

5. Reload and verify

fs_cli -x "reloadxml"
fs_cli -x "sofia profile external restart reloadxml"
fs_cli -x "sofia status gateway SipandVoip"

The gateway should report State: REGED. Use sofia global siptrace on to capture a trace while testing.

If it does not register

• 401 / 403 on REGISTER — username or password is wrong, or the device is presenting a different username in the auth header than in the From header. Check both.
• No response at all — a firewall is dropping SIP. Allow outbound UDP 5060 (or TCP/TLS 5061) and the RTP range to our network, and make sure any SIP ALG on the router is disabled. SIP ALG causes more problems than it solves.
• Registers, but one-way or no audio — almost always NAT. Set the external address on your PBX, forward or open the RTP port range, and confirm the device is advertising its public IP in SDP.
• Calls drop after ~30 seconds — SIP session timers or ACK not reaching us. Enable session timers and re-check NAT traversal settings.
• Outbound rejected with 404 — number format. Send E.164 with the leading + or full international format without a trunk prefix.

Still stuck? Send us the SIP trace and we will look at it against our side of the call.