Connect FreeSWITCH to TelecomsXchange

This guide connects FreeSWITCH to TelecomsXchange with a Sofia gateway for outbound calls and an ACL for inbound calls on routes you sell or numbers you bought. For every TelecomsXchange value in one place, see Connect a SIP platform.

Before you start#

• A SIP account with FreeSWITCH's public IP, for example 203.0.113.10, or a VoIP login and password from Edit Account → Add SIP Digest Auth.
• A purchased route and its tech prefix from My Interconnections, for example 80039#.
• The external Sofia profile running with ext-sip-ip and ext-rtp-ip set to your public IP if you are behind NAT.
• To sell: a route with IP Address:Port pointing at FreeSWITCH, for example 203.0.113.10:5080, and the values in Interconnect Info.

How do I send calls to TelecomsXchange?#

1. Create the gateway file:

<!-- conf/sip_profiles/external/tcxc.xml -->
<include>
  <gateway name="tcxc">
    <param name="proxy" value="sip01.telecomsxchange.com:5060"/>
    <param name="register" value="false"/>          <!-- TCXC is peering only -->
    <param name="register-transport" value="udp"/>
    <param name="caller-id-in-from" value="true"/>  <!-- put the caller ID in From -->
    <!-- IP authentication: username and password are not checked;
         digest authentication: the account's VoIP Login and VoIP Password -->
    <param name="username" value="VOIP_LOGIN"/>
    <param name="password" value="VOIP_PASSWORD"/>
  </gateway>
</include>

With IP authentication, leave any placeholder values in username and password; TelecomsXchange recognises FreeSWITCH by its source IP. With digest authentication, use the real VoIP Login and VoIP Password: FreeSWITCH answers the 401 or 407 challenge with them without registering.

2. Load it: in fs_cli, run sofia profile external rescan, then sofia status gateway tcxc.

3. Add a dialplan extension that strips + or 00 and puts the tech prefix in front. TECH_PREFIX is the route's prefix from My Interconnections:

<!-- conf/dialplan/default/90_tcxc.xml -->
<include>
  <extension name="to-tcxc">
    <condition field="destination_number" expression="^(?:\+|00)?([1-9]\d{6,14})$">
      <action application="set" data="tcxc_prefix=80039#"/>  <!-- TECH_PREFIX -->
      <action application="set" data="hangup_after_bridge=true"/>
      <action application="bridge" data="sofia/gateway/tcxc/${tcxc_prefix}$1"/>
    </condition>
  </extension>
</include>

Run reloadxml. Put the extension in the context your users or customers dial from, after any local extensions it would otherwise catch.

How do I receive calls from TelecomsXchange?#

Calls on routes you sell, and on numbers you bought, come from 174.136.44.213 over UDP, from source ports 5060 to 5072, with the number in E.164 format such as 14165551234.

1. Allow 174.136.44.213 on your firewall for SIP to the profile's port, and open your RTP range (by default 16384 to 32768).
2. Add an ACL list:

<!-- conf/autoload_configs/acl.conf.xml, inside <network-lists> -->
<list name="tcxc" default="deny">
  <node type="allow" cidr="174.136.44.213/32"/>
</list>

3. Apply it to the profile that receives the calls, then run reloadacl and restart that profile (sofia profile external restart):

<!-- conf/sip_profiles/external.xml, in <settings> -->
<param name="apply-inbound-acl" value="tcxc"/>

If the profile already applies an inbound ACL, add the node to that list instead.

4. Calls on the external profile enter the public context. Route only TelecomsXchange calls onward:

<!-- conf/dialplan/public/10_from_tcxc.xml -->
<include>
  <extension name="from-tcxc">
    <condition field="network_addr" expression="^174\.136\.44\.213$"/>
    <condition field="destination_number" expression="^(\d+)$">
      <action application="transfer" data="$1 XML from-tcxc"/>
    </condition>
  </extension>
</include>

In the from-tcxc context, bridge each number to an extension, or for a seller route to your upstream gateway. If your route has an Outgoing Prefix or strips digits, match the number in that format.

How do I receive calls on numbers I bought?#

Set the number's SIP Endpoint (Primary) in DID Settings to FreeSWITCH, for example sip:14165551234@203.0.113.10:5080 for the external profile. The call arrives from 174.136.44.213, passes the ACL, and the from-tcxc extension above transfers it by destination_number.

How do I test it?#

1. In fs_cli, run sofia global siptrace on (and sofia loglevel all 9 for more detail), or capture with sngrep port 5060 or port 5080.
2. Place a call: originate sofia/gateway/tcxc/80039#447700900123 &echo.
3. Check the INVITE's Request-URI starts with the tech prefix, and read the reply code.
4. In the portal, watch the call in Live Calls and find its SIP code in Logs → CDR (Voice). Turn tracing off with sofia global siptrace off.

What if something goes wrong?#

References#

• FreeSWITCH gateways configuration
• Sofia gateway authentication parameters
• FreeSWITCH access control lists
• mod_sofia