well if pyte was around he would likely have a conclusive answer, but try this as an aid.

I pulled this from a v3 configuration which does know how to configure SIPSTATION trunks in FreeSWITCH. (which happens to be fairly generic and would work with many SIP providers, but your best service will clearly be with SIPSTATION :)

<?xml version="1.0"?>
<include>
  <profile name="sipinterface_1">
    <domains>
      <domain name="all" alias="true" parse="false"/>
    </domains>
    <settings>
      <param name="rtp-ip" value="192.168.1.5"/> <!-- should be your pbx IP -->
      <param name="sip-ip" value="192.168.1.5"/> <!-- should be your pbx IP -->
      <param name="sip-port" value="5060"/>
      <param name="ext-rtp-ip" value="auto-nat"/>
      <param name="ext-sip-ip" value="auto-nat"/>
      <param name="auth-calls" value="true"/>
      <param name="aggressive-nat-detection" value="true"/>
      <param name="user-agent-string" value="Configured by TCAPI"/>
      <param name="apply-nat-acl" value="nat.auto"/> <!-- I don't know if this will give issues -->
      <param name="context" value="context_2"/> <!-- this should be your incoming context for calls -->
      <param name="rtp-timer-name" value="soft"/>
      <param name="codec-prefs" value="$${global_codec_prefs}"/> <!-- I think this needs to be configured somewhere -->
      <param name="inbound-codec-negotiation" value="generous"/>
    </settings>
    <gateways>
      <gateway name="trunk_1">
        <param name="username" value="your_sip_username"/>
        <param name="password" value="your_sip_password"/>
        <param name="register" value="true"/>
        <param name="realm" value="trunk1.freepbx.com"/>
        <param name="ping" value="25"/> <!-- this helps to keep nat holes open -->
        <param name="expire-seconds" value="60"/>
      </gateway>
    </gateways>
  </profile>
</include>