I've hooked up my FreeSWITCH to an external SIP gateway.

The following scenarios work:

FS -> FS

SIP GW -> FS

But the following scenario does not work:

FS -> SIP GW

I only have one single dial plan in my setup:

<include>

  <extension name="test-dialplan">

    <condition field="destination_number" expression="^\+(\d{3,20})$">

      <action application="export" data="dialed_extension=$1"/>

      <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/>

      <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/>

      <action application="bridge" data="user/${dialed_extension}@${domain_name}"/>

      <action application="bridge" data="user/${dialed_extension}@${domain_name},sofia/gateway/sip-outbound/+${dialed_extension}"/>

    </condition>

  </extension>

</include>

My users are named the same as real GSM numbers (country code and number).

I would like my FS to call internally if the user exists, if not, then route it to my SIP gateway for connecting to the GSM world.

As mentioned, everything works except outgoing calls to the SIP gateway.

What goes wrong?