[Freeswitch-users] FS and Asterisk connectivity

[Brian Snipes]

I wish to connect FS to * for interoperability testing and can't seem to
get my configs correct.  Has anyone done this already and if so can you
post your configs?

I can call from FS to the asterisk side via a dialplan entry where
^9(.*)$ passes to a gateway I've setup:

    <extension name="out9_asterisk">
      <condition field="destination_number" expression="^9(.*)$">
        <action application="bridge" data="sofia/gateway/asterisk/$1"/>
      </condition>
    </extension>

sip_profiles/default/asterisk.xml :

<include>
  <gateway name="asterisk">
  <param name="username" value="freeswitch"/>
  <param name="realm" value="x.x.x.x"/>
  <param name="password" value="password"/>
  <param name="register" value="false"/>
  </gateway>
</include>

The problem I get with this is no audio either way.  I am on rev 8099.

TIA,
Brian