[Freeswitch-users] FS and Asterisk connectivity

Brian Snipes bsnipes at snipes.org
Mon Apr 14 12:49:48 PDT 2008

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"/>

sip_profiles/default/asterisk.xml :

  <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"/>

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


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