[Freeswitch-users] Sending SAVPF INVITE to Opensips

[James Mortensen]

Hello,

I have a bandwidth.com number pointed to opensips, and a WebRTC peer
registered with Opensips.  I'm trying to dial the 10 digit number from a
cell phone and connect the call through FreeSWITCH to the Chrome WebRTC
client.


I defined opensips as a gateway, in the external profile:

<include>
   <gateway name="ws-Opensips">
     <!-- <param name="from-user" value="fromuser"/> -->
     <param name="from-domain" value="54.X.X.75"/>
     <param name="proxy" value="54.X.X.75"/>
     <param name="expire-seconds" value="600"/>
     <param name="register" value="false"/>
     <param name="retry_seconds" value="30"/>
     <param name="extension" value="18257773456"/>
     <param name="context" value="public"/>
     <param name="avpf" value="yes"/>
     <param name="username" value="11234"/>
     <param name="password" value="password"/>
   </gateway>
</include>


In the public dialplan context, I added in a condition to catch the INVITE
coming in from opensips and pass it to a context I've called
"default-inbound". See the second condition:

 <extension name="from_opensips">
    <condition field="network_addr" expression="^54\.X\.X\.75$"
break="never"> <!--CUSTOMIZE-->
      <action application="transfer" data="${destination_number} XML
default"/>
    </condition>
    <condition field="network_addr" expression="^54\.X\.X\.111$">
<!--CUSTOMIZE Use a third context here -->
      <action application="transfer" data="${destination_number} XML
default-inbound"/>
    </condition>
  </extension>


Then, in the default-inbound context, I match the dialed number, answer the
call leg from the PSTN, and then try to transfer back through opensips to
oversip and to Chrome.  The problem is that I either end up sending back
AVP INVITES, or Opensips refuses to authenticate the user.

<extension name="bandwidth.com inbound bridge">
    <condition field="destination_number" expression="^\+1(5035551212)$">
       <action application="answer" />
       <action application="set" data="variable_sip_auth_username=11234"/>
       <action application="set"
data="variable_sip_auth_password=password"/>
       <action application="bridge"
data="sofia/external/ws-Opensips/11234 at 54.X.X.75"/>

    </condition>
  </extension>


As you can see, I've hard-coded my peer, 11234, in the configuration. This
is a registered user on Opensips.

How can I get FreeSWITCH to send the SAVPF INVITE through to Opensips for
the WebRTC portion of the call leg?  I apologize if this is covered
somewhere, but I've been wracking my brain on this for days and am not
getting anywhere.

The Opensips configuration I have works with existing Asterisk 11 servers,
and I'm hoping I can just simply plug in FreeSWITCH servers seamlessly into
the mix.

Thank you!

James
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